Resource Management in Satellite Networks

Giovanni Giambene
Editor
Resource
Management in
Satellite Networks
Optimization and Cross-Layer Design
Resource Management in Satellite
Networks
Resource Management in Satellite
Networks
Giovanni Giambene
Universit degli Studi di Siena
13
Giovanni Giambene
Dipartimento di Ingegneria DellInformazione
Universit degli Studi di Siena
Via Roma, 56
53100 Siena
ITALY
Library of Congress Control Number: 2007922349
ISBN 0-387-36897-3 e-ISBN 0-387-53991-3

ISBN 978-0-387-36897-9 e-ISBN 978-0-387-53991-1
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2007 Springer Science+Business Media, LLC

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Acknowledgements
The volume Editor and the authors would like to acknowledge

the FP6 EU Network of Excellence SatNEx II (IST-027393) and,
in particular, activities ja2230&2330 for the research work as the
basis of this book.
Preface
Nowadays, satellites are used for a variety of purposes, including sensors and
data collection, weather, maritime navigation and timing, Earth observation,
and communications. In particular, satellite transmissions have an important
role in telephone communications, television broadcasting, computer commu-
nications as well as navigation.
The use of satellites for communications was a brilliant idea of Arthur C.
Clarke who wrote a famous article in October 1945 in the Wireless World jour-
nal, entitled Extra Terrestrial Relays - Can Rocket Stations Give Worldwide
Coverage? that described the use of manned satellites in orbits at 35,800
km altitude, thus having synchronous motion with respect to a point on the
Earth. This article was the basis for the use of GEOstationary (GEO) satel-
lites for telecommunications. Subsequently, he also proved the usefulness of
satellites as compared to transatlantic telephone cables.
Satellite communications deserve the special merit to allow connecting
people at great distances by using the same (homogeneous) communication
system and technology. Other very signicant advantages of the satellite ap-
proach are: (i ) easy fruition of both broadcast and multicast high bit-rate
multimedia services; (ii ) provision of backup communication services for users
on a global scale (this feature is very important for emergency scenarios and
disaster relief activities); (iii ) provision of services in areas that could not be
reached by terrestrial infrastructures; (iv ) support of high-mobility users.
Three broad areas where satellites can be employed are: xed satellite
service, broadcast satellite service, and mobile satellite service. Particularly
relevant is the signicant global success of broadcast satellite services for both
analogue and digital audio/TV by exploiting the inherent wide coverage area
of GEO satellites. At the beginning of the 21st century more than 70 million
European homes watch TV programs through direct satellite reception or
through cable distribution systems.
New satellite system architectures are being envisaged to be fully IP-based
and support digital video broadcasting and return channel protocols, such as
DVB-S, DVB-S2 and DVB-RCS. Trends in telecommunications indicate that
viii Preface
four growing market areas are messaging and navigation services, mobility ser-
vices, video delivery services, and interactive multimedia services. In addition
to this, interesting areas for investigation with big potential markets are: the
extension of the DVB-S2/-RCS standard for mobile usage, satellite IP net-
works interconnected with terrestrial wireless systems, and the convergence
of satellite communications and remote sensing for Earth observation.
Satellite resources (i.e., radio spectrum and transmission power) are costly
and satellite communications impose special constraints with respect to ter-
restrial systems in terms of path loss, propagation delay, fading, etc. These
are critical factors for supporting user service level agreements and Quality of
Service (QoS).
The ISO/OSI reference model and the Internet protocol suite are based
on a layered protocol stack. Protocols are designed such that a higher-layer
protocol only makes use of the services provided by the lower layer and is
not concerned with the details of how the service is being provided; proto-
cols at the dierent layers are independently designed. However, there is tight
interdependence between layers in IP-based next-generation satellite commu-
nication systems. For instance, transport layer protocols need to take into
account large propagation delays, link impairments, and bandwidth asymme-
try. In addition to this, error correction schemes are implemented at physical,
link and (in some cases) transport layers, thus entailing some ineciencies and
redundancies. Hence, strict modularity and layer independence of the layered
protocol model may lead to a non-optimal performance.
Satellite resources are costly and must be eciently utilized in order to
provide suitable revenue to operators. Users, however, do not care about the
platform technology adopted and employed resource management scheme, but
need QoS provision. Unfortunately, resource utilization eciency and QoS
support are conicting needs: typically, the best utilization is achieved in the
presence of a congested system, where QoS can diculty be guaranteed. A
new possible approach addressing both these issues is represented by the cross-
layer design of the air interface, where the interdependency of protocols at
dierent layers is exploited with the aim to perform a joint optimization or a
dynamic adaptation. The innovation of this approach relies on the fact that
it introduces direct interactions event between non-adjacent protocol layers
with the aim to improve system performance.
The main aim of this book is to address the novel research area of cross-
layer air interface design for satellite systems and provide a complete de-
scription of available methods, showing the possible eciency improvements.
A particular interest has been addressed here to the protocol stack dened
by the ETSI TC-SES/BSM (Satellite Earth Stations and Systems / Broad-
band Satellite Multimedia) working group for IP-based satellite networks. In
this framework, a protocol stack architecture has been identied, where lower
layers depend on satellite system implementation (satellite-dependent layers)
and higher layers are those typical of the Internet protocol stack (satellite-
independent layers). These two blocks of stacked protocols are interconnected
Preface ix
through the SI-SAP (Satellite-Independent - Service Access Point) interface

that has acquired a crucial importance for the denition of cross-layer inter-
actions and signaling.
This book has been conceived in the framework of the SatNEx Network of
Excellence (www.satnex.org, project IST-507052, 20042006) that has made
possible a tight cooperation of many European partners. Since the beginning
(January 2004), SatNEx devoted the sub-work-package 2430, namely joint
activity 2430 (ja2430), to the investigation of cross-layer issues that were soon
considered as an original research eld. Such activity attracted the interest of
more than 14 SatNEx partners. In particular, research groups at the following
European Universities or research Institutions contributed to ja2430:
AUTh - Aristotle University of Thessaloniki, Greece
CNIT - Consorzio Nazionale Interuniversitario per le Telecomunicazioni,
Italy
DLR - Deutsches Zentrum f ur Luft- und Raumfahrt e.V., Germany
FhI - Fraunhofer Institute for Open Communication Systems, Germany
ISTI - National Research Council (CNR), ISTI Institute, Italy
RWTH - Rheinisch-Westfalische Technische Hochschule Aachen,
COMNETS, Germany
TeSA - France
TUG - Graz University of Technology, Austria
UAB - Universidad Autonoma de Barcelona, Spain
UC3M - Universidad Carlos III de Madrid, Spain
UoA - University of Aberdeen, UK
UniS - University of Surrey, Centre for Communication Systems Research,
UK
UToV - University of Rome Tor Vergata, Department of Electronic En-
gineering, Italy
UVI - Universidad de Vigo, Departamento de Ingeniera Telematica, Spain.
I had the pleasure to coordinate the ja2430 activities, organizing 4 peri-

odical meetings (plus ad hoc meetings dedicated to the coordination of this
book activity), where objectives (organized according to Focus Topics, FTs),
common scenarios and strategies were identied. In particular, the FTs below
were dened, thus contributing to the dierent parts of this book:
FT 1: QoS for multimedia trac
FT 2: Radio resource management
FT 3: Protocol integration.
The main objective of ja2430 has been the study of novel radio resource
management schemes able to support multimedia trac with QoS guarantee
in future satellite communication systems. Our aim has been to propose mod-
ications to the ISO/OSI standard protocol stack by considering interactions
x Preface
and even new interfaces among non-adjacent protocol layers. Such approach
can be particularly important in order to optimize the performance (i.e., e-
ciency) of resource management protocols.
After more than one year of SatNEx ja2430 activities, it was decided in
September 2005 to organize the results obtained in a book. With the end of
SatNEx activities in March 2006, the work of this book continued in SatNEx
II (IST-027393, 20062009) in the two new sub-work-packages deriving from
ja2430, that is ja2330 (entitled: Radio Resource Allocation and Adaptation)
and ja2230 (entitled: Cross-Layer Protocol Design).
The activity carried out for this book has been a very good opportunity
for the SatNEx community to integrate the competencies of dierent partners
considering all the parts of the system design (i.e., propagation issues, resource
management techniques, link design, QoS, transport protocols, etc.) and es-
pecially because SatNEx is unique in that its expertise covers both broadband
(xed) and mobile satellite systems. This has been an ideal condition for the
study of mechanisms that involve interactions among several protocol layers.
Besides Part I of this book that is aimed to introduce satellite communica-
tions (Chapter 1), resource management techniques (Chapter 2), QoS issues
(Chapter 3) and cross-layer design methods (Chapter 4), the two following
parts are conceived according to the ETSI SES/BSM protocol stack, thus
distinguishing cross-layer issues involving satellite-dependent layers (Part II,
Chapters 5, 6 and 7) from those of satellite-independent layers (Part III,
Chapters 8, 9 and 10).
Before concluding this preface, I would like to say that I feel honored to
have coordinated this book work rst in the framework of ja2430 and then
in ja2230&ja2330. I take this opportunity to thank SatNEx for the econom-
ical support received and all the SatNEx Colleagues who have provided a
continuous support to this initiative. Finally, a very special thank is for my
Collaborator, Dr. Ing. Paolo Chini, for his signicant support in helping me
during these years of hard work on the book. Many thanks also to my Col-
laborator, Dr. Ing. Ivano Alocci, for his kind support.
Giovanni Giambene
CNIT - University of Siena
Via Roma, 56 - 53100 Siena, Italy
Phone: +39 0577 234603
Fax: +39 0577 233602
E-mail: giambene@unisi.it
Curriculum Vitae
Dr. Giovanni Giambene
Giovanni Giambene was born in Florence, Italy, in 1966. He received the Dr.
Ing. degree in Electronics from the University of Florence, Italy, in 1993 and
the Ph.D. degree in Telecommunications and Informatics from the University
of Florence, Italy, in 1997. From 1994 to 1997, he was with the Electronic En-
gineering Department of the University of Florence, Italy. He was Technical
External Secretary of the European Community COST 227 Action, entitled
Integrated Space/Terrestrial Mobile Networks. He also contributed to the
Resource Management activity of the Working Group 3000 within the RACE
Project, called Satellite Integration in the Future Mobile Network (SAINT,
RACE 2117). From 1997 to 1998, he was with OTE of the Marconi Group,
Florence, Italy, where he was involved in a GSM development program. In
the same period he also contributed to the COST 252 Action (Evolution of
Satellite Personal Communications from Second to Future Generation Sys-
tems) research activities by studying the performance of Packet Reservation
Multiple Access (PRMA) protocols suitable for supporting voice and data
transmissions in low earth orbit mobile satellite systems. In 1999 he joined
the Information Engineering Department of the University of Siena, Italy, rst
as research associate and then as assistant professor. He teaches the advanced
course of Telecommunication Networks at the University of Siena. From 1999
to 2003 he participated to the project Multimedialit` a , nanced by the Ital-
ian National Research Council (CNR). From 2000 to 2003, he contributed to
the activities of the Personalised Access to Local Information and services
for tOurists (PALIO) IST Project within the fth Research Framework of
the European Commission (www.palio.dii.unisi.it). At present, he is involved
in the SatNEx network of excellence of the FP6 programme in the satellite
eld, as work package leader of two groups on radio access techniques and
cross-layer air interface design (www.satnex.org). He is also vice-Chair of the
COST 290 Action (www.cost290.org), entitled Trac and QoS Management
in Wireless Multimedia Networks (Wi-QoST).
Contents
Acknowledgements . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . v
Preface . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . vii
Contents . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . xiii
List of Contributors . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . xix
List of Acronyms and Abbreviations . . . . . . . . . . . . . . . . . . . . . . xxiii
Part I Resource Management Framework for Satellite

Communications
1 INTRODUCTION TO SATELLITE
COMMUNICATIONS AND RESOURCE
MANAGEMENT . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3
1.1 Satellite communications . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3
1.2 Basic issues in the design of satellite communication systems . 10
1.3 Multiple access techniques . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 12
1.4 Radio interfaces considered and scenarios . . . . . . . . . . . . . . . . . . 15
1.4.1 S-UMTS . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 15
1.4.2 DVB-S standard . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 16
1.4.3 DVB-RCS standard . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 17
1.4.4 DVB-S2 standard . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 23
1.4.5 Numerical details on the selected scenarios for
performance evaluations . . . . . . . . . . . . . . . . . . . . . . . . . . . 27
1.5 Satellite networks . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 28
1.5.1 SI-SAP interface overview . . . . . . . . . . . . . . . . . . . . . . . . . 31
1.6 Novel approaches for satellite networks . . . . . . . . . . . . . . . . . . . . 34
1.6.1 Horizontal approach . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 34
xiv Contents
1.6.2 Vertical approach . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 34

1.7 Conclusions . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 37
References . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 39
2 ACTIVITY IN SATELLITE RESOURCE

MANAGEMENT . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 43
2.1 Introduction . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 43
2.2 Frequency/time/space resource allocation schemes . . . . . . . . . . 46
2.3 Power allocation and control schemes . . . . . . . . . . . . . . . . . . . . . 50
2.4 CAC and handover algorithms . . . . . . . . . . . . . . . . . . . . . . . . . . . 51
2.4.1 Handover algorithms . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 53
2.5 RRM modeling and simulation . . . . . . . . . . . . . . . . . . . . . . . . . . . 54
2.6 Related projects in Europe . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 55
2.6.1 TWISTER: Terrestrial Wireless Infrastructure
integrated with Satellite Telecommunications for
E-Rural applications . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 56
2.6.2 MAESTRO: Mobile Applications & sErvices based on
Satellite & Terrestrial inteRwOrking . . . . . . . . . . . . . . . . 56
2.6.3 SatNEx: Satellite Network of Excellence . . . . . . . . . . . . . 57
2.6.4 NEWCOM: Network of Excellence in Wireless
COMmunications . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 57
2.6.5 VIRTUOUS: Virtual Home UMTS on Satellite . . . . . . . 58
2.6.6 COST Actions . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 58
2.6.7 The ISI Initiative . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 59
2.7 Conclusions . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 60
References . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 61
3 QoS REQUIREMENTS FOR MULTIMEDIA

SERVICES . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 67
3.1 Introduction . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 67
3.2 Services QoS requirements . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 68
3.2.1 Performance requirements for conversational services . . 70
3.2.2 Performance requirements for interactive services . . . . . 73
3.2.3 Performance requirements for streaming services . . . . . 74
3.2.4 Performance requirements for background
services-applications . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 76
3.3 IP QoS frameworks/models . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 76
3.4 Broadcast and multicast services . . . . . . . . . . . . . . . . . . . . . . . . . 80
3.4.1 Delayed real-time service over GEO satellite
distribution systems . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 83
3.4.2 Scenario characterization and results . . . . . . . . . . . . . . . . 85
3.5 Experimental results on QoS . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 89
3.6 Conclusions . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 92
References . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 93
Contents xv
4 CROSS-LAYER APPROACHES FOR RESOURCE

MANAGEMENT . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 95
4.1 Introduction . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 95
4.2 Literature survey on cross-layer methods . . . . . . . . . . . . . . . . . . 96
4.3 The need of a cross-layer air interface design . . . . . . . . . . . . . . . 102
4.4 Cross-layer design: requirements depending on the satellite
scenario . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 105
4.4.1 Broadband satellite scenario requirements
(DVB-S/S2) . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 105
4.4.2 Mobile satellite scenario requirements (S-UMTS) . . . . . 108
4.4.3 LEO satellite scenario requirements . . . . . . . . . . . . . . . . . 108
4.5 Conclusions . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 111
References . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 113
Part II Cross-Layer Techniques for Satellite-Dependent Layers
5 ACCESS SCHEMES AND PACKET SCHEDULING

TECHNIQUES . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 119
5.1 Introduction . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 119
5.2 Uplink: access schemes . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 120
5.2.1 Random access in UMTS and application to S-UMTS . 121
5.2.2 The Packet Reservation Multiple Access (PRMA)
protocol . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 129
5.2.3 Adopting PRMA-like schemes in S-UMTS . . . . . . . . . . . 131
5.2.4 Stability analysis of access protocols . . . . . . . . . . . . . . . . 132
5.3 Downlink: scheduling techniques . . . . . . . . . . . . . . . . . . . . . . . . . . 134
5.3.1 Survey of scheduling techniques . . . . . . . . . . . . . . . . . . . . 134
5.3.2 Scheduling techniques for HSDPA via satellite . . . . . . . 139
5.3.3 Scheduling techniques for broadcast and multicast
services in S-UMTS . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 152
5.3.4 Packet scheduling with cross-layer approach . . . . . . . . . 164
5.4 Conclusions . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 170
References . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 173
6 CALL ADMISSION CONTROL . . . . . . . . . . . . . . . . . . . . . . . . . . 177

6.1 Introduction to Call Admission Control . . . . . . . . . . . . . . . . . . . 177
6.2 CAC and QoS management . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 179
6.3 CAC algorithms for GEO satellite systems . . . . . . . . . . . . . . . . . 184
6.3.1 CAC schemes for MF-TDMA networks . . . . . . . . . . . . . . 184
6.3.2 CAC schemes for CDMA networks . . . . . . . . . . . . . . . . . 188
6.4 Handover and CAC algorithms for non-GEO satellite systems 189
6.4.1 Intra-satellite handover and CAC schemes . . . . . . . . . . . 191
6.4.2 Inter-satellite handover and CAC schemes . . . . . . . . . . . 194
xvi Contents
6.5 Directions for further research . . . . . . . . . . . . . . . . . . . . . . . . . . . . 199

6.6 Conclusions . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 200
References . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 201
7 DYNAMIC BANDWIDTH ALLOCATION . . . . . . . . . . . . . . . 207

7.1 Dynamic bandwidth allocation: problem denition . . . . . . . . . . 207
7.1.1 Survey of allocation approaches . . . . . . . . . . . . . . . . . . . . 209
7.2 DBA schemes for DVB-RCS scenarios . . . . . . . . . . . . . . . . . . . . . 211
7.3 Recent developments on DBA techniques . . . . . . . . . . . . . . . . . . 213
7.3.1 DVB-RCS dynamic channel allocation using
control-theoretic approaches . . . . . . . . . . . . . . . . . . . . . . . 213
7.3.2 Dynamic bandwidth de-allocation . . . . . . . . . . . . . . . . . . 214
7.3.3 Dynamic bandwidth allocation with cross-layer issues . 214
7.3.4 Joint timeslot optimization and fair dynamic
bandwidth allocation in a system employing adaptive
coding . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 218
7.3.5 Dynamic bandwidth allocation for handover calls . . . . . 233
7.4 Conclusions . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 234
References . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 237
Part III Cross-Layer Techniques for Satellite-Independent Layers
8 RESOURCE MANAGEMENT AND NETWORK LAYER 243

8.1 Introduction . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 243
8.2 Overview IP QoS framework . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 244
8.2.1 Integrated services . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 244
8.2.2 Dierentiated services . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 246
8.2.3 Multiprotocol Label Switching (MPLS) . . . . . . . . . . . . . 247
8.3 Resource management for IP QoS . . . . . . . . . . . . . . . . . . . . . . . . 248
8.3.1 Relative DiServ by MAC Scheduling . . . . . . . . . . . . . . . 249
8.4 QoS mapping over satellite-independent service access point . 256
8.4.1 Model-based techniques for QoS mapping
and support . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 257
8.4.2 A measurement-based approach for QoS mapping
and support . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 258
8.4.3 Performance evaluation and discussion . . . . . . . . . . . . . . 262
8.5 QoS provisioning for terminals supporting dual network
access - satellite and terrestrial . . . . . . . . . . . . . . . . . . . . . . . . . . . 264
8.6 Switched Ethernet over LEO satellite: implicit cross-layer
design exploiting VLANs . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 270
8.6.1 Protocol harmonization and implicit cross-layer design
via IEEE VLAN . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 272
8.6.2 Performance evaluation . . . . . . . . . . . . . . . . . . . . . . . . . . . 273
Contents xvii
8.7 Conclusions . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 282
References . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 285
9 RESOURCE MANAGEMENT AND TRANSPORT

LAYER . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 289
9.1 Introduction . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 289
9.2 Overview of TCP over satellite . . . . . . . . . . . . . . . . . . . . . . . . . . . 290
9.2.1 TCP standard mechanisms . . . . . . . . . . . . . . . . . . . . . . . . 291
9.2.2 Criticalities of TCP on satellite links . . . . . . . . . . . . . . . . 292
9.2.3 Survey of proposed solutions . . . . . . . . . . . . . . . . . . . . . . . 293
9.3 Cross-layer interaction between TCP and physical layer . . . . . 294
9.4 Cross-layer interaction between TCP and MAC . . . . . . . . . . . . 298
9.4.1 A novel TCP-driven dynamic resource
allocation scheme . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 299
9.5 Overview of UDP-based multimedia over satellite . . . . . . . . . . . 305
9.5.1 Cross-layer methods for UDP . . . . . . . . . . . . . . . . . . . . . . 307
9.6 Conclusions . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 307
References . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 309
10 CROSS-LAYER METHODS AND STANDARDIZATION

ISSUES . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 313
10.1 Introduction . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 313
10.2 Cross-layer design and Internet protocol stack . . . . . . . . . . . . . . 314
10.3 Cross-layer methodologies for satellite systems . . . . . . . . . . . . . 314
10.3.1 Implicit and explicit cross-layer design methodologies . 315
10.3.2 Cross-layer techniques categorized in terms of the
direction of information ow . . . . . . . . . . . . . . . . . . . . . . . 315
10.4 Potential cross-layer optimizations for satellite systems . . . . . . 317
10.4.1 Optimizations aiming at QoS harmonization across
layers . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 317
10.4.2 Optimization of the Radio Resource Management . . . . 318
10.4.3 Optimizations combining higher and lower layers . . . . . 319
10.5 Cross-layer signaling for satellite systems . . . . . . . . . . . . . . . . . . 320
10.6 Standardization issues . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 322
10.6.1 Standardization bodies and groups . . . . . . . . . . . . . . . . . 323
10.6.2 European Conference of Postal and
Telecommunications Administrations . . . . . . . . . . . . . . . 323
10.6.3 ETSI . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 323
10.6.4 DVB . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 326
10.6.5 International Telecommunication Union . . . . . . . . . . . . . 330
10.7 Conclusions . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 330
References . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 333
Index . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 335
List of Contributors
Rafael Asorey Cacheda Nedo Celandroni

UVI - Universidad de Vigo, CNR-ISTI - National Research
Dep. Ingeniera Telematica, ETSI Council (CNR), ISTI
Telecomunicacion, Campus, 36200 Institute, Via G. Moruzzi, 1,
Vigo, Spain San Cataldo, 56124 Pisa, Italy
rasorey@det.uvigo.es nedo.celandroni@isti.cnr.it
Kostantinos Avgeropoulos
AUTh - Aristotle University of Wei Koong Chai
Thessaloniki, Thessaloniki, UniS - University of Surrey, CCSR,
Panepistimioupolis, 54124, Greece Centre for Communication Systems
k.avgeropoulos@gmail.com Research, Guildford,
Surrey GU2 7XH, UK
Paolo Barsocchi W.Chai@surrey.ac.uk
CNR-ISTI - National Research
Council (CNR), ISTI
Institute, Via G. Moruzzi, 1, Paolo Chini
San Cataldo, 56124 Pisa, Italy CNIT - University of Siena
paolo.barsocchi@isti.cnr.it Research Unit, Via Roma, 56,
53100, Siena, Italy
Ulla Birnbacher chini7@unisi.it
TUG - Graz University of
Technology, Inst. Comm. Net. and
Satellite Comm., Ineldgasse 12, Antonio Cuevas
A-8010 Graz, Austria UC3M - Universidad Carlos III de
ulla.birnbacher@tugraz.at Madrid,
Avda. Universidad 30, 28911
Daniel Castro Garca Leganes, Spain
INFOGLOBAL, Spain acuevas@it.uc3m.es
xx List of Contributors
Franco Davoli Javier Herrero S

anchez
CNIT - University of Genoa INFOGLOBAL, Spain
Research Unit, Via Opera Pia, 13,
16145, Genova, Italy Du Hongfei
franco.davoli@cnit.it UniS - University of Surrey, CCSR,
Centre for Communication Systems
Gorry Fairhurst Research, Guildford,
UoA - University of Aberdeen, Surrey GU2 7XH, UK
Department of Engineering, H.Du@surrey.ac.uk
Fraser Noble Building, Stylianos Karapantazis
Aberdeen AB24 3UE, UK AUTh - Aristotle University of
gorry@erg.abdn.ac.uk Thessaloniki, Thessaloniki,
Panepistimioupolis, 54124, Greece
Erina Ferro
skarap@auth.gr
Council (CNR), ISTI Georgios Koltsidas
Institute, Via G. Moruzzi, 1, AUTh - Aristotle University of
San Cataldo, 56124 Pisa, Italy Thessaloniki, Thessaloniki,
erina.ferro@isti.cnr.it Panepistimioupolis, 54124, Greece
fractgkb@auth.gr
Giovanni Giambene
CNIT - University of Siena Victor Y. H. Kueh
Research Unit, Via Roma, 56, UniS - University of Surrey, CCSR,
53100, Siena, Italy Centre for Communication Systems
giambene@unisi.it Research, Guildford,
Surrey GU2 7XH, UK
Samuele Giannetti victor unis@yahoo.co.uk
CNIT - University of Siena
Michele Luglio
Research Unit, Via Roma, 56,
UToV - University of Rome Tor
53100, Siena, Italy
Vergata,
giannetti13@unisi.it
Via del Politecnico, 1,
Francisco Javier Gonz alez 00133 - Roma, Italy
Castano luglio@uniroma2.it
UVI - Universidad de Vigo, Vincenzo Mancuso
Dep. Ingeniera Telematica, ETSI UToV - University of Rome Tor
Telecomunicacion, Campus, 36200 Vergata,
Vigo, Spain Via del Politecnico, 1,
javier@det.uvigo.es 00133 - Roma, Italy
vincenzo.mancuso@ieee.org
Alberto Gotta
CNR-ISTI - National Research Mario Marchese
Council (CNR), ISTI CNIT - University of Genoa
Institute, Via G. Moruzzi, 1, Research Unit, Via Opera Pia, 13,
San Cataldo, 56124 Pisa, Italy 16145, Genova, Italy
alberto.gotta@isti.cnr.it Mario.Marchese@unige.it
List of Contributors xxi
Giada Mennuti Veronica Pasqualetti

CNIT - University of Florence CNIT - University of Siena
Research Unit, Via di S. Marta, 3, Research Unit, Via Roma, 56,
50139, Firenze, Italy 53100, Siena, Italy
giada@lenst.det.unifi.it pasqualetti@unisi.it
Maurizio Mongelli
CNIT - University of Genoa Tommaso Pecorella
Research Unit, Via Opera Pia, 13, CNIT - University of Florence
16145, Genova, Italy Research Unit, Via di S. Marta, 3,
Maurizio.Mongelli@unige.it 50139, Firenze, Italy
Antoni Morell pecos@lart.det.unifi.it
UAB - Universitat Aut onoma de
Barcelona,
Francesco Potort`
Dpt. Telecommunications and
Systems Engineering,
Council (CNR), ISTI
Engineering School,
Bellaterra 08193 - Barcelona, Spain Institute, Via G. Moruzzi, 1,
antoni.morell@uab.es San Cataldo, 56124 Pisa, Italy
Potorti@isti.cnr.it
Jos
e Ignacio Moreno Novella
UC3M - Universidad Carlos III de
Madrid, Cesare Roseti
Avda. Universidad 30, 28911 UToV - University of Rome Tor
Leganes, Spain Vergata,
jmoreno@it.uc3m.es Via del Politecnico, 1,
00133 - Roma, Italy
Seounghoon Oh
roseti@ing.uniroma2.it
RWTH - Rheinisch-Westfalische
Technische Hochschule
Aachen / Aduwati Sali
COMNETS, Kopernikusstr. 16, UniS - University of Surrey, CCSR,
D-52074 AACHEN, Germany Centre for Communication Systems
oh@comnets.rwth-aachen.de Research, Guildford,
Antonio Pant` o Surrey GU2 7XH, UK
CNIT - University of Catania A.Sali@surrey.ac.uk
Research Unit, Viale A. Doria, 6,
95125, Catania, Italy Gonzalo Seco Granados
antonio.panto@cnit.it UAB - Universitat Autonoma de
Cristina P arraga Niebla Barcelona,
DLR - German Aerospace Center, Dpt. Telecommunications and
Insitute of Comms. and Systems Engineering,
Navigation, Oberpfaenhofen, 82234 Engineering School,
Wessling, Germany Bellaterra 08193 - Barcelona, Spain
cristina.parraga@dlr.de gonzalo.seco@uab.es
xxii List of Contributors
Petia Todorova
Mara Angeles Vazquez Castro
FhI - Fraunhofer Institute for Open UAB - Universitat Autonoma de
Communication Systems - FOKUS, Barcelona,
Kaiserin - Augusta - Alee 31, 10589 Dpt. Telecommunications and
Berlin, Germany Systems Engineering,
Petia.Todorova@ Engineering School,
fokus.fraunhofer.de Bellaterra 08193 - Barcelona, Spain
Orestis Tsigkas angeles.vazquez@uab.es
AUTh - Aristotle University of
Thessaloniki, Thessaloniki,
Panepistimioupolis, 54124, Greece Fausto Vieira
torestis@auth.gr UAB - Universitat Autonoma de
Alessandro Vanelli-Coralli Barcelona,
UoB - University of Bologna Dpt. Telecommunications and
DEIS/ARCES, Systems Engineering,
Viale Risorgimento, 2, Engineering School,
40136 - Bologna, Italy Bellaterra 08193 - Barcelona, Spain
avanelli@deis.unibo.it fvieira@sunaut.uab.es
List of Acronyms and Abbreviations
3G 3rd Generation B-ISDN Broadband Integrated Services

3GPP 3rd Generation Partnership Digital Network
Project BLER Block Error Rate
4G 4th Generation BM-SC Broadcast-Multicast Service
AAA Authentication, Authorization Center
and Accounting BO Bandwidth Occupation
ABC Always Best Connected BoD Bandwidth on Demand
ABR Available Bit Rate BPM BSM Protocol Manager
AC Adaptive Coding BPSK Binary Phase Shift Keying
ACK Acknowledgement BS Base Station
ACM Adaptive Coding and BSA Broadband Satellite Access
Modulation BSM Broadband Satellite Multimedia
ADSL Asymmetric Digital Subscriber BSM ID BSM Identier
Line BSS Broadcasting Satellite Service
AF Assured Forwarding BTP Burst Time Plan
AICH Acquisition Indicator Channel CA Congestion Avoidance
AIMD Additive Increase Multiplicative CAC Call Admission Control
Decrease CBP Call Blocking Probability
AP Access Point CBQ Class-Based Queuing
API Application Programming CBR Constant Bit Rate
Interface CCM Constant Coding Modulation
APP Application layer CDM Code Division Multiplexing
APSK Amplitude and Phase Shift CDMA Code Division Multiple Access
Keying CDMA/HDR CDMA/High Data Rate
AQM Active Queue Management CDP Call Dropping Probability
AR Access Router CDVT Cell Delay Variation Tolerance
ARP Address Resolution Protocol CEN European Committee for
ARQ Automatic Repeat reQuest Standardization
ASC Access Service Class CENELEC European Committee for
ASD Aggregated System Demand Electro-technical
ATM Asynchronous Transfer Mode Standardization
AVBDC Absolute Volume Based CEPT European Conference of Postal
Dynamic Capacity and Telecommunications
AWGN Additive White Gaussian Noise Administrations
BCH Bose-Chaudhuri-Hocquenghem CF/DAMA Combined Free/Demand
(in Chapter 1) Assignment Multiple Access
BCH Broadcast Channel C/I Carrier-to-Interference ratio
(in Chapter 5) CIF-Q Channel Condition -
BDP Bandwidth-Delay Product Independent Fair Queuing
BE Best Eort C/I PS C/I Proportional Scheduler
BER Bit Error Rate CIST Common Internal Spanning
BGAN Broadband Global Area Tree
Network CLR Cell Loss Ratio
BGAN-X BGAN Extension project CM Control Module
xxiv Acronyms
CMF Control and Monitoring DVB-RCT DVB-Return Channel via

Functions Terrestrial
C/N Carrier power-to-Noise power DVB-S Digital Video Broadcasting via
ratio Satellite
CN Core Network DVB-S2 DVB-Satellite version 2
COPS Common Open Policy Service DVB-T DVB-Terrestrial
COST Co-operation in the eld of DVB-TM DVB-Technical Module
Scientic and Technical EBU European Broadcasting Union
Research ECC Electronic Communications
CP Complete Partitioning Committee
CQI Channel Quality Indicator ECN Explicit Congestion
CR Capacity Request Notication
CRA Continuous Rate Assignment ECSS European Co-operation on
CRC Cyclic Redundancy Check Space Standardization
CS Complete Sharing EDF Earliest Deadline First
C-SAP Control-SAP EF Expedited Forwarding
CSI Channel State Information EHF Extremely High Frequency
cwnd congestion window EIRP Eective Isotropic Radiated
DAMA Demand Assignment Multiple Power
Access EMC ElectroMagnetic Compatibility
DBA Dynamic Bandwidth Allocation EqB Equivalent Bandwidth
DBAC Dynamic Bandwidth Allocation ERA European Research Area
Capabilities ERM EMC and Radio spectrum
DBRA Dynamic Bandwidth and Matters
Resource Allocation ESA European Space Agency
DBS Direct Broadcast Satellite ETSI European Telecommunications
DBS-RCS DBS with Return Channel Standards Institute
System EU European Union
DCA Dynamic Channel FA Fixed Assignment
(or Capacity) Allocation FACH Forward Access Channel
DCCH Dedicated Control Channel FC FIFO Maximum Capacity
DCH Dedicated Channel FCA Free Capacity Assignment
DDP Delay Dierentiation (in Chapters 1, 7, 8 and 9)
Parameter FCA Fixed Channel Allocation
DDQ Delay Dierentiation Queuing (in Chapter 2)
DiServ Dierentiated Service FCFS First Come First Served
DLL Data Link Layer FCT Frame Composition Table
DMBS Double-Movable Boundary FDD Frequency Division Duplexing
Strategy FDM Frequency Division
DOCSIS-S Data Over Cable Service Multiplexing
Interface Specication for FDMA Frequency Division Multiple
Satellite Access
DP Dierentiation Parameter FEC Forward Error Correction
DPSK Dierential Phase Shift Keying FER Frame Erasure Rates
DRA Dynamic Resource Allocation (in Chapter 3)
DRT Delayed Real-Time FER Frame Error Rate
DS Direct Sequence (in Chapter 5)
DSCH Downlink Shared Channel FH Frequency Hopping
DSCP DiServ Code Point FHO Fast HandOver
DSNG Digital Satellite News FI Fairness Index
Gathering F id frame ID
DTCH Dedicated Trac Channel FIFO First In First Out
D-TDMA Dynamic TDMA FL1-HARQ Fast L1 hybrid ARQ
DTH Direct-To-Home FMT Fade Mitigation Techniques
DULM Data Unit Labeling Method F nb frame number
dupACKs duplicate ACKs FP Framework Programme
DVB Digital Video Broadcasting FSK Frequency Shift Keying
DVB-C DVB-Cable FSS Fixed Satellite Service
DVB-CAS DVB-Conditional Access FTP File Transfer Protocol
System FZC Forward Erasure Correction
DVB-GBS DVB-Global Broadcast Service GB Guaranteed Bandwidth
DVB-H DVB-Handheld GEO Geosynchronous
DVB-RCC DVB-Return Channel via Cable (Geostationary) Earth Orbit
DVB-RCL DVB-Return Channel GM Guaranteed Minimum
for LMDS GOPs Group of Pictures
DVB-RCS DVB-Return Channel via GoS Grade of Service
Satellite GPRS General Packet Radio Service
Acronyms xxv
GPS Generalized Processor Sharing LEO Low Earth Orbit

GSM Global System for Mobile LLC Logical Link Control
Communications LLC/SNAP LLC/Sub-Network Access
GW Gateway or Trac Gateway Protocol
HCA Hybrid Channel Allocation LMDS Local Multipoint Distribution
HDTV High Denition Television System
HLS Hierarchical Link Sharing LoS Line of Sight
HNS Hughes Network Systems LP Low-Priority
HP High-Priority LRD Long Range Dependent
HPA High Power Amplier LSP Label Switched Path
HPD Hybrid Proportional Delay LSR Label Switching Router
HSDPA High Speed Downlink Packet LTFS Long-Term Fairness Server
Access LUI Last Useful Instant
HS-DPCCH High Speed Dedicated Physical MAC Medium Access Control
Control Channel MAC-hs MAC/HS-DSCH
HS-DSCH High Speed-DSCH MAN Metropolitan Area Network
HS-PDSCH High Speed Physical Downlink MBMS Multimedia Broadcast
Shared Channel Multicast Services
HTML HyperText Mark-Up Language MBU Minimum Bandwidth Unit
IAB Internet Architecture Board MCS Master Control Station
IBR Information Bit Rate MEO Medium Earth Orbit
ICMP Internet Control Message MF Multi-Frequency
Protocol MF-TDMA Multi Frequency -
IEEE Institute of Electrical and Time Division Multiple Access
Electronics Engineers MLI Maximum Legal Increment
IETF Internet Engineering Task MLPQ Multi-Level Priority Queuing
Force MMPP Markov-Modulated Poisson
IFR Increasing Failure Rate Processes
IM Inter-Modulation MMS Multimedia Messaging Service
IMT International Mobile MN Mobile Node
Telecommunications MODCOD Modulation and Coding
IntServ Integrated Service MOS Mean Opinion Score
IP Internet Protocol MPE Multi Protocol Encapsulation
IPA Innitesimal Perturbation MPEG Moving Picture Experts Group
Analysis MPEG2-TS Moving Picture Experts Group
IP-CAS IP-based Conditional Access 2 - Transport Stream
System MPLS Multiprotocol Label Switching
IPoS Internet Protocol over Satellite M-SAP Management-SAP
ISDN Integrated Services Digital MSL Minimum Scheduling Latency
Network MSS Maximum Segment Size
ISI Integral Satcom Initiative MSTP Multiple STP
ISLs Inter-Satellite Links MTs Multicast Terminals
ISN Interactive Satellite Network MTCH MBMS point-to-multipoint
ISO/OSI International Standard Trac Channel
Organization/Open System MTU Maximum Transfer Unit
Interconnection NBS Nash Bargaining Solution
ISP Internet Service Provider NCC Network Control Center
IST Information Society NCR Network Clock Reference
Technologies ND Neighbor Discovery
ITU International NGN Next-Generation Network
Telecommunication Union NoE Network of Excellence
ITU-D ITU - Telecommunication nrt-VBR non-real-time-VBR
Development sector OBP On-Board Processor
ITU-R ITU - Radiocommunication OC Optimized Centralized
sector OFDM Orthogonal Frequency Division
ITU-T ITU - Telecommunication Multiplex
sector OP Optimized Proportional
IWFQ Idealized Wireless Fair Queuing PAB Proportional Allocation of
IWU Inter-Working Unit Bandwidth
KKT Karush-Kuhn-Tucker PCPCH Physical Common Packet
L1 Layer 1 (physical layer) Channel
L2 Layer 2 (link/MAC layer) pdf probability density function
L3 Layer 3 (network layer) PDS Proportional Dierentiated
LAN Local Area Network Service
LC LUI Maximum Capacity PDU Protocol Data Unit
LDP Label Distribution Protocol P-EDF Prioritized-EDF
LDPC Low Density Parity Check PEP Performance Enhancing Proxy
xxvi Acronyms
PER Packet Error Rate SCr Service Credit

PF Proportional Fair SD Satellite-Dependent
PG Processing Gain SDMA Spatial Division Multiple
PHB Per-Hop Behavior Access
PHY Physical layer S-DMB Satellite Digital Multimedia
PLFRAME Physical Layer Frame Broadcasting
PLP Packet Loss Probability SDR Satellite Digital Radio
PLR Packet Loss Rate SDTV Standard Denition Television
PMPP Pareto-Modulated Poisson SF Spreading Factor
Processes SFM Stochastic Fluid Models
PN Pseudo Noise S-HSDPA HSDPA via Satellite
POTS Plain Old Telephone Service SI Satellite-Independent
PRACH Physical Random Access SIR Signal-to-Interference Ratio
Channel SI-SAP Satellite-Independent -
PRC Power Ramping Control Service Access Point
PRMA Packet Reservation Multiple SL Super-frame Length
Access SLA Service Level Agreement
PRMA-HS PRMA with Hindering States S-MBMS Satellite MBMS
PSK Phase Shift Keying SMEs Small and Medium Enterprises
PSNR Peak Signal to Noise Ratio SMG Special Mobile Group
PSTN Public Switched Telephone SMS Short Message Service
Network SNIR Signal to Noise and
QAM Quadrature Amplitude Interference Ratio
Modulation SOHO Small Oce - Home Oce
QID Queuing Identier SP Simple Proportional
QoS Quality of Service SPC Smith Predictor Controller
QoSMO QoS Mapping Optimization SR Slot Request
QPSK Quadrature Phase Shift Keying SRD Short Range Dependent
RA Random Access SS Slow Start
RAB Radio Access Bearer ssthresh slow start threshold
RACH Random Access Channel ST Satellite (interactive) Terminal
RAN Radio Access Network STB Set-Top-Box
RAT Robust Audio Tool STFQ Stochastic Fairness Queuing
RB Reserved Bandwidth STP Spanning Tree Protocol
RBDC Rate Based Dynamic Capacity S-UMTS Satellite-UMTS
RCBC Reference Chaser Bandwidth SWTP Satellite Waiting Time
Controller Priority
RC-PSTN Return Channel - PSTN TB Transport Block
RCQI Relative Channel Quality Index TBTP Terminal Burst Time Plan
RCS Return Channel via Satellite TC Transported Capacity
RCST Return Channel Satellite TCA Trac Conditioning
Terminal Agreement
RED Random Early Detection TCP Transmission Control Protocol
RF Radio Frequency TC-SES Technical Committee for
RHC Receding Horizon Controller Satellite Earth Stations and
RLC Radio Link Control Systems
RNC Radio Network Controller TCT Time Composition Table
RRM Radio Resource Management TDM Time Division Multiplexing
RSP Recovery Service Provider TDMA Time Division Multiple Access
RSTP Rapid STP TE Terminal Equipment
RSVP Resource Reservation Protocol Telnet TELetype NETwork
RT Real Time TF Transport Format
RTD Round Trip propagation Delay TFC Transport Format Combination
RTO Retransmission TimeOut TFCI Transport Format Combination
RTP Real-time Transport Protocol Indication
RTT Round Trip Time TFCS Transport Format Combination
rt-VBR real-time-VBR Set
SAC Satellite Access Control TFRC Transport Format and
S-ALOHA Slotted-ALOHA Resource Combination
SBFA Server-Based Fairness Approach TIST Telecommunications,
S-CCPCH Secondary Common Control Information Science and
Physical Channel Technology
SCED Service Curve-based Earliest TM Transmission & Multiplexing
Deadline rst TOS Type Of Service
SCPC Single Carrier Per Channel TR Trunk Reservation
SCPS-TP Space Communications Protocol TS Time Slot
Specication-Transport Protocol TS nb timeslot number
Acronyms xxvii
TTI Transmission Time Interval VPI/VCI Virtual Path Identier/

T-UMTS Terrestrial UMTS Virtual Channel Identier
TWTA Traveling-Wave-Tube VPN Virtual Private Network
Amplier VQMP Peak Video Quality
UBR Unspecied Bit Rate Measurement
UDP User Datagram Protocol VR-JT Variable Rate - Jitter Tolerant
UE User Equipment VR-RT Variable Rate - Real Time
UL Upper Limit VSAT Very Small Aperture Terminal
UMTS Universal Mobile VSF Variable Spreading Factor
Telecommunications System WAN Wide Area Network
UPC Usage Parameter Control W-CDMA Wideband Code Division
URAN UMTS Radio Access Network Multiple Access
U-SAP User-SAP WCI Wireless Channel Information
UT User Terminal WFBoD Weighted Fair Bandwidth-on-
VBDC Volume Based Dynamic Demand
Capacity WFQ Weighted Fair Queuing
VBR Variable Bit Rate WiFi Wireless Fidelity
VC Virtual Channel WiMAX Worldwide Interoperability for
VCM Variable Coding and Microwave Access
Modulation WLAN Wireless LAN
VLAN Virtual Local Area Networks WP Work Package
VLL Virtual Leased Line WRR Weighted Round Robin
VoIP Voice over IP XTP eXpress Transfer Protocol
VP Virtual Partitioning
Part I
Resource Management Framework for Satellite

Communications
1
INTRODUCTION TO SATELLITE
COMMUNICATIONS AND RESOURCE
MANAGEMENT
Editor: Giovanni Giambene1
Contributors: Paolo Chini1 , Giovanni Giambene1
1
CNIT - University of Siena, Italy
1.1 Satellite communications

Multimedia communications have been widely supported by terrestrial infras-
tructures that employ optical bers in backbone links to achieve huge capacity.
A technological alternative is represented by the use of satellites for providing
multimedia broadband services to xed and mobile users in several scenarios
where terrestrial networks cannot be used or are congested.
Today, still a large number of persons living in remote areas or in
underdeveloped regions do not have a realistic perspective of achieving access
to high-speed Internet for many years. This problem constitutes a serious
obstacle to making the benets of the Information Society available to all.
Such digital divide problem can be solved by satellite communications that
can easily reach the dierent regions on the Earth by providing everywhere
the same service types. Satellites are an important delivery platform of
information society services, such as interactive TV and mobile, high-speed
Internet access.
The most important reasons for the diusion of satellite communications
can be summarized as follows [1]:
Ubiquitous coverage: a single satellite can reach every potential user across
an entire continent. This is a very signicant feature, especially in low
population density areas or over the see, where the realization of terrestrial
infrastructures would be not viable.
4 Giovanni Giambene
Support to mobile users: a mobile user, which is situated in the satellite

coverage area, can easily communicate with other xed or mobile users.
Reduced cost: with satellite communications, cost is independent of the
distance. Moreover, satellite networks can easily cover a great part of the
Earth, thus reaching a very big potential market of customers. This is an
important opportunity in order to provide services at aordable costs.
Variety of connectivity: it is possible to provide, in a simple and economic
way, point-to-multipoint and broadcast communications, without complex
multicast routing protocols (used in meshed terrestrial networks).
Rapid deployment and easy management of the network : once a satellite is
launched it can immediately reach a high number of users. With satellites,
multimedia services can be provided to a wide multitude of users on broad
areas in a quicker way than using a terrestrial infrastructure.
Bandwidth exibility: it is possible to provide simplex, duplex, narrow-
band, symmetric and asymmetric bandwidth. Moreover, satellites can
allow a broadband access to end-users, thus representing a possible solution
to the last mile problem.
Very good books in the eld of satellite communications, providing excel-

lent basis on this eld are detailed in references [2]-[7].
Satellites are situated on suitable orbits around the Earth; on the basis of
their altitude, they can be classied into three main categories [1] (see Figure
1.1):
Low Earth Orbit (LEO) satellites at a height between 500 and 2,000 km
of altitude, i.e., below the Van Allen radiation belts. The Earth rotation
period is about 100 minutes and the satellite visibility time is around 15
minutes. These orbits can be polar or inclined.
Medium Earth Orbit (MEO) may be circular or elliptical in shape at a
height between 8,000 and 12,000 km of altitude (between the two Van
Allen radiation belts). The rotation period is 5-12 hours and the satellite
visibility time is 2-4 hours.
Geosynchronous Earth Orbit (GEO) is on the Earths equatorial plane
at a height of about 35,780 km with a rotation period of 24 hours and
a satellite visibility time of 24 hours. Many GEO satellites are allocated
on distinct slots on the equatorial plane orbit. The GEO satellite altitude
and the equatorial orbit have been determined to allow that GEO satellites
rotate at the same speed of the Earth. Hence, a GEO satellite remains in
a stationary position in the sky with respect to a xed point on the Earth;
this is a desired feature for telecommunication purposes.
The balance between the gravity force versus the Earth and the centrifugal
one determines the satellite orbital speed. The three Keplers laws regulate
the satellite orbital motion.
A satellite communication system is formed by a number of satellites,
typically with the same orbit type (i.e., GEO, MEO or LEO) that cover a
Chapter 1: INTRODUCTION TO SATELLITE COMMUNICATIONS 5
Fig. 1.1: Description of satellite orbit types.
region or the whole Earth, thus forming a constellation.

Three GEO satellites are sucient to cover all the Earth, excluding
Polar Regions. GEO satellites are well suited for global-coverage broad-
cast/multicast services and also for regional mobile and xed communication
services. MEO and LEO satellites are non-stationary with respect to a user on
the Earth; hence, dierent satellites alternatively provide telecommunication
service coverage to a given area on the Earth. A global MEO system needs a
constellation of 10-12 satellites to assure a minimum elevation angle greater
than 30 . LEO systems are characterized by constellations of more than 40
satellites with minimum elevation angle from 10 to 40 . A minimum elevation
angle of about 40 (30 ) is recommended in the MEO (LEO) case in order
to have high link availability and acceptable delay variations. Moreover, LEO
and MEO satellite systems allow lower propagation delays and hence, lower
end-to-end latency in transferring data than GEO satellites.
GEO satellites are very big and can host a huge payload; high power
and large antennas are needed to assure a reliable link with Earth stations.
MEO satellites are smaller than GEO ones, so that launching operations are
less expensive. Finally, LEO satellites are smaller and less expensive to build
and to launch than GEO and MEO. Launchers allowing the transport of
multiple satellites permit to reduce the cost to have an operational LEO
satellite constellation.
The coverage area (footprint) of a satellite is divided into many cells (each
irradiated by an antenna spot-beam) in order to concentrate the energy on a
small area. Thus, it is also possible to shape the area served by a satellite on the
Earth. Moreover, multi-spot-beam coverage permits remarkable advantages,
like an ecient distribution of resources (e.g., reusing the same frequency)
or a lower cost of the Earth terminal equipment (e.g., antennas with small
6 Giovanni Giambene
size, since narrower surfaces are irradiated on the Earth, thus having a higher
power per surface unit).
Frequency bands (of interest for satellite communications) and related
designations are listed below [1],[3],[5]:
L band from 1 to 2 GHz

S band from 2 to 4 GHz
C band from 4 to 8 GHz
X band from 8 to 12 GHz
Ku band from 12 to 18 GHz
K band from 18 to 26 GHz
Ka band from 26 to 40 GHz
V band from 40 to 75 GHz.
These bands, composing the microwave spectrum, are actively used in

commercial and military satellite communications. The typical frequency band
allocations for satellite communications, adopted for dierent services, are
detailed below considering uplink/downlink cases:
Fixed Satellite Service (FSS): 6/4 GHz (C band), 8/7 GHz (X band),
14/12-11 GHz (Ku band), 30/20 GHz (Ka band), 50/40 GHz (V band).
These services concern communications with xed terrestrial terminals;
moreover, they are often broadband (typically in the range of 1-200
Mbit/s) due to both the available Radio Frequency (RF) bandwidth and
suitable link performance by using terrestrial xed directional antennas.
Even if these services have been originally allocated to GEO satellites, also
non-GEO system allocations are possible.
Broadcasting Satellite Service (BSS): 2/2.2 GHz (S band), 12 GHz (Ku
band), 2.6/2.5 GHz (S band). These services deal with direct broadband
broadcast transmissions through public operators. In particular, the Ku
band segment of BSS has been reserved for orbit positioning and dedicated
channels for individual nations employment. This service has been mainly
allocated to GEO satellites, but, like in the FSS case, also non-GEO
satellites are possible.
Mobile Satellite Service: 1.6/1.5 GHz (L band), 30/20 GHz (Ka band).
These services are related to communications with mobile Earth stations
(e.g., ships, vehicles, aircrafts, and also persons). An example of mobile
satellite service is the Inmarsat system, operating in the L band with
GEO satellites for land-mobile services. These bands have been assigned
later also to non-GEO satellite networks.
Note that L, S and C bands are already congested; X band is typically

reserved for government use (military xed communications); Ku band is
used by the majority of satellite digital broadcast systems as well as for
current Internet access systems. Finally, Ka band allows higher bandwidths
with smaller antennas (with respect to Ku band), but presents the problem
of signicant signal impairment in the presence of bad wheatear conditions

(e.g., rain).
A transponder is a receiver-transmitter unit on a communication satellite.
It receives a signal from the Earth (uplink), manages it and retransmits it
back to Earth at a dierent frequency (downlink). A satellite has several
transponders in its payload. Two dierent types of transponders can be
distinguished as follows:
Bent-pipe transponder (i.e., the transponder acts as a simple repeater).
On board, the signal is simply amplied and retransmitted, but there is
no improvement in the signal-to-noise ratio since also background noise is
amplied.
Regenerating transponder : a transponder demodulates and decodes the
received signal, thus performing signal recovery before retransmitting it.
Since at some point base-band signals are available, other activities are
also possible, such as routing and beam-switching (in case of multi-beam
satellite antenna). Satellites with regenerating transponders and on board
processing capabilities can also employ Inter-Satellite Links (ISLs) with
other satellites of the same constellation, thus permitting the routing of
the signal in the sky.
It is important to provide here some interesting data for current state-of-
the-art GEO satellites.
The Astra 1H satellite has 32 transponders with 24/32 MHz bandwidth
(total bandwidth of 1 GHz). Each transponder has a trac capacity of
25-30 Mbit/s.
The AmerHis satellite (51 transponders) has a hybrid payload with 4
channels, each with 36 MHz for a total capacity of 174 Mbit/s. Moreover,
there is a DVB-RCS transponder that can manage up to 64 carriers, each
with 0.5 Mbit/s and a DVB-S transponder with a capacity of 54 Mbit/s;
see the following Section 1.4 for more details on DVB-RCS and DVB-S
systems.
Tables 1.1 and 1.2 below provide a survey of some satellite communication
systems that are currently operational or planned [8],[9]; for the denition of
the dierent access techniques, please refer to the following Section 1.3.
A typical satellite network architecture is shown in Figure 1.2, where we
can see the Earth station permitting the interconnection via a gateway to the
terrestrial core network.
Satellite communications are broadcast in nature. Hence, satellites do
not oer an adequate reliability from the security and privacy standpoint.
Practically, it is possible that a malicious user can hear what the others are
communicating. Therefore, it is necessary to adopt appropriate cryptography
algorithms to control network accesses and to protect transmissions.
Recently, the Broadband Global Area Network (BGAN) system has ac-
quired momentum to provide several services via Inmarsat-4 satellites (e.g.,
8 Giovanni Giambene
Fig. 1.2: Basic satellite network architecture.
System Orbit type, Services Access Frequency

altitude [km] scheme bands
GlobalStar 48 LEO, 1414 Mobile satellite system Combined Uplink:
voice and data services FDMA & 1610.0-1626.5
CDMA MHz (L band)
(uplink and Downlink:
downlink) 2483.5-2500 MHz
(S band)
Iridium 66 LEO, 780 Mobile satellite system FDMA/ Uplink:
voice and data services TDMA - 1616-1626.5
TDD for MHz (L Band)
both uplink Downlink:
and 1610-1626.5 MHz
downlink (L Band)
ICO 12 MEO ICO is planning a family FDMA/ Uplink:
(new ICO) (10 active), of quality voice, wireless TDMA - 1980-2010 MHz
10355 (changed Internet and other FDD Downlink:
to 10390 km, packet-data services 2170-2200 MHz
late 1998) (C/S bands)
Table 1.1: Description of the characteristics of the main satellite communication

systems (operational or planned) for non-GEO orbits.
telephony and ISDN calls; Internet/Intranet connection; SMS and MMS;

UMTS location-based services like information on maps or local travel in-
formation), rstly to xed terrestrial user terminals, and secondly to mobile
terminals on planes, ships or land areas. BGAN satellites operate in the L
band. It is possible to adapt the transmission power, bandwidth, coding rate
and modulation scheme to terminal capabilities and to channel conditions,
in order to achieve high transmission eciency and exibility. The baseline
system allows communications from 4.5 to about 512 kbit/s to 3 classes
System Orbit type, Services Access Frequency

altitude [km] scheme bands
Spaceway 16 GEO With Spaceway, large Uplink: Uplink:
+ 20 MEO, businesses, telecommuters, FDMA/ 27.5-30 GHz
36000 - 10352 Small Oce - Home TDMA Downlink:
Oce (SOHO) users and Downlink: 17.7-20.2 GHz
consumers will have access TDMA Ka band
to two-way, high-data-rate
applications such as
desktop videoconferencing,
interactive distance
learning and Internet
services
Thuraya 2 GEO Voice telephony, fax, FDMA Uplink:
data, short messaging, 1626.5-1660.5 MHz
location determination, Downlink:
emergency services, high 1525-1559 MHz
power alerting L/C bands
Eutelsat GEO satellites Single digital TV Uplink: Uplink: 13.75, 14-
(operator) (e.g., Hotbird 4, programme broadcasting, DVB-RCS 14.50, 29.50-30 GHz
Hotbird 6) digital radio broadcasting, (TDMA) Downlink: 10.70, 10.86-
equipped with interactive multimedia Downlink: 12.75, 19.70-20.20 GHz
the Skyplex services and Internet DVB-S Ku and Ka band
regenerating connectivity
transponder
Wildblue GEO High-speed broadband Uplink: Uplink: 5.9-6.4 GHz
(Anik F2) Internet access, satellite TDMA (C band), 14-14.5 GHz
television, distance Downlink: (Ku band), 28.35-28.6
learning and telemedicine MF-TDMA and 29.25-30 GHz
(Ka band)
Downlink: 3.7-4.2
(C band), 11.7-12.2
(Ku band), 18.3-18.8
and 19.7-20.2 GHz
(Ka band)
IPStar GEO Broadband access, Uplink: Uplink: 13.775-13.975,
Intranet and VPN, MF-TDMA 14-14.5 GHz
Broadcast/Multicast, Downlink: Downlink: 10.95-11.2,
Video on Demand, Voice, TDM/ 11.5-11.7, 12.2-12.75
Leased Circuit/Trunking, OFDM GHz
Video Conferencing
Inmarsat 11 GEO Simultaneous voice & TDMA Uplink: 1.626-1.66,
(10 active sats.): data, Internet & Intranet 1.98-2.025 GHz
4 Inmarsat-2, content and solutions, Downlink: 1.525-1.559,
5 Inmarsat-3, Video-on-demand, 2.16-2.22 GHz
2 Inmarsat-4 videoconferencing, fax,
e-mail, phone and LAN
access
Table 1.2: Description of the characteristics of the main satellite communication

systems (operational or planned) for GEO orbits.
of portable terminals. The enhanced system (BGAN-X, BGAN Extension

project) has been developed to serve omni-directional and directional mobile
terminals, extending the classes from 3 to 11.
10 Giovanni Giambene
1.2 Basic issues in the design of satellite communication

systems
Satellite communications represent an attractive solution to provide broad-
band and multimedia services. To make the upcoming satellite network
systems fully realizable, meeting new services and application Quality of
Service (QoS) requirements, many technical challenges have to be addressed
as described below [1]-[5].
Round Trip propagation Delay (RTD)
RTD is the propagation delay along a link (back and forth). In the satellite
case, its value depends on the satellite orbit, the relative position of the user
on the Earth, and the type of satellite [1],[3],[5]. In particular, if the satellite
is regenerating, RTD involves a single hop from the Earth to the satellite and
back to the Earth; whereas, if the satellite is bent-pipe, RTD typically involves
a double hop (from Earth to satellite to Earth and back) since layer 2 control
functions are in the Earth station. In case of GEO regenerating satellites, RTD
varies in the range 239-280 ms. In particular, RTD is 239.6 ms for an Earth
station placed on the Earth equator in the point below the satellite; whereas,
RTD is about 280 ms for an Earth station placed at the edge of the satellite
coverage area (i.e., seeing the satellite with the minimum allowed elevation
angle). Note that RTD can be also referred to an end-to-end connection,
involving many links (the satellite type is not relevant for such RTD). In the
GEO case, this end-to-end RTD value (between a message transmission and
the reception of the relative reply) varies from 480 to 558 ms; this value can
increase due to processing, queuing and on-board switching operations.
The RTD values increase with the satellite orbit altitude and reduces
with the elevation angle. LEO and MEO satellites are situated at low
altitudes, so they allow lower RTD values than GEO. High RTD values
cause several problems for both interactive and real-time applications (e.g., an
evident and troublesome echo in phone calls); moreover, also reliable transport
layer protocols can experience problems since the end-to-end delay loop is
dominated by the propagation delay contribution due to the satellite segment.
The maximum RTD value (RT Dmax ) for a given satellite constellation also
depends on the minimum elevation angle (mask angle), i.e., the elevation angle
at the edge of coverage. The RT Dmax characteristics for LEO satellite systems
are described in Figure 1.3.
Atmospheric eects
The eects of atmosphere (subdivided in troposphere and ionosphere) can be

summarized as follows [2]:
Fig. 1.3: RT Dmax level curves in ms for LEO satellite constellations in the plane
Minimum elevation angle [in degrees] versus LEO satellite constellation altitude [in
km].
Atmospheric gasses. Oxygen (dry air) and water vapor determine an

attenuation of the electromagnetic signal that depends on the transmission
frequency: below 10 GHz, it is possible to ignore the inuence of the
atmospheric gasses; between 10 and 150 GHz, molecular oxygen dominates
the total attenuation (in this region the local attenuation peaks are at 22.3
GHz -Ka band- and at 60 GHz -V band-, respectively due to water vapor
and molecular oxygen); whereas, above 150 GHz, the eect of water vapor
is dominant.
Rain attenuation. This type of attenuation is the most signicant one
among the atmospheric eects. There are several prediction models to
establish the quantity of rain fall attenuation, depending on some pa-
rameters, such as the rain fall rate probability distributions, the slant
path length, and the rain height. With these parameters it is possible
to characterize the level of rain and the relative attenuation (e.g., rain,
widespread rain, showery rain, rainstorm, etc.).
Fog and clouds. The attenuation eects of fog and clouds are not so impor-
tant for systems operating below 30 GHz; while, they are signicant above
30 GHz. This type of attenuation is related to frequency, temperature and
liquid water density (expressed in g/m3 ). Empirical models (one of them
is recommended by ITU) are used to predict fog and clouds attenuation.
Scintillation. This is a phenomenon that aects satellite communication

systems operating above 10 elevation angle and below 10 GHz (Ku band).
This eect consists of small and quite rapid uctuations due to some
irregularities in the troposphere refractive index. As for the reception in
a mobile environment, the signal can be faded and enhanced by these
uctuations.
Channel losses
In satellite networks, Bit Error Rate (BER) is very high, due to the above-
mentioned atmospheric eects. The quality of the satellite link can be subject
to rapid degradation that can cause long sequences of erroneous bits. These
burst errors cause an on-o behavior for the channel. With the use of Forward
Error Correction (FEC) codes (e.g., Reed-Solomon codes, convolutional codes,
etc.), it is possible to reduce remarkably BER at the expenses of a lower
information bit-rate (i.e., part of the available capacity is spent in sending
redundancy bits).
Satellite lifetime
Satellites have an average life span due to the components ageing process, the
eect of radiations, the necessity of new components, etc. GEO satellites have
a lifetime in the range of 10-15 years. MEO satellites have an operational
period of 10-12 years. Finally, LEO satellites are ecient between 5 and 8
years, mainly due to radiation eects.
1.3 Multiple access techniques
Multiple access is the ability of a large number of Earth stations to simul-

taneously interconnect their respective multimedia trac ows via satellite
[1],[10]. These techniques permit to share the available capacity of a satellite
transponder among several Earth stations. The most common techniques are:
Frequency Division Multiple Access (FDMA),

Time Division Multiple Access (TDMA),
Code Division Multiple Access (CDMA),
A mix of the above schemes (e.g., combining TDMA and CDMA or FDMA
and TDMA).
These dierent multiple access techniques are surveyed below. Note that
another form of multiple access is also allowed in the presence of a multi-
spot-beam antenna on the satellite. This technique is called Spatial Division
Multiple Access (SDMA) [11]. With a multi-spot-beam antenna, some beams
may re-use the same frequencies, provided that the cross-interference (due to
beam radiation pattern side-lobes) is negligible. Usually, beams separated by

more than two or three half-power beam-widths can use the same frequen-
cies; this frequency reuse technique permits increasing the utilization of air
interface resources.
FDMA
In FDMA, the total bandwidth is divided into equal-sized parts; an Earth

station is permanently assigned with a portion around a carrier or carriers.
FDMA requires guard bands to keep the signals well separated. The trac
capacity of an Earth station is limited by its allocated bandwidth and the
Carrier power-to-Noise power ratio (C/N). The carrier frequencies and the
bandwidths assigned to all the Earth stations constitute the satellites fre-
quency plan. FDMA requires the simultaneous transmission of a multiplicity
of carriers through a common Traveling-Wave-Tube Amplier (TWTA) on the
satellite. The TWTA is highly non-linear (it produces maximum output power
at the saturation point, where the TWTA is operating in the non-linear region
of its characteristics) and the Inter-Modulation (IM) products generated by
the presence of multiple carriers produce interference. The only way to reduce
IM distortion is to lower the input signal level, so that the TWTA can operate
in a more linear region. For a given carrier, the dB dierence between the
single-carrier input power level at saturation and the input power level for that
particular carrier in multi-carrier FDMA operations is called input backo.
The corresponding output transmission power reduction in dB is called output
backo.
TDMA
In TDMA, the total bandwidth is usually divided into time slots, organized
according to a periodic structure, called frame. Each slot is used to convey
one packet. Hence, TDMA is well suited for packet trac. In TDMA uplink
transmissions, Earth stations take turns sending bursts through a common
satellite transponder. As for TDMA downlink transmissions from a satellite,
only one carrier is used. Hence, TDMA provides a signicant advantage, since
it permits a transponders TWTA to operate at or near saturation, thus
maximizing downlink C/N. However, interference is not totally eliminated,
since it is present in the form of inter-symbol interference that must be
minimized by means of appropriate ltering. TDMA is easy to recongure
for changing trac demands, it is robust to noise and interference and allows
mixing multimedia trac ows.
While in TDM (Time Division Multiplexing) all data come from the same
transmitter and the clock and time frequencies do not change, in TDMA
each frame contains a number of independent transmissions. Each station has
to know when to transmit and must be able to recover the carrier and the
data synchronization for each received burst in time to sort out all desired
base-band channels. This task is not easy at low C/N values. A long preamble
is generally needed, which decreases system eciency.
A group of Earth stations, each at a dierent distance from the satellite,
must transmit individual bursts of data in such a way that bursts arrive
at the satellite in correspondence with the beginning of the assigned slots.
Stations must adjust their transmissions to compensate for variations in
satellite movements, and they must be able to enter and leave the network
without disrupting its operation. These goals are accomplished by exploiting
the TDMA organization in frames, which contain reference bursts that permit
establishing absolute time for the network.
Reference bursts are generated by a master station on the ground in
a centralized-control satellite network. Each burst starts with a preamble,
which provides synchronization and signaling information and identies the
transmitting station. Reference bursts and preambles constitute the frame
overhead. The smaller the overhead, the more ecient the TDMA system,
but the greater the diculty in acquiring and maintaining synchronism.
Time access to the satellite link can be managed either in centralized or in
distributed mode. Centralized control is generally more robust. On the other
hand, the distributed control is more responsive to trac variations, since it
allows an update in one RTD.
CDMA
The signals are encoded, so that information from an individual transmitter

can be detected and recovered only by a properly synchronized receiving
station that knows the code used (scrambling code) for transmissions.
In a decentralized satellite network, only the pairs of stations that are
communicating need to coordinate their transmissions (i.e., they need to
use the same code). The concept at the basis of CDMA is spreading the
transmitted signal over a much wider band (Spread Spectrum). This technique
was developed as a jamming countermeasure for military applications in the
1950s. Accordingly, the signal is spread over a band PG times greater than
the original one, by means of a suitable modulation based on a Pseudo Noise
(PN) code. PG is the so-called Processing Gain. The higher the PG, the higher
the spreading bandwidth and the greater the system capacity. Suitable codes
must be used to distinguish the dierent simultaneous transmissions in the
same band. The receiver must use a synchronous code sequence with that of
the received signal, in order to de-spread correctly the desired signal. There
are two dierent techniques for obtaining spread spectrum transmissions:
Direct Sequence (DS), where the user binary signal is multiplied by the PN
code with bits (called chips) whose length is basically PG times smaller
that that of the original bits. This spreading scheme is well suited for
Binary Phase Shift Keying (BPSK) and Quadrature Phase Shift Keying
(QPSK) modulations.
Frequency Hopping (FH), where the PN code is used to change the

frequency of the transmitted symbols. We have a fast hopping if frequency
is changed at each new symbol, whereas a slow hopping pattern is obtained
if frequency varies after a given number of symbols. Frequency Shift Keying
(FSK) modulation is well suited for the FH scheme.
Comments and comparisons among the access techniques
The drawback of TDMA is the need to size Earth stations for the entire
system capacity (transponder bandwidth), even though the single terminal
uses a small portion of that. An interesting solution is given by the hybrid
combination of Multi-Frequency (MF) with TDMA systems, which takes some
advantages of both FDMA and TDMA [12]. In MF-TDMA the transponder
spectrum is divided into several carriers, thus allowing the sizing of the station
on a narrower bandwidth. Each carrier, in turn, is shared in TDMA mode.
The transmission of the trac occurs in time slots that may belong to dierent
carriers. When a single modulator is used, slots of a transmission need not to
overlap in time (i.e., simultaneous transmissions on dierent frequencies are
not allowed). The MF-TDMA technique eciently supports trac streaming,
while maintaining exibility in capacity allocation.
1.4 Radio interfaces considered and scenarios

Dierent standardized air interfaces are available for satellite communication
systems. In particular, this book is focused on both the satellite extension
of the terrestrial Universal Mobile Telecommunications System (UMTS) [1]
and the Digital Video Broadcasting via Satellite (i.e., DVB-S, DVB-S2 and
DVB-RCS) [13]-[16]. In addition to this, scenarios have been considered that
combine together dierent aspects, such as: satellite orbit type, mobile or xed
users, adopted air interface. In particular, the following scenarios have been
identied:
Scenario 1: Satellite-UMTS (S-UMTS) for mobile users through GEO
bent-pipe satellite;
Scenario 2: DVB-S/DVB-RCS for xed broadband transmissions via
GEO bent-pipe satellite;
Scenario 3: LEO constellation with regenerating satellites for the provi-
sion of multimedia services to mobile users adopting handheld devices.
1.4.1 S-UMTS
Satellite communication systems should be able to provide to mobile users the

same access characteristics of the terrestrial counterparts. We refer here to
the provision of 3rd Generation (3G) mobile communication services through
satellites. In particular, the interest is on the extension of the UMTS standard

to the satellite context (S-UMTS). The ETSI S-UMTS Family G specication
set aims at achieving the satellite air interface fully compatible with the
terrestrial W-CDMA-based UMTS system [17]-[20]. S-UMTS will not only
complement the coverage of the Terrestrial UMTS (T-UMTS), but it will
also extend its services to areas where the T-UMTS coverage would be either
technically or economically not viable.
The satellite radio access network of the S-UMTS type should be connected
to the UMTS core network via the Iu interface [1],[21]. S-UMTS is expected to
be able to support user bit-rates up to 144 kbit/s that appear to be sucient
to provide multimedia services to users on the move, employing typically small
devices [22].
With the evolution of terrestrial 3G systems standardization, the High
Speed Downlink Packet Access (HSDPA) has been dened to upgrade current
terrestrial 3G (W-CDMA) systems to provide high bit-rate downlink trans-
mission to users. HSDPAs improved spectrum eciency enables users with
downlink speeds typically from 1 to 3 Mbit/s. Hence, capacity-demanding
applications are possible, such as video streaming. The mandatory codec for
streaming applications is H.263, with settings depending on the streaming
content type and the streaming application.
The novel HSDPA air interface is based on the application of Adaptive
Coding and Modulation (ACM) and multi-code operation depending on the
channel conditions (forward link) that are feed back by the User Equipment
(UE) to the Node-B. The interest in this book is on the study for the possible
extension of HSDPA via satellite, as an upgrade of S-UMTS specications. In
this case, all resource management functions for the S-HSDPA air interface are
managed by the base station (i.e., Node-B) on the Earth that is directly linked
to the Radio Network Controller (RNC) that operates as a gateway towards
the core network. More details on this study will be provided in Chapter 5.
1.4.2 DVB-S standard
DVB-S has been designed for primary and secondary distribution in the bands
of FSS and BSS [13]. Such systems should be able to provide direct-type
services (Direct-To-Home, DTH) both to the single consumer having an
integrated receiver-decoder, to systems with a collective antenna and to the
terminal stations of cable-TV. The frequency bands for feeder and user links
may occupy Ku/Ku, Ku/Ka and K/Ka bands.
Below the transport layer and the IP layer the Multi Protocol Encapsula-
tion (MPE) provides segmentation & reassembly functions for the generation
of Moving Picture Experts Group 2 - Transport Stream (MPEG2-TS) packets
of 188 bytes (xed length). A TCP header of 20 bytes, an IP header of
20 bytes and an MPE header + CRC trailer of 12 + 4 bytes are added
to packets from the application layer; the resulting blocks are fragmented
in payloads of MPEG2-TS packets. All the data ows transported in single
MPEG2-TS are of the TDM type. In the channel adaptation section, packets
are processed in several steps, such as: channel encoding (outer Reed-Solomon
coding, convolutional interleaver, inner convolutional encoding, puncturing),
base-band shaping of impulses, and QPSK modulation. The resulting DVB-S
transmissions via satellite are very robust, considering a minimum BER of
about 1011 . As an example, a typical data rate of about 38 Mbit/s is obtained
with modern satellite transponders that have a bandwidth of about 33 MHz
[13].
1.4.3 DVB-RCS standard
One of the reasons for the denition of a DVB standard with satellite
return channel (DVB - Return Channel via Satellite, DVB-RCS) has been
the increasing request of interactive applications and services with major
informative volumes (1 ) that could not be achieved with a DVB-S-based
system, where the return channel (realized through a terrestrial link via
modem) cannot permit an adequate bit-rate capacity (maximum 64 kbit/s).
The specications of DVB-RCS use and modify the DVB-S ones [14],[15];
moreover, they are independent of frequency, making easier to realize network
and security mechanisms with an ecient transport layer. The DVB-S channel
has been named Forward Channel, while the Return Channel is related to
the link from the end-user back to the content network (see Figure 1.4).
The return channel has a variable bit-rate up to a maximum of 2 Mbit/s
and can dynamically assign its time-frequency resources (according to an
MF-TDMA air interface) to the requesting terminals. The Return Channel
Satellite Terminal (RCST) transmission capacity is constrained. According to
the standardization, RCSTs can be single-user (144-384 kbit/s) or corporate
(2 Mbit/s).
The standard [14],[15] denes a reference model for the Interactive Satellite
Network (ISN) architecture, composed of a certain number of RCSTs, a GEO
bent-pipe satellite, and the following elements:
Network Control Center (NCC): it provides Control and Monitoring

Functions (CMF); moreover, it produces timing & control signals that
one or several Feeder Stations transmit for the ISN operations.
Trac Gateway (GW): it is a router that sends/receives data to/from
the RCSTs, managing the exchange of data with public, proprietary and
private providers.
Feeder : it is the Earth station that transmits Forward Link (DVB-S)
signal, where user data and ISN timing & control signals are multiplexed
together.
1
Recently, also other systems have been standardized for broadband satellite access
such as DOCSIS-S and IPoS [23].
Figure 1.4 shows a simplied version of the DVB-S/DVB-RCS system

architecture where NCC, GW and Feeder are collapsed into the NCC, i.e.,
in a single Earth station.
Fig. 1.4: Example of DVB-S/DVB-RCS system architecture.
Air interface characteristics of DVB-RCS
In order to operate successfully an ISN, it is important to use the satellite

resources as eciently as possible. Therefore, Bandwidth on Demand (BoD)
schemes (also known with the name of Demand Assignment Multiple Access,
DAMA, techniques) have been introduced in the DVB-RCS standardization
in order to improve the utilization of satellite resources in the presence of
distinct trac classes.
The DVB-RCS standard species a MAC layer in which the NCC controls
the allocation of the uplink capacity for RCST transmissions. BoD is dened
as a set of MAC protocols and algorithms that allow an RCST to request
resources to the NCC, when the RCST has trac to pass to GW.
Return link transmissions are based on an MF-TDMA air interface,
where RCSTs transmit their data using a range of carrier frequencies (with
potentially dierent bandwidth size), each of them organized in super-frames,
frames and time-slots. The NCC assigns to each active RCST a set of bursts,
each of them is dened by frequency, bandwidth, starting time and duration.
Dierent carriers can have the same or dierent timeslots characteristics,
thus having a xed or a dynamic timeslot structure. In the former case,
timeslots have xed characteristics, in terms of bandwidth and duration.
Whereas, in the latter case, besides bandwidth and time-slot duration, both
transmission rate and code rate can be changed in consecutive slots. Such
exibility allows a better RCST adaptivity to the variable requirements of
multimedia transmissions.
The return link time and frequency organization of the air interface is
depicted in Figure 1.5. Each super-frame is characterized by a superframe id,
and can be assigned to a group of RCSTs. In turn, each super-frame is
divided in parts, characterized by a superframe counter that can be divided in
frames, identied by a frame number (F nb) or by a frame ID (F id). Frames
can have dierent duration, bandwidth and composition of timeslots. Each
frame is divided in timeslots characterized by a timeslot number (TS nb);
also timeslots can be organized in slot groups with similar characteristics.
Fig. 1.5: Organization of the resources in the MF-TDMA air interface.
The RCST is responsible for analyzing, estimating and requesting the

needed capacity for uplink transmissions (DAMA case), and for distributing
the allocated capacity to the internal applications according to some rules.
In particular, when an RCST has data to transmit, it rst explicitly requests
the needed capacity to the NCC (Capacity Request, CR, message). The NCC
allocates return channel time slots based on each requests and informs all
RCSTs of allowable transmission slots by using Terminal Burst Time Plan
(TBTP) messages, sent regularly (e.g., once per super-frame) over the forward
channel. Each RCST looks at the received TBTP and transmits data during
the allocated time slots.
Allocation methods and trac classes in DVB-RCS
Five capacity allocation methods (layer 2) are dened in the DVB-RCS

standard [14],[15]:
Continuous Rate Assignment (CRA),
Rate Based Dynamic Capacity (RBDC),
Volume Based Dynamic Capacity (VBDC),
Absolute Volume Based Dynamic Capacity (AVBDC) and
Free Capacity Assignment (FCA).
Note that CRA is a xed capacity allocation, while RBDC, VBDC and
AVBDC are DAMA schemes. Finally, with FCA the NCC assigns unutilized
resources in a super-frame (after the fulllment of the other request types),
without any particular requests made by RCSTs. In allocating resources, the
NCC adopts the following priority order:
CRA > RBDC > A(V BDC) > F CA.
Details on the capacity allocation methods are provided below.
Continuous Rate Assignment (CRA): CRA is a rate capacity that

shall be provided in full for every super-frame while required. CRA is a xed
(and static) allocation of resources after an initial set-up phase with a nego-
tiation between the RCST and the NCC. With CRA, a given number of time
slots (i.e., packets) are continuously assigned to that RCST every super-frame
until that RCST sends the assignment release message. CRA would typically
be subscription-based: the user subscribed to a certain constant rate, and the
RCST has automatically assigned this constant rate at log-on. CRA should be
used for trac, which requires a xed guaranteed rate, with minimum delay
and minimum delay jitter, such as the Constant Bit Rate (CBR) class of ATM
networks. The CRA allocation method could also be used in conjunction with
RBDC to manage a Variable Bit Rate (VBR) trac that could not tolerate the
request-allocation loop delay. In this case, CRA would guarantee a minimum
bit-rate and RBDC should provide an additional dynamic capacity.
Rate Based Dynamic Capacity (RBDC): RBDC is a rate capacity

that is dynamically requested by the RCST. RBDC capacity shall be provided
in response to explicit CR messages from the RCST to the NCC, such requests
being absolute (i.e., corresponding to the full rate currently being requested).
Each request shall override all previous RBDC requests from the same RCST,
and shall be subject to a maximum rate limit negotiated directly between the
RCST and the NCC, RBDCmax . To prevent an RCST anomaly resulting in
a hanging capacity assignment, the last RBDC request received by the NCC
from a given RCST shall automatically expire after a time-out period, whose
default value is 2 super-frames, such expiry resulting in the RBDC being reset
to zero rate. CRA and RBDC could be used in combination, as previously
explained. A typical application for RBDC over a GEO satellite could be to
support the Available Bit Rate (ABR) trac class of ATM networks.
Volume Based Dynamic Capacity (VBDC): VBDC is a volume

capacity, dynamically requested by the RCST. VBDC capacity shall be
provided in response to explicit CR messages from the RCSTs to the NCC,
such requests being cumulative (i.e., each request shall add to all previous
requests from the same RCST). The request indicates a total number of needed
trac slots (i.e., packets) that can be shared between several super-frames;
successive VBDC requests add up. VBDC should be used only for trac
that can tolerate delay jitter, such as the Unspecied Bit Rate (UBR) trac
class of ATM or standard IP trac. VBDC and RBDC can also be used
in combination for ABR trac, with the VBDC component providing a
low priority capacity extension above the guaranteed limit of the RBDC
category. MAC parameters are the minimum (V BDCmin ) and the maximum
(V BDCmax ) volume request.
Absolute Volume Based Dynamic Capacity (AVBDC): AVBDC is

a volume capacity that is dynamically requested by the RCST. This AVBDC
capacity shall be provided in response to explicit CR messages from the
RCST to the NCC, such requests being absolute (i.e., a request replaces the
previous ones from the same RCST). The request indicates a total number of
trac slots that can be shared between several super-frames; a new AVDBC
allocation cancels the previous ones. AVBDC is similar to VBDC and should
be used instead of VBDC for the initial request or when the RCST senses
that the VBDC request might be lost (re-initialization of a previous request);
this might happen when requests are sent on contention bursts (see the next
description on related signaling methods) or when channel conditions (e.g.,
packet error rate, Eb /N0 ) are degraded. AVBDC is suitable to support the
same trac classes of VBDC.
Free Capacity Assignment (FCA): FCA is a volume capacity that

shall be assigned to RCSTs from capacity, which would be otherwise unused.
Such capacity assignment shall be automatic, not involving any requests from
the RCSTs to the NCC. In particular, FCA should not be mapped to any
trac category since availability is highly variable. The assigned capacity
is intended as a bonus, which can be used to reduce delays on any trac
type that can tolerate delay jitter. It should be noted that the term free in
FCA refers to spare system capacity. CRA and FCA can also be viewed as
two mechanisms to grant dynamically capacity to an RCST without explicit
requests. FCA resources should be distributed to RCSTs according to the
following criterions ranked by priority:
1. Performance optimization of TCP/IP in order to reduce the occurrence

of TCP timeouts;
2. Equity (i.e., equal sharing of resources according to a round-robin scheme).
RBDC and VBDC methods are quite similar, but they dierentiate on the
basis of:
The type of requested capacity (i.e., capacity expressed as a bit-rate in
RBDC, or capacity expressed in terms of packets in VBDC);
Request characteristics that are absolute in RBDC and cumulative in
VBDC.
RBDC appears a more complex scheme since it involves a technique to
estimate the requested bit-rate. With such a scheme, it is however, possible
to follow better the bursty characteristics of the input trac.
In order to send CR messages from the RCSTs to the NCC two signaling
methods are available:
In-band signaling. CRs are encapsulated in a Satellite Access Control
(SAC) format and can be sent in SYNC bursts or normal MPEG2 data
bursts using Data Unit Labeling Method (DULM), typically employed to
send control and administrative information to the NCC.
Out-of-band signaling. A minislot method is used (with or without con-
tentions): minislots are periodically assigned to an RCST (or a group of
RCSTs) for the transmission of shorter bursts than those used for trac
purposes.
To each transmission request made by an RCST, latency is associated
mainly due to RTD. The Minimum Scheduling Latency (MSL) is the minimum
delay between the computation of a CR and the time when it is possible to
use the requested capacity by an RCST. In case of a bent-pipe satellite, MSL
entails the following contributions (see Figure 1.6):
CR evaluation and transmission;
Round trip time between the RCST and the NCC (500 ms for a GEO
bent-pipe satellite);
Processing delay on the NCC (80 ms);
TBTP transmission time from the NCC;
TBTP processing delay on the RCST.
A typical choice for the super-frame length is 500 ms that also corresponds
to the TBTP and CR transmission periodicity. A possible value for the frame
length is 50 ms.
The DVB-RCS standard envisages 4 priorities (i.e., trac classes) that are
listed below in order of decreasing urgency level [15]:
The Real Time (RT) class for the applications that require strong time
constraints (e.g., VoIP and videoconference);
Fig. 1.6: Delay contributions in the process to allocate resources to RCSTs.
The Variable Rate - Real Time (VR-RT) class is for variable bit-rate jitter-
sensitive trac;
The Variable Rate - Jitter Tolerant (VR-JT) for variable bit-rate jitter-
tolerant trac (e.g., FTP application);
The Jitter Tolerant Priority trac class.
An RCST may queue all trac arriving from the user interface, using
separate queues for ows that are subject to dierent transmission priorities
(i.e., service classes) [15]. As an example, one layer 2 queue shall be provided
for each of the priorities (i.e., RT, VR-RT, VR-JT, JT); each queue should
be served with a capacity allocation method (or a combination of them). For
instance: CRA for RT, RBDC for VR, VBDC/AVBDC+FCA for JT.
Typically, at the IP level 4-16 queues can be managed according to specic
IP QoS classes; while at layer 2, typically 4 queues are envisaged [24],[25].
Hence, the IP QoS service classes (i.e., layer 3 queues) need to be adequately
mapped into equivalent MAC QoS classes (i.e., layer 2 queues).
Trac generated at the RCST is rst classied and packets are stored into
one of several layer 3 queues. From layer 3 we have MPE encapsulation (see
Figure 1.7) and the generation of layer 2 packets (e.g., MPEG2-TS) provided
to suitable queues, waiting for transmission.
In a connectionless network, the prioritization of voice packets in both
directions is crucial in order not to degrade the voice quality. Thus, the priority
element plays an important role in the BoD architecture and must be present
in all steps of the transmission.
1.4.4 DVB-S2 standard
After 10 years from the denition of DVB-S in 2003, the European DVB
consortium has developed a second-generation standard for satellite broadcast
transmissions, named DVB-S2 [16]. Such system employs the most recent
Fig. 1.7: MPE encapsulation for IP trac.
advances in channel coding (e.g., Low Density Parity Check, LDPC, described
below) combined with several modulation types (i.e., QPSK, 8PSK, 16APSK
and 32APSK).
Besides broadcasting services, DVB-S2 can be employed for interactive
point-to-point applications (e.g., Internet access) by using new modulation
schemes and new operation modes that permit to optimize the modulation and
coding schemes depending on channel conditions. In order to allow that DVB-
S continues to operate during the transition period, the DVB-S2 standard
also provides transmission means compatible with the satellite decoders of
rst-generation (Set-Top-Box, STB).
A DVB-S2 transmitter is composed by the following functional blocks
that are described below [16],[26]: mode adaptation, stream adaptation, FEC
encoding, modulation mapping, physical layer framing, base-band ltering
and quadrature modulation.
Mode adaptation
There are three (application-dependent) operation modes for DVB-S2: Con-

stant Coding Modulation (CCM), Variable Coding and Modulation (VCM)
and Adaptive Coding and Modulation (ACM) [27].
CCM is a constant protection system, which represents the simplest mode
of DVB-S2; it is similar to the DVB-S one, since all data frames are
modulated and coded with the same xed parameters. Unlike DVB-S,
DVB-S2 uses an LDPC inner error correction code.
VCM can be applied to give distinct error protection levels to dierent
services (e.g., robust protection for SDTV and less-robust protection for
HDTV, audio, multimedia). In fact, the DVB-S2 standard supports the
transmission of dierent services on the same carrier, each operating with
its own modulation scheme and coding rate. VCM performs a kind of
multiplexing operation at the physical layer to provide distinct services

with dierent characteristics.
ACM is a functionality oered by DVB-S2, in case of interactive and point-
to-point applications, when a return channel is available. ACM permits to
change dynamically the coding rate and the modulation scheme on the
basis of the measured channel conditions at the site that must receive the
frame. The sender site dynamically acquires information on the receiving
conditions by means of the return channel.
The following description considers the dierent service scenarios where

DVB-S2 can be used. In particular, the following application areas are
considered: Broadcast Services, Interactive Services, Digital TV Contribution
and Satellite News Gathering and other Professional Services/applications.
More details are provided below in relation to the operation modes.
Broadcast Services are provided via DVB-S2 with the exibility of VCM.
There are also Backwards Compatible-Broadcast Services used for a joint
interoperability with DVB-S decoders, and optimized Non-Backwards
Compatible-Broadcast Services.
Interactive Services are designed to operate with existing DVB return
channel standards (e.g., RC-PSTN, RCS, etc.). DVB-S2 can use both CCM
and ACM.
Digital TV Contribution and Satellite News Gathering applications refer
to point-to-point, or point-to-multipoint communications of multiple or
single MPEG-TS, by means of CCM or ACM modes.
Professional Services/applications mainly consists of professional point-
to-point and point-to-multipoint applications (e.g., data content distribu-
tion); for these services, DVB-S2 uses CCM, VCM or ACM techniques.
Stream adaptation
This operation is applied to perform padding (to complete a base-band frame)

and base-band scrambling.
FEC encoding
FEC permits to achieve excellent performance also in the presence of high

levels of noise and interference. FEC is achieved with the concatenation of
BCH (Bose-Chaudhuri-Hocquenghem) outer codes and LDPC inner codes.
This technique permits to achieve a performance quite close to the Shannon
limit. BCH outer codes are used to avoid error oors at low BER values. The
selected LDPC codes [12] operate with code rates of 1/4, 1/3, 2/5, 1/2, 3/5,
2/3, 3/4, 4/5, 5/6, 8/9 and 9/10, depending on the adopted modulation and
the system requirements. In particular, coding rates 1/4, 1/3 and 2/5 are used,
combined with QPSK modulation in the presence of poor link conditions.
Depending on the application area, the FEC coded blocks have very large
lengths (64800 bits for delay-tolerant applications, or 16200 bits). In the VCM
and ACM cases, FEC and modulation mode can be varied in dierent frames,
but they are constant in a frame.
Finally, bit interleaving shall be applied to 8PSK, 16APSK and 32APSK
FEC coded bits.
Modulation mapping
Four constellations can be used for the transmitted payload, depending on

the application area (see Figure 1.8) [28], as described below:
QPSK and 8PSK are typically suggested for broadcast applications, since
they have a quasi-constant envelope so that they can operate inside the
non-linear region of satellite transponders (i.e., close to the saturation).
Gray mapping can be used for these modulations.
The 16APSK and 32APSK modes, mainly proposed for professional appli-
cations (these modulations could also be used for broadcasting), require a
higher level of available C/N and the adoption of advanced pre-distortion
methods to reduce the non-linearity eects in transponders.
DVB-S2 is expected to achieve spectral eciencies ranging from 0.5

bit/s/Hz up to 4.5 bit/s/Hz.
Physical layer framing
This sub-system, synchronously with the FEC frames, generates the Physical
Layer Frame (PLFRAME), supporting also some tasks, such as: dummy
PLFRAME insertion, physical layer signaling, optional pilot symbols insertion
and physical layer scrambling for energy dispersion.
A DVB-S2 system can be used with two congurations: single carrier
per transponder and multi-carrier per transponder (the bandwidth of the
transponder is divided with Frequency Division Multiplexing, FDM, among
dierent carriers and related bands).
In case of ACM mode, the DVB-S2 air interface varies exibly coding
and modulation techniques to maximize performance and coverage. This is
achieved through the TDM transmission of a sequence of PLFRAMEs, where
the coding and modulation format can change for each new PLFRAME.
Base-band ltering and quadrature modulation
This function is used for a tighter bandwidth shaping (squared-root raised

cosine) and to generate the radio frequency signal.
Fig. 1.8: The four possible DVB-S2 constellations before physical layer scrambling.
1.4.5 Numerical details on the selected scenarios for performance

evaluations
This sub-Section provides some basic characteristics and numeric values for
the parameters that have been used when evaluating the performance of the
techniques proposed in the following Chapters of this book for the dierent
scenarios. The details are provided below.
Scenario 1: S-UMTS as well as S-HSDPA

GEO satellite
Multi-spot-beam satellite antenna
Bent-pipe satellite
Terrestrial gateway containing the scheduler (MAC layer)
Direct return link via satellite for channel quality measurements in case of
point-to-point services
Mobile users (mean speed equal to 60 km/h)
GOOD-BAD Markov channel model (typically, 6 s mean GOOD duration
and 2 s mean BAD duration) [29]
IP-based trac ows with UMTS transport layer encapsulation
Trac sources: video sources (sum of ON/OFF Markovian sources) [30]
and Web sources (2-MMPP arrival process of Pareto-distributed data-
grams) [31].
Scenario 2: DVB-S/DVB-RCS
GEO satellite
Single beam or multi-spot-beam satellite antenna
Bent-pipe satellite
Architecture involving an NCC and at least a GW
Fixed users
Direct return link for channel quality measurements; typically, Ka band is
used (maximum capacity 2 Mbit/s)
Forward link in K band
Channel model: only troposphere eects (rain scintillation and gas) have
to be considered. Basically an Additive White Gaussian Noise (AWGN)
model has been adopted with a given packet error rate (uncorrelated losses)
IP-based trac ows with MPE encapsulation and generation of packets
according to the MPEG2-TS format
Trac sources of the FTP type (elephant TCP connections).
Scenario 3: LEO constellation
A Teledesic-like LEO system (the Boeing design with 288 satellites):

altitude of 1375 km, and satellite capacity of 32 Mbit/s
Multi-spot-beam satellite antenna
End-users must switch from spot-beam to spot-beam and from satellite to
satellite, resulting in frequent intra- and inter-satellite handovers
We assume a two-dimensional mobility model: users move in straight lines
and at constant speed (satellite ground track speed composed with the
Earth rotation speed)
All the spot-beam footprints are identical in shape and size (approximated
by rectangles, 1790 km 1790 km)
Trac assumptions (study made in Chapter 8, Section 8.6): non-real-time
trac for email or FTP and real-time multimedia trac, e.g., interactive
voice and video applications. For each class: (i ) new calls arrive in the foot-
prints according to independent Poisson processes; (ii ) call holding times
are exponentially distributed. Within each trac class, three dierent user
types are considered that are dierentiated depending on the call holding
time and bit-rates.
1.5 Satellite networks

A satellite network can play several roles [32]. In particular, it can be used as
Access Network for nal users or it can be part of the Core Network. Some
examples are shown in Figure 1.9.
The ETSI TC-SES/BSM (Satellite Earth Stations and Systems / Broad-
band Satellite Multimedia) working group had the task to focus on IP layer
Fig. 1.9: Examples for the use of satellite links in telecommunication networks.
interworking, to dene a new network architecture and to include alternative

families of lower layer air interfaces [33]. A Broadband Satellite Multimedia
(BSM) network is divided into 5 domains, as specied in ETSI TR 101 984
[32]:
User Domain, representing the group of end-users;

Access Domain that denotes the access network that is used to connect to
the service provider (e.g., ADSL, UMTS, satellite);
Distribution Network : this is an intermediate network that is interposed
between the access network and the core network;
Core Network, representing the backbone transport network that is used
to connect the routers on a geographical area;
Content Domain that represents the area where contents and information
are stored to be made available to users.
The user requesting contents should access them feeling like as he/she was
directly connected to the source of the information, the Content Domain;
practically, many domains are traversed that are transparent to the user.
Let us now consider the BSM network functions from the protocol stack
standpoint (see Figure 1.10) that can involve dierent layers, as specied in
ETSI TR 101 985 [34]:
The BSM network operates at layer 2, like a bridge.

The BSM network operates at layer 3, so that the satellite Earth stations
are routers.
The BSM network operates at a layer above the 3rd one: the satellite
Earth stations are gateways. In this case, these stations can perform a
more accurate routing based not only on the IP datagram header, but also
on information of higher layer headers. In such a case, the Earth station can
implement special functions, like Performance Enhancing Proxies (PEPs)
that are important in order to improve the TCP performance in satellite

networks.
Fig. 1.10: BSM general network architecture.
Very Small Aperture Terminal (VSAT) networks are a special case of BSM
networks where the user terminal employs a small antenna (i.e., VSAT) and
simplied equipment so as to reduce costs. This small satellite terminal can
be used for one way and/or interactive communications. VSATs can support
several applications, such as: satellite news gathering, supervisory control
and data acquisition, inquiry/response, TV and audio broadcasting, data
distribution. VSAT networks are based on GEO satellites (typically of the
bent-pipe type) according to a star topology: an Earth station acts as a hub
(= gateway to the terrestrial network and master control station), receiving
and transmitting all the data uxes from/to VSATs. The forward link (from
the hub to VSATs) is via GEO satellite. The return link (from VSAT to the
hub) is typically via a terrestrial Public Switched Telephone Network (PSTN)
link (to simplify the antenna design on the VSAT). Hence, forward and return
links have an asymmetrical capacity; anyway recent advances in this eld also
allow the return link via satellite. Referring to the network architecture in
Figure 1.10, the VSAT includes the client and the Earth station on the left;
whereas, the hub coincides with the Earth station on the right. Dierent
VSAT platforms use various technologies in order to access the satellite radio
space segment and to share it among multiple users. One of the problems
that VSAT networks have faced during their evolution has been the lack of
compliance to any specic standards. In the last years, standardization bodies
have established new standards to support satellite Internet [23]. The DVB
standard has been the rst one to be published, and ETSI adopted DVB-RCS
for satellite return link transmissions. Another standard is IPoS (Internet
Protocol over Satellite) developed by HNS (Hughes Network Systems) and
standardized by ETSI. Finally, DOCSIS-S (Data Over Cable Service Interface

Specication for Satellite), a modication to the DOCSIS cable-modem has
been proposed for adapting it to the transmissions over satellite.
Let us focus on satellite IP networks. The ETSI TC-SES/BSM working
group has dened the protocol stack architecture shown in Figure 1.11 where
lower layers depend on satellite system implementation (Satellite-Dependent,
SD, layers) and higher layers are those typical of the Internet protocol stack
(Satellite-Independent, SI, layers). These two blocks of stacked protocols are
interconnected through the SI-SAP (Satellite-Independent - Service Access
Point) interface. Only a small number of generic functions need to cross the
SI-SAP; in particular: address resolution, resource management, trac classes
QoS.
The SI-SAP interface is logically divided into three SAPs, each of them
with a suitable function and security characteristics, as described in the ETSI
TS 102 465 standard [35]. In particular, we have:
SI-U-SAP (User-SAP ): transfer of IP packets between the users;

SI-C-SAP (Control-SAP ): transfer of control data and of service signaling
for SI-U-SAP;
SI-M-SAP (Management-SAP ): transfer of management information.
The protocol stack organization dened by TC-SES/BSM (see Figure 1.11)

has been taken as the basis for the organization of the work in this book, where
after a rst part with introductory concepts, the second part deals with SD
layers and the third part focuses on SI protocol layers. More details on the
BSM protocol stack are provided in the following sub-Section.
1.5.1 SI-SAP interface overview
SI-SAP denes an interface between SI upper layers and SD lower layers,

that applies to all air interface families for satellite communication systems
[32],[34]. SI-SAPs correspond to the endpoints of BSM bearer services. SI-SAP
is used to dene standard SI bearer services that are built upon lower
layer transmission services. Point-to-point, point-to-multipoint, multipoint-
to-multipoint and broadcast bearer services are dened as the edge-to-edge
services provided by the BSM sub-network. SI-SAP provides an abstract
interface allowing BSM protocols (BSM address resolution, BSM resource
management, etc.) to perform over any BSM family (i.e., layer 1 and 2
technology) [33]. For trac handling purposes, SI-SAP uses a BSM Identier
(BSM ID) and Queuing Identiers (QIDs):
The BSM ID uniquely identies a BSM network point of attachment
and allows IP layer address resolution protocols (equivalent to Address
Resolution Protocol, ARP for IPv4 and Neighbor Discovery, ND for IPv6)
to be used over the BSM.
Fig. 1.11: Standardized protocol stack.
QIDs are abstract queues (SI-SAP level) that represent the layer 2 queues
in a general way to allow the mapping with layer 3 ones (note that
using a QoS support mechanism at layer 3, dierent queues are needed).
QIDs are a way to hide specic SD layer implementations (i.e., BSM
technology) from the IP layer. Each QID queue is characterized by
QoS-specic parameters (owspecs, path label or Dierentiated Service,
DiServ, marking) and is associated to lower layer transfer capabilities
(i.e., capacity allocation methods) and buer management policies [36].
The SD layers are responsible for assigning satellite capacity to these
abstract queues (e.g., in DVB-RCS we can consider the allocation methods
such as CRA, VBDC, etc., and combinations of them). The mapping of
IP queues to QIDs is exible: there is no strict constraint for a one-to-one
mapping, but we may also consider that more IP queues correspond to the
same QID (in this case, a scheduler should be used at layer 3 to determine
the service order of the dierent queues to be mapped to the same QID).
BSM networks use a suitable and general categorization of trac ows
in trac classes that can be mapped to classical IP QoS classes [25]. In
particular, 8 trac classes, i.e., service priority levels, are dened from 0
for emergency services to 7 for low priority broadcast/multicast trac.
Other functional blocks are involved in the management of the queues in
BSM protocol architecture; the interested reader may refer to [36].
All the BSM services (data transfer, address management, group adver-
tisement, etc.) use SI-SAP primitives [37]. These primitives are classied
into functional groups within the User plane (U), Control plane (C) and
Management plane (M). The primitives (exchanged between the upper layers
and the lower layers) are of the following four types:
The REQUEST primitive type is used when the SI layer is requesting a

service from the SD layer.
The INDICATION primitive type is used by the SD layer to notify the SI
layer of activities. This type may either be related to a REQUEST type
at the peer entity, or may be an indication of an unsolicited lower layer
event.
The RESPONSE primitive type is used by the SI layer to acknowledge the
receipt of the INDICATION type from the SD layer.
The CONFIRM primitive type is used by the SD layer to conrm that the
activity requested by a previous REQUEST primitive has been completed.
The services provided at the SI-SAP level are divided into functional
groups for U-, C-, and M-plane, as described below [37]. Each service uses
one or more dierent types of the above-mentioned primitives.
U-plane services
Data Transfer : These services are used to send and receive user data
via the SI-SAP. Data transfer services can be used for both unicast and
multicast data transfer.
C-plane services
Address Resolution: A mechanism to associate a BSM ID address to a
given IPv4 unicast or multicast address. A successful address resolution
service returns the associated BSM ID. The BSM ID can be either a
Unicast ID for unicast services or a Group ID for multicast services.
Resource Reservation: These services are used to open, modify and
close SD layer queues (for both unicast and multicast ows) to be used
by SI layers. This function assigns the QID and denes or modies
the properties of the abstract queue that is associated with that QID.
Resource reservation is required only for sending data (not for receiving
data).
Group Receive/Send : They are mechanisms to activate and congure
the SD layers to receive/send a needed multicast service. These services
are used to associate a multicast group address (e.g., an IPv4 Class D
address, or an IPv6 multicast address) with a series of SD parameters.
Flow Control : These primitives allow activating and adjusting the SD
layers to provide SI-SAP ow control for a specic QID (i.e., on one or
more of the SD layer queues).
M-plane services
At present, no M-plane services are dened in the standard.
1.6 Novel approaches for satellite networks

The increasing demand for multimedia broadband services and high-speed
Internet access via satellite requires the denition of an optimized satellite
protocol stack as well as the full integration of the satellite network with
terrestrial ones. Consequently, two innovative approaches are at present
conceived [38], namely horizontal approach and vertical approach.
1.6.1 Horizontal approach
We expect that dierent wireless technologies (e.g., wireless local area net-
works, cellular systems, satellite networks) need to co-operate to allow the
best radio coverage to the users, depending on their locations, mobility
characteristics, applications, user prole, etc. This is in accordance with the
Always Best Connected (ABC) paradigm. Therefore, it is necessary that the
use of the resources in the dierent Radio Access Networks (RANs) be globally
coordinated by means of a resource brokerage function. Such intelligence is
centralized and allocates sessions to RANs or switches them from one to
another, when some conditions are meet.
1.6.2 Vertical approach
The ISO/OSI reference model and the Internet protocol suite are based on
a layering paradigm. Each protocol solves a specic problem by using the
services provided by modules below it and gives a new service to upper layers.
The disadvantages of such approach can be detailed as follows:
The needs of a service provided by the communication system to its users
are dened at the top-level, but the hierarchy and the overall system
performance are built upon the bottom-level.
The bottom level does not communicate directly, but through intermediate
layers with the top-level. Information is lost during this layer-by-layer top-
down conversion.
Layers are independently optimized. However, in many cases, the close
interaction among them should be considered.
A strict modularity and layer independence may lead to non-optimal
performance in IP-based next-generation satellite communication systems.
Finally, since both radio and power resources are strongly constrained on the
satellite, a protocol optimization is mandatory. Such optimization requires a
vertical design of the air interface protocol stack. Such cross-layer approach
entails new interfaces across the layers, which exchange control information
beyond the standard ISO/OSI structure. Cross-layer interfaces can be between
or beyond adjacent protocol layers. Although interfaces between adjacent
layers are in general preferable, there can be the need for ecient and direct in-
teractions between non-adjacent layers [39]; in general, a layer should be aware
of the internal state of the other layers of the protocol stack. For instance,
OSI layer 3 (e.g., IP) and above often need direct interfaces to OSI layer 2,
e.g., for handover support. Another example concerns transmission parameters
(e.g., transmission mode, channel coding and persistency degree for link layer
retransmissions) that must be related to application characteristics (e.g., type
of information, source coding, etc.), network characteristics, user preferences
and context of use. Finally, lower layers (i.e., 1 and 2) should be aware of
higher layer (i.e., 3 and 4) behaviors in order to take appropriate decisions on
trac management.
Cross-layer methods can be classied into two broad groups as follows:
Implicit cross-layer design: there is no exchange of signaling among dier-

ent layers during operation, but in the design phase cross-layer interactions
are taken into consideration for a joint optimization.
Explicit cross-layer design: signaling interactions among (non-)adjacent
protocol levels are employed so that the internal state of a protocol can be
made available to the protocols at dierent layers for dynamic adaptations.
The above distinction of methods is at the basis of all the dierent

cross-layer schemes presented in the following Chapters of this book.
As for explicit cross-layer methods, new interfaces are needed beyond adja-
cent layers. Dierent solutions have been proposed to support the cross-layer
exchange of signaling information; an interesting method has emerged from
the following papers [40]-[43], where a global coordinator of the dierent
layers is considered allowing to acquire status information from the dierent
protocols to store it in a shared memory and to set the internal state of the
protocols to be adaptable to dierent events. A possible implementation of
the global coordinator could be to exploit the capabilities of the management
plane of the protocol stack that can interact and coordinate the dierent
layers. The management plane could exploit the control service access points
between layers to send control broadcast signaling to all the layers for their
respective actions. More details on these aspects are provided in Chapter 4.
Finally, referring to the ETSI TC-SES/BSM protocol stack architecture
shown in Figure 1.11, it is important to note here that the cross-layer methods
involving SD and SI layers will require the adoption of suitable primitives
crossing the SI-SAP interface. In order to explain BSM cross-layer interactions
and their relations to SI-SAP, some examples are proposed below for both
implicit and explicit cross-layer design cases.
BSM Protocol Manager
The BSM Protocol Manager (BPM) has been conceived in the BSM protocol
stack to maintain QoS and evaluate the BSM performance [44]. BPM resides
above the SI-SAP and denes how IP protocols and packet markings are
interpreted and transmitted through the BSM, which SI protocols are used
and how they in turn trigger the SD functions (see Figure 1.12). BPM
has interfaces at dierent levels of the BSM protocol stack. In particular,
BPM interacts with a specic middleware to establish transport level and
application level PEPs, communicates with bandwidth brokers and potentially
with service discovery and security/authentication functions. BPM directly
interacts with IP protocols, including Multiprotocol Label Switching (MPLS)
for route discovery and Integrated Service (IntServ) or Dierentiated Service
(DiServ) models (see Chapter 8, Section 8.2, for more details). For all these
reasons the BPM could represent a viable solution to implement the so-called
global coordinator (explicit cross-layer approach design). In such a case
suitable primitives should be designed to support cross-layer signaling through
the C-plane.
Fig. 1.12: Protocol manager and BSM protocol architecture [44].
Implicit cross-layer design examples

An example of implicit cross-layer (i.e., joint protocol design/optimization)
could be PHY layer adaptation in the presence of ACM with thresholds
between modes that have been selected to optimize the transport layer
performance [39].
Explicit cross-layer design examples

It is possible to distinguish between explicit cross-layer involving layers across
SI-SAP or involving layers in SD or SI.
An example of the rst method is when cross-layer signaling is provided

through the SI-SAP to connect the management of the layer 3 queues with
that of layer 2 ones (e.g., information regarding the length of the layer 2
queue, used by layer 3). This mechanism is important to support QoS. In
particular, the SI layer requests layer 2 queue status information through a
C-plane primitive (namely, a REQUEST primitive) and the SD layer answers
by means of another C-plane primitive (namely, a CONFIRM primitive). As
already stated, it is possible that BPM manages this information exchange.
Finally, an example of cross-layer information exchange not involving the
use of SI-SAP is that between layer 1 and layer 2. Such signalling can be used
for the MODCOD switching of DVB-S2. In such case, a C-plane primitive is
used to request, to notify or to update information (respectively REQUEST,
INDICATION, and RESPONSE primitives).
1.7 Conclusions
Satellite systems are an attractive solution to provide multimedia commu-

nication services in wide areas of the Earth, also reaching those regions
that lack of terrestrial telecommunication infrastructures. In this framework,
this Chapter has provided an introduction to the features of satellites for
communications, including: orbit types (GEO, MEO, LEO), atmospheric
attenuation phenomena and related packet losses, multiple access schemes
and the air interfaces of main interest for this book (i.e., S-UMTS and
DVB-S/-S2/-RCS).
In this Chapter, a special attention has been also given to the basic aspects
(characteristics, constraints, etc.) related to the management of satellite
resources in S-UMTS and DVB-S/-S2/-RCS systems. Such information will
be essential in the study of the resource management schemes that will be
carried out in the next Chapters, each addressing these techniques from a
dierent prospective. This Chapter has also provided three main scenarios
with numerical details that will be adopted for numerical evaluations in this
book.
Within the research community a range of issues are currently being
investigated that are expected to improve the eciency and the capacity
of satellite communication systems. Towards this aim, this Chapter has
introduced the novel cross-layer approach for the optimized design of the
satellite air interface; many techniques based on this new paradigm will be
described throughout this book.
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2
ACTIVITY IN SATELLITE RESOURCE
MANAGEMENT
Editor: Erina Ferro1
Contributors: Erina Ferro1 , Franco Davoli2 , Petia Todorova3
1
CNR-ISTI - Research Area of Pisa, Italy
2
CNIT - University of Genoa, Italy
3
FhI - Fraunhofer Institute - FOKUS, Berlin, Germany
2.1 Introduction
The ecient exploitation of common resources is an important aspect in

networking, at all protocol layers. In satellite networking, in particular, there
are a number of physical layer issues that have to be addressed in the design
of the system:
Fading
Delay spread
Doppler shift
Limited spectrum
Path loss and thermal noise.
Given these issues, the goal of Radio Resource Management (RRM) is to

optimize bandwidth (capacity) utilization and Quality of Service (QoS), in
the presence of trac ows generated by services with dierent requirements.
Whenever resources or their modications are requested, the goal of RRM is
to optimize request satisfaction and, at the same time, to try to maintain a
44 Erina Ferro
certain degree of fairness among all users.

End-user QoS in satellite/terrestrial networks depends on the QoS achieved
at each layer of the network, based on satellite-dependent and independent
functions to be performed at the layer interfaces. The co-operation of all
network layers from top to bottom, as well as of every network element,
is fundamental. Each layer should use ecient technologies and counteract
any performance degradation factors in order to fulll the user performance
requirements.
As an example of co-operative work, the following actions are considered
in order to optimize system performance.
Bandwidth-ecient modulation and encoding schemes have to be used at

the physical layer, to improve the Bit Error Rate (BER) and the power
level performance under poor weather conditions, such as heavy rain.
Guaranteed bandwidth must be provided at the data link layer by using
ecient bandwidth-on-demand multiple access schemes and by studying
the interaction of mechanisms in the presence of congestion and fading.
The provision of a specic bandwidth to be oered by the physical layer
to the upper layers implies the existence of a bandwidth allocation scheme
that shares the available bandwidth among the dierent user terminals
with dierent trac classes.
The network layer is the lowest layer that deals with source-to-destination
delivery of connection requests (in circuit-switched networks) or packets
(in packet-switched networks); it must know about the topology of the
communication subnet and choose the appropriate paths through it.
Ecient routing policies must be implemented at this level in order to
select paths with the lowest congestion probability. Regarding IP trac
management, user mobility has to be adequately taken into account.
Hence, network layer protocols must provide a prioritized management
for trac coming from users that incur in handover phases (such as
in the presence of non-GEO satellites). Additionally, mechanisms for
IP-layer QoS provision have to be adequately mapped to MAC layer RRM
protocols); indeed, besides considering the protocol layering overhead, the
service capacity to network layer queues is provided by MAC queues that,
generally, are not in one-to-one correspondence to the former ones. This
point will be highlighted in some detail in Chapter 8, Section 3.
At the transport layer, TCP connections, which currently constitute the
bulk of the trac transferred in the Internet, tend to occupy all the
available bandwidth. The nature of most TCP trac is asymmetric,
with data owing in one direction and acknowledgments in the opposite
direction. This translates into dierent bandwidth requirements from the
sender and the receiver, respectively. Bandwidth assignment and link
quality have a strong impact on the TCP goodput.
At the application layer, dierent trac types (e.g., real-time trac and
non-real-time trac) must have specic service level agreements and a
Chapter 2: ACTIVITY IN SATELLITE RESOURCE MANAGEMENT 45
monitoring action has to be performed jointly with the network layer in

order to adaptively modify the service priority.
Several strategies for the optimization of resource management have

been investigated; resource management schemes are strongly related to
the trac. For example, supporting high bit-rate switched trac over the
radio interface and maintaining the QoS requested by single applications put
new requirements on resource management. In addition to the variation in
the demands due to the multimedia trac nature, there are other system
variations that have a strong impact on the adopted RRM technique. These
include changes in the link quality experienced by each terminal due to the
weather conditions, mobility, jamming, and other factors. As a matter of fact,
RRM policies, along with network planning and air interface design, determine
QoS performance at the network level and the individual user level. The RRM
techniques encompass frequency and/or time channels, transmitted power,
and access to base stations. The goal is to control the amount of resources
assigned to each user to maximize some performance indicators, such as the
total network throughput, the total resource utilization and the total network
revenue, or to minimize other indicators, such as the end-to-end delay and the
real-time transmission jitter, subject to some constrains such as the maximum
call dropping rate and/or the minimum signal-to-noise ratio.
The better the RRM technique adopted, the better the performance of
the overall system. It is however clear that the overall performance might be
improved by considering the co-operation of several protocol layers together,
which is commonly called cross-layer approach. In this case, new functions
need to be introduced in the protocol stack to enable interactions even
between non-adjacent protocol layers. In designing a cross-layer architecture
for satellite networks (as in other cross-layer designs), the architectural
implications and the principle of layer separation [1] should be carefully
considered. Relatively few studies have been published to-date on cross-layer
optimization in a satellite context (a recent survey can be found in [2]).
Cross-layer approaches for the satellite environment are deeply surveyed in
Chapter 4 and some numerical results are provided in the following Chapters.
Comprehensive surveys on satellite RRM can be found in [3]-[7]. Reference
[8] provides an account on Call Admission Control (CAC) in the more general
wireless environment. A possible grouping of the RRM techniques in the
literature can be attempted in the following three categories:
1. Frequency/time/space resource allocation schemes (such as channel al-

location, scheduling, transmission and coding rate control, beam and
bandwidth allocation);
2. Power allocation and control schemes, which control the transmitter
power;
3. CAC and handover algorithms, which control the access port connection.
46 Erina Ferro
An overview of the most recent research activities in the RRM eld follows.
Of course, the overview cannot be exhaustive, as new material is continuously
produced.
2.2 Frequency/time/space resource allocation schemes
Papers [9] and [10] treat the RRM subject from the scheduling perspective.
In [9], the authors propose a transmission scheduling method that deals

with the problem of determining Super-frame Length (SL), when allocating
the return channel resources to the capacity requests from satellite interactive
terminals. A main purpose of this method is to minimize the SL in order to
reduce scheduling-wait-time as well as to improve resource utilization. This
method provides great exibility in scheduling, by limiting the SL as much
as possible, and also achieves high resource utilization, by smoothing the
time-varying demands with an overload control.
In [10], the packet-scheduling function has been investigated within the ac-
cess scheme of a unidirectional satellite system, providing point-to-multipoint
services to mobile users. It is interesting how the authors here regard the
satellite system as an overlay multicast/broadcast layer, which complements
point-to-point 3rd Generation (3G) mobile terrestrial networks. The satellite
access scheme features maximum commonalties with the Frequency Division
Duplexing (FDD) air interface of the Terrestrial Universal Mobile Telecom-
munications System (T-UMTS), also known as Wideband Code Division Mul-
tiple Access (W-CDMA), thus enabling close integration with the terrestrial
3G mobile networks and cost-ecient handset implementations. Attention
focuses on one of the radio resource management entities relevant to this
interface: the packet scheduler. The lack of channel-state information and the
point-to-multipoint service set the dierence between the packet scheduler in
the satellite radio interface from its counterpart in point-to-point terrestrial
mobile networks. The authors formulate the scheduler tasks and describe
adaptations of two well-known scheduling disciplines, multilevel priority queu-
ing and weighted fair queuing schemes, as candidates for the time-scheduling
function.
Papers from [11] to [15] address the RRM problem from the transmission and
rate control point of view.
Reference [11] models the Ka band channel by using a Markov process

to capture the impact of the time correlation in weather conditions. A
rate adaptation algorithm is developed to optimize the data rate, based
on real-time feedback on the measured channel conditions. The algorithm
achieves both higher throughput and link availability as compared to a
constant rate scheme. In [12], the authors consider a resource allocation
problem for a satellite network, where variations of fading conditions are added
to those of trac load. Two novel optimization approaches are addressed.
The rst one, considered in more detail in [13], is based on the minimization
over a discrete constraint set, by using an estimate of the gradient, obtained
through a relaxed continuous extension of the performance measure. The
computation of the gradient estimation relies on the innitesimal perturbation
analysis. The second approach adopts an open-loop feedback control strategy,
aimed at providing optimal reallocation strategies as functions of the state
of the network. A functional optimization problem is proposed, and a neural
network-based technique is used in order to approximate its solution.
In [14] and [15], the authors propose an adaptive global strategy, which
handles link congestion and channel conditions in multimedia satellite net-
works. The overall control system also includes CAC, an aspect mentioned
later in this Chapter. However, we include these papers in the present
group, in order to emphasize the presence of adaptive coding. In [15], in
particular, a performance comparison is presented for a xed admission
control strategy versus the new adaptive CAC scheme for a Direct Broadcast
Satellite (DBS) network with Return Channel System (DBS-RCS). The trac
considered includes both Available Bit Rate (ABR) trac and Variable Bit
Rate (VBR) trac. The dynamic channel conditions in the satellite link
consider time-varying error rates due to external eects, such as rain. In order
to maximize the resource utilization, for both xed and adaptive approaches,
assignments of the VBR services are determined based on the estimated
statistical multiplexing gain and other system attributes, namely, video source,
data transmission and channel coding rates.
Papers from [16] to [37] deal with the RRM topic from the bandwidth allocation
point of view.
In interactive satellite networks, the delay between a request and the recep-
tion of the reply is a key issue, due to the basic latency of the satellite link. The
solution oered in [16],[17] for GEO satellites comprises a prediction-based
resource-allocation policy and a scheduling time period as short as possible.
A resource-allocation problem is mathematically formulated as a non-linear
integer programming problem, considering uncertain future trac conditions,
and the author develops a real-time heuristic solution algorithm. Computa-
tional complexity analysis and extensive simulation results demonstrate the
very good performance of the proposed method in terms of computational
eciency and heuristic solution quality.
In [18],[19] the authors propose a scheme for Dynamic Bandwidth Alloca-
tion Capabilities (DBAC) that is not based on classical circuit-switching, but
allows changing the capacity of each connection dynamically without tearing
down and setting up the connection. The analysis of the proposed DBAC
scheme shows a signicant increase in the overall utilization of the capacity,
compared to a plain circuit-switching solution.
48 Erina Ferro
The work in reference [20] focuses on resource allocation and CAC issues
in broadband satellite networks; the authors propose a resource allocation
algorithm that integrates three classes of services at the MAC layer: Constant
Bit Rate (CBR), bursty data, and best eort services. They propose a Double-
Movable Boundary Strategy (DMBS) in order to establish a resource-sharing
policy among these service classes over the satellite uplink channel. DMBS is
a dynamically controlled boundary policy, which adapts the allocation deci-
sion to variable network loading conditions. CAC and bandwidth allocation
decisions are taken at the beginning of each control period, after monitoring
the lling level of the trac request queues. The authors dene a threshold
level for the bursty data request queue in order to regulate the CAC process.
The impact of the queue threshold value on the performance of the DMBS
allocation policy is evaluated. A dynamic variation of this metric is also
proposed to enhance the system response for interactive applications.
Reference [21] provides an overview of Broadband Satellite Access (BSA)
systems, with an emphasis on resource management and interworking tech-
niques to support IP-based multimedia services. Some key innovations are
described, including Combined Free/Demand Assignment Multiple Access
(CF/DAMA) for dynamic satellite bandwidth allocation, and an architecture
for DiServ provisioning over BSA systems. A CF/DAMA scheme for dynamic
satellite bandwidth allocation is also the subject of the work proposed in [22];
this scheme allows the return channel capacity to be eciently shared among
many user terminals.
In [23], the resource allocation problem that arises in the context of a
Medium Earth Orbit (MEO) satellite system with half-duplex communication
capabilities is addressed. MEO satellite systems are characterized by relatively
large propagation delays and intra-beam delay variations, which result in
resource consumption. The authors propose a channel classication scheme,
in which the available carriers are partitioned into classes and each class is
associated with a range of satellite propagation delays.
References [24] and [25] deal with the problem of QoS provisioning for
packet trac. In [24], the authors address the problem by considering a
resource allocation scheme that takes advantage of proper statistical trac
modeling to predict future bandwidth requests. This approach takes into
consideration DiServ-based trac management to guarantee QoS priority
among dierent users. Moreover, the satellite onboard switching problem is
also addressed by considering a suitable implementation of the DiServ policy
based on a cellular neural network.
In [25], the problem of providing guaranteed QoS connections over a
Multi Frequency - Time Division Multiple Access (MF-TDMA) system that
employs Dierential Phase Shift Keying (DPSK) is studied. The problem is
divided into two aspects: resource calculation and resource allocation. The
authors present algorithms for performing these two tasks and evaluate their
performance in the case of a Milstar Extremely High Frequency - Satellite
Communication (EHF-SATCOM) system.
References [26] and [27] present an algorithm for resource allocation in

satellite networks to obtain time/frequency plans for a set of terminals with a
known geometric conguration under interference constraints. The goal is to
maximize the system throughput while guaranteeing that the dierent types
of demands are satised, each type using a dierent amount of bandwidth.
The proposed algorithm relies on two main techniques. The rst generates
admissible congurations for the interference constraints, whereas the second
uses linear and integer programming with column generation.
In [28], the authors consider the problem of how a Geostationary Earth
Orbit (GEO) satellite should assign bandwidth to several service providers
(operators) so as to meet some minimum requirements on one hand, and to
perform the allocation in a fair way on the other. They provide a compu-
tational method to optimize allocation fairness in polynomial time, taking
practical issues into account.
References [29] and [30] consider the problem of allocating the uplink
bandwidth of a satellite transponder among hierarchies of Earth stations, for
guaranteed bandwidth and best-eort trac types. CAC actions are taken
locally at the Earth stations within the allocated bandwidth partition, which
is recomputed either periodically or upon request, by considering dynamic
variations in trac and channel parameters (with a cross-layer interaction
between physical and MAC layers).
The work in [31],[32] proposes a new DiServ-based scheme of bandwidth
allocation during congestion, termed Proportional Allocation of Bandwidth
(PAB). This method can be used in satellite networks based on GEO, MEO,
and LEO (Low Earth Orbit) constellations, in order to transport IP trac
and to provide QoS. In PAB, during congestion, all ows get a share of IP
available bandwidth, proportional to their subscribed information rate.
Reference [33] considers an architecture to interconnect remotely located
heterogeneous terrestrial distribution nodes in a mesh topology, by means
of an onboard regenerative satellite. An emulated DVB-S (Digital Video
Broadcasting via Satellite) regenerative environment is created, by using
an actual transparent GEO satellite. Furthermore, a dynamic bandwidth
mechanism is proposed, to be applied directly on the DVB-S stream of the
uplink of each distribution node. This mechanism enables the provision of
interactive IP-based multimedia services, at a guaranteed QoS.
The work in [34] focuses on dynamic resource allocation algorithms for
sharing the limited uplink resources of a future satellite system among many
bursty users with varying QoS requirements. The data rates provided to
each terminal are selected to dierentiate multiple QoS priority levels, to
provide fairness and to maximize system capacity under time-varying channel
conditions and trac loads.
In [35], Weighted Fair Bandwidth-on-Demand (WFBoD) technique is
dened and analyzed. It is a resource management process for broadband
multimedia GEO satellite systems that provides fair and ecient resource
allocation, coupled with a well-dened MAC-level QoS framework (compatible
50 Erina Ferro
with ATM and IP QoS frameworks) and a multi-level service segregation into
a large number of users with diverse characteristics. WFBoD is also integrated
with the CAC process. Simulation results show that WFBoD can guarantee
QoS for both non-real-time and real-time VBR ows.
A consolidated approach for Voice over IP (VoIP) over satellite networks
based on the ETSI DVB-RCS standard is adopted in [36]. This paper
addresses the role of Bandwidth on Demand (BoD) in the optimization of
VoIP bandwidth allocation, and assesses the impact of BoD mechanisms on
voice quality. The tradeo between voice quality and bandwidth eciency
is investigated under dierent DVB-RCS-specic capacity request/allocation
strategies; it is demonstrated that DVB-RCS provides an ecient platform
for the integrated support of a variety of satellite VoIP applications.
Reference [37] compares BoD in an MF-TDMA environment and Single
Carrier Per Channel (SCPC) from a practical perspective and evaluates the
economical advantages of BoD.
2.3 Power allocation and control schemes

Normally, the literature considers three types of uplink power control tech-
niques [5]:
Open loop. One station receives its own transmission carrier (relayed by the
satellite) and relies on its measurement of beacon fading in the downlink,
in order to perform uplink power control.
Closed loop. Two Earth stations lie within the same beam coverage and
an Earth station can receive its own transmission carrier. Uplink power
control based on this carrier is erroneous due to changes in input and
output backos under uplink and downlink fading. It must be based on
the reception of a distinct carrier transmitted by another station.
Feedback loop. A central control station monitors the levels of all carriers
it receives, and commands the aected Earth stations to adjust their
uplink powers accordingly. This technique has inherent control delays, and
requires more Earth segment and space segment resources.
Regarding downlink, power control allocates additional power to the trans-
mission carrier(s) at the satellite in order to compensate for rain attenuation.
As downlink fading occurs, downlink carrier power degrades and sky noise
temperature seen by the Earth station increases. Power control correction is
required to maintain carrier to noise ratio.
Papers from [38] to [41] treat RRM from the power allocation and control
scheme perspective.
In [38], the authors consider the problem of using narrow transmission

spot-beams on the satellite to support a broad spectrum of users with small
terminals at high rates. Since satellite transmitter resources are expensive and
there can be many spot-beam coverage cells within the satellite service area,
it is attractive to look for some form of agile-scanning beam system and to
time-share these precious resources. An optimized design of both the multi-
beam antenna pattern and the scheduling can further improve the eciency of
transmission and power management. The advantage of parallel multi-beams
in terms of spectral eciency and power gain is shown, and the issue of multi-
beam power allocation based on trac demands and channel conditions over
satellite downlinks with power and delay constraints is addressed. The study
indicates that the use of a parallel multi-beam scheme with optimum power
allocation can achieve a substantial power gain and a reasonable proportional
fairness. By coupling power allocation with multi-beam scheduling when there
are less active beams than cells, the authors show that a modest number of
active parallel beams suces to cover many cells eciently.
In [39], the author analyzes a power-sharing multiple-beam mobile satellite
system in the Ka band with high trac variations from one beam to another.
In order to cope with the multiple-beam varying trac problem, the author
proposes an oset reector antenna, fed through an equal phase-shift active
array. This active array consists of hundreds to thousands of equal phase-shift
elements.
A power allocation policy is developed in [40] for a multi-beam satellite
downlink, which transmits data to dierent ground locations over time-varying
channels. The packets destined to each ground location are stored in separate
queues and the server rate for each queue depends on the power allocated to
that server and the channel state, according to a concave rate-power curve.
A method for satellite network conguration is proposed in [41]. It controls
the transmitted power of multiple Earth stations, and establishes the received
power-dierences among them to generate the capture eect.
2.4 CAC and handover algorithms

This topic is widely treated in Chapter 6. This Chapter only aims at providing
an overview.
Arriving calls are granted/denied access to the network by the CAC
scheme based on predened criteria, taking into consideration network loading
conditions. The trac of admitted calls is then controlled by other RRM
techniques, such as scheduling, handover, power, and rate control schemes.
CAC is extensively studied as an essential tool for congestion control and QoS
provisioning. In terrestrial wireless networks, CAC is more sophisticate than in
cabled networks, due to unique features of wireless networks such as multiple
access channel interference, channel impairments, handover requirements and
limited bandwidth. As in terrestrial wireless networks [8], in satellite networks
there are several reasons for using CAC schemes, including:
52 Erina Ferro
To control the handover failure probability in LEO constellations. Blocking

a new call is surely better than dropping an in-progress call; regardless
of the CAC procedure used, the criterion is maintaining active calls in
progress and blocking new calls that might lead to an increase of the call
dropping probability.
To limit the network trac level to guarantee packet-level QoS parameters
(packet delay, delay jitter and throughput). Some CAC procedures can
estimate packet delay and delay jitter from available resources in multiple-
class networks (see [8] and references [130],[131] therein).
To ensure a minimum transmission rate. This can be achieved either by
limiting network load (see [8] and references [7],[67],[132] therein), by
minimizing the transmission rate degradation (i.e., the condition where
the transmission rate is below a minimum value) (see [8] and references
[128],[133] therein) or by estimating the allocated transmission rate as an
admission criterion (see [8] and reference [101] therein).
CAC schemes can be classied according to various design options [8]

(centralization, information scale, service dimension, optimization, decision
time, information type, information granularity, considered link).
A number of policies have been derived for resource sharing in CAC,
rst for cabled networks, and then for wireless networks in general. The
simplest CAC rule is Complete Sharing (CS), i.e., connections are simply
admitted if sucient resources are available at the time of the request,
without considering the importance of a connection when they are allocated.
In the CS policy, the only system constraint is the overall capacity C. In the
presence of multiple services, this policy may suer from some problems such
as unfairness, in the sense that it can monopolize the resources, it may lead
to poor resource utilization and, nally, it may yield poor long-run average
revenue. As an almost opposite situation, in the Complete Partitioning (CP)
type of policies, every trac class is allocated a set of resources that can be
used only by that class. Other policies have been derived to provide optimized
access to resources, and Ross [42] provides an extensive discussion about a
number of dierent solutions. Actually, optimal approaches should be based
on Markov decision processes, given a certain cost function to be minimized
(or maximized) as a performance index; however, they must take into detailed
account any allowable network state and state transition, which is impractical
even for networks of modest complexity. The functional form of the optimal
policies is usually unknown. Therefore, a set of generally sub-optimal policies
with xed structure (which can be often described by a set of parameters) has
been developed. They are simpler to implement and, in some special cases, do
correspond to the optimal one; among others, we can cite the above mentioned
CP, Trunk Reservation (TR) [43], Guaranteed Minimum (GM) [44], and Upper
Limit (UL) policies [44],[45]. Comparisons have been made between these
policies and the optimal one. The results indicate that CP, TR, GM, and
UL policies outperform the CS one when signicant dierences among classes
exist in requirements for bandwidth and oered load [46]. Obviously, once one
of such xed-structure policies has been selected, parametric optimization can
be adopted in order to choose the best values of parameters that minimize
a given cost function (or maximize a performance index).
As already mentioned, reference [15], besides considering adaptive coding,
also treats the RRM problem from the CAC point of view. This is also done
in [20] and [29],[30], among others. Reference [8] provides an account on CAC
in the more general wireless environment.
In [47], the authors combine CAC with the issue of optimal energy
allocation for communication satellites. The objective is to choose the requests
for transmission to serve so that the expected total reward is maximized. The
special case of a single energy-constrained satellite is considered. Rewards
and demands from users for transmission (energy) are random and known
only at request time. Using a dynamic programming approach, an optimal
policy is derived that is characterized in terms of thresholds. Furthermore,
in the special case where energy demand is unlimited, an optimal policy is
obtained in closed form.
In [48], a real-time trac handling strategy, including distributed CAC and
trac resource management schemes, is harmonized with an in-band signaling
technique for burst-based bandwidth requests and an eective policy for the
allocation of radio resources.
2.4.1 Handover algorithms
In wireless mobile networks, many users share radio bandwidth. An important

property of the network is that user devices change their access points several
times. As their coverage area changes continuously, in order to maintain con-
nectivity, end-users must switch between spot-beams and satellites, and, thus
frequent intra- and inter-satellite handover attempts occur. This fact causes
technical problems, requiring fair sharing of bandwidth between handover
connections and new connections. One of the main problems to be solved
in RRM is the handover management strategy in order to provide low call
dropping probability and to keep high resource utilization.
Several approaches for handover prioritization proposed for terrestrial
cellular systems have been studied in the recent literature for mobile satellite
systems. The solutions include the guard channel scheme, a handover queuing
where the highest priority is oered to handover calls, which are organized in
a separate queue, and novel CAC algorithms, taking into account handover
calls.
In [49], user location information is exploited for adaptive bandwidth
reservation for handover calls. In a beam, bandwidth reservation for handover
is allocated adaptively, by calculating the possible handovers from neighboring
beams. A new call request is accepted if its originated beam has sucient
amount of available bandwidth for new calls.
The key idea of the algorithm in [50] is that bandwidth has to be reserved in
54 Erina Ferro
a particular number of beams S the call may handover into, in order to prevent
handover dropping during a call. The balance between new call blocking and
handover call blocking depends on the selection of predetermined threshold
parameters for new and handover calls.
In [51], a probabilistic resource reservation strategy for real-time services
was proposed, based on the concept of sliding windows to predict the necessary
amount of reserved bandwidth for a new call in future handover beams.
In [52], CAC and handover are based on user location. The system traces all
user locations in each beam and updates user handover-blocking parameters.
Reference [53] proposes an intra-satellite handover management scheme
for LEO satellites, called Q-WIN, specically tailored to the QoS needs of
multimedia applications. This scheme is based on priority queues, combined
with the sliding virtual window concept for call admission. Simulation results
conrm that, compared to the allocation schemes, Q-WIN oers low Call
Dropping Probability (CDP), thus providing for reliable handover of calls in
progress, acceptable Call Blocking Probability (CBP) for new calls and high
resource utilization.
In [54], a guaranteed handover scheme is proposed. According to this
method a new call is admitted in the network only if there is an available
channel in the current cell and, simultaneously, in the rst transit cell. When
the rst handover occurs, a channel-reservation request is issued to the next
candidate transit cell, and so on. If all channels are busy, the request is queued
in a FIFO list, until the next handover occurrence. The call is not forced
to terminate provided that an available channel has been reserved in the
meanwhile.
In [55], dierent queuing policies for handover requests are proposed. The
handover requests, queued up to a maximum time interval (which is a function
of the overlapping area of contiguous cells), are served according to a FIFO
or a Last Useful Instant (LUI) scheme (that is, a handover request is queued
ahead of any other requests already in the queue that have a longer residual
lifetime).
In [56], a novel inter-satellite handover management scheme tailored for
multimedia LEO satellite systems is proposed and evaluated. This scheme
relies on queuing handover requests of dierent service classes in separate
queues. The queue that stores handover requests of real-time services receives
higher priority.
2.5 RRM modeling and simulation

There is a wealth of work on RRM modeling and simulation. References from
[57] to [60] are just a few examples.
In [57], the authors describe the modeling and simulation of an FDMA
(Frequency Division Multiple Access) satellite BoD service. This class of
resource allocation processes, which includes BoD applications, is identied
and compared with common resource allocation processes. Within this class,
the bidirectional and possibly asymmetric nature of resource requests, the
existence of both booked (advance notication) and immediate resource
requests, the allowance of modications to resource requests and the multiple
resource constraints (e.g., bandwidth and power) present unique modeling
challenges. In particular, we can consider three fundamental components:
modeling the resource requests, modeling the fundamental resource allocation
algorithm and modeling the processing of individual resource requests.
In [58], the authors focus on modeling and evaluating the bandwidth
requirements of the next-generation of satellite communication technologies,
which will support future aeronautical applications. The authors interest is
on the real-time delivery of high-resolution weather maps to the cockpit as a
particularly demanding future application. In such scenario, the use of LEO
and GEO satellite networks for ecient data delivery is investigated. The
authors propose a joint uni-cast and broadcast communication technique that
oers bandwidth reduction.
In [59], a new analytical model for equal allocation of divisible computation
and communication load is developed. Equal load allocation is attractive in
multiple processor systems when real-time information on processor and link
capacity, which is necessary for optimal scheduling, is not available. This
model includes a detailed accounting of solution reporting time.
Reference [60] presents a generalized notation as well as graph algorithms
for resource management problems. Impairment graphs can be used for
frequency planning, whereas ow graphs are suitable for channel access
problems. To evaluate the performance of the resource management, service
criteria (such as blocking or Carrier-to-Interference ratio, C/I) or eciency
criteria (bandwidth requirements) are derived from the graphs. The resource
management techniques are applied to satellite networks with non-GEO orbits
that entail time-varying network topologies. As a simple example, the channel
assignment and capacity optimization of the EuroSkyWay system are shown.
For a deeper inspection, a comparison of Fixed, Dynamic and Hybrid Channel
Allocation schemes (FCA, DCA, HCA) for a typical MEO satellite scenario
is provided. The author also investigates satellite diversity and its impact on
bandwidth requirement and transmission quality.
2.6 Related projects in Europe
A number of satellite-related projects have been funded by the European

Commission in both the Fifth and the Sixth Framework Programmes (FP5,
FP6), as well as in COST Actions. In sub-Sections 2.6.1-2.6.4, we limit our
overview to a few FP6 projects. Additional information can be found in
http://cordis.europe.eu.int/en/home.html. Finally, sub-Section 2.6.5 mentions
a recent COST Action and sub-Section 2.6.6 describes a new initiative in the
satellite eld for the FP7 EU programme.
56 Erina Ferro
2.6.1 TWISTER: Terrestrial Wireless Infrastructure integrated

with Satellite Telecommunications for E-Rural applications
http://www.twister-project.net/
TWISTER is a project led by EADS Astrium and was selected for

co-funding by the European Commission in the 1st call for proposals of the
Aeronautics and Space priority of FP6.
This project started on February 1, 2004, and will operate validation
sites throughout Europe for 3 years, through the deployment of up to 105
satellite access points in combination with radio networks. These validation
sites support innovative applications to meet the specic needs of rural user
communities in the domains of agriculture, education, community services,
healthcare and e-business. This project emphasizes usages that benet from
broadband access. The objective of TWISTER is to support the development
and widespread adoption of satellite communication services (like educational
and health care services between islands, or e-business) to deliver broadband
services to rural areas. User satisfaction is evaluated to propose improvements
and to specify a roadmap for further services deployment. The integration of
space-based infrastructure with terrestrial systems aims at achieving a seam-
less broadband coverage in rural areas. TWISTER investigates a number of
hybrid satellite-wireless architectures and validates their on-site performance.
The TWISTER consortium, involving many actors in the telecom value
chain (user communities, service providers, satellite operators, equipment
manufacturers) creates the necessary conditions to deploy successfully such
satellite solutions over Europe as a complement to terrestrial networks for the
benet of the population and the economy.
2.6.2 MAESTRO: Mobile Applications & sErvices based on

Satellite & Terrestrial inteRwOrking
http://ist-maestro.dyndns.org/MAESTRO
The MAESTRO project aims at studying technical implementations of

innovative mobile satellite systems, targeting close integration and interwork-
ing with 3G and beyond-3G mobile terrestrial networks. MAESTRO seeks
to specify and to validate the most critical services, features, and functions
of satellite system architectures, achieving the highest possible degree of
integration with terrestrial infrastructures. It aims not only at assessing the
satellite system technical and economical feasibility, but also at highlighting
their competitive assets on the way they complement terrestrial solutions.
In the frame of the MAESTRO project, innovative and convergent solu-
tions pursue: (i ) the successful and cost eective deployment of 3G multimedia
services over mobile satellite networks; (ii ) the reduction of the digital divide
between urban and rural areas and regions by ensuring service continuity over
heterogeneous GPRS/UMTS networks.
2.6.3 SatNEx: Satellite Network of Excellence
http://www.satnex.org
SatNEx is an FP6 research Network of Excellence (NoE), funded by

the European Commission, which combines the research excellence of 22
major players in the eld of satellite communications [61]-[63]. The pri-
mary goal of SatNEx is to achieve a long-lasting integration of European
research in satellite communications, and to develop a common knowledge
base. This collected expertise will support the European satellite industry
through standardization, collaboration/consultancy and training. Through
co-operation of outstanding universities and research organizations with ex-
cellent expertise in satellite communications, SatNEx is building a European
virtual center of excellence in satellite communications and will contribute to
the realization of the European Research Area (ERA). A dedicated satellite
platform links partners in a broadcast, multicast or unicast conguration,
providing training and video-conferencing capabilities, and promoting the
simplicity and cost-eectiveness of using satellites for this purpose. SatNEx
has established an advisory board incorporating key representatives of the
European space industry, satellite service providers, and standardization and
regulation organizations. SatNEx is steered by these players in providing a
critical mass of resources and expertise, to make Europe a world force in the
eld of satellite communications. Part of the SatNEx mission is to disseminate
internal research and expertise.
2.6.4 NEWCOM: Network of Excellence in Wireless

COMmunications
https://newcom.ismb.it/public/index.jsp
NEWCOM is a European NoE that links in a cooperative way many

leading research groups addressing the strategic objective Mobile and wireless
systems beyond 3G, a frontier research area of the priority thematic area
of IST. This network involves 54 partners from 18 countries, comprising 40
universities and 14 companies. The major objective is a distributed European
university with common research projects and, in the longer term, a shared
doctoral school. This NoE is devoted to the terrestrial wireless environment.
However, some of the research topics, such as cross-layer optimization and
recongurable radio, share common aspects with the satellite world.
58 Erina Ferro
2.6.5 VIRTUOUS: Virtual Home UMTS on Satellite
http://www.ebanet.it/virtuous.htm
The VIRTUOUS project [64], ended in 2002, aimed at identifying, design-

ing and demonstrating a feasible, pragmatic, smooth migration path towards
Terrestrial and Satellite UMTS (T-UMTS and S-UMTS). VIRTUOUS pur-
sued the achievement of the following specic objectives:
Design, development and implementation of both a URAN (UMTS Radio

Access Network ) Radio Technology Independent part and two URAN
Radio Technology Dependent parts, able to handle a terrestrial and a
satellite link, respectively;
Development of two hardware test beds, representative of satellite and
terrestrial UMTS physical layers, respectively;
Denition of the S-UMTS components;
Design, development and implementation of appropriate terminal and
network Inter-Working Units (IWUs), aiming at integrating the GPRS
and the UMTS segments;
Implementation, integration and testing of a demonstrator including three
segments: GPRS, terrestrial UMTS and satellite UMTS;
Trials of meaningful UMTS services, with voice over IP as a candidate
application.
2.6.6 COST Actions
European Co-operation in the eld of Scientic and Technical Research

(COST) is an intergovernmental framework for the co-ordination of nationally-
funded research at European level, based on a exible institutional structure.
Established in 1971, COST has developed into one of the largest frameworks
for research co-operation. The 34 member countries of COST include the 25
EU member states, Bulgaria, Croatia, Iceland, Norway, Romania, Serbia and
Montenegro, FYR of Macedonia, Switzerland and Turkey. Moreover, Israel
is a co-operating state. COST also welcomes Institutions from non-COST
countries to join individual actions for mutual benet. COST networks are
called Actions. Co-operation takes the form of concerted activities, i.e., the
co-ordination of nationally funded research activities. Some of the early
COST actions have helped to pave the way for other European research
programs, such as the EU Framework Programs (launched in 1983) and
the EUREKA initiative (started in 1985; see http://www.eureka.be). COST
plays an important role in scientic and technical co-operation in Europe,
encouraging European synergy and networking and helping further European
integration.
COST covers a wide range of scientic and technological areas: agriculture,
biotechnology and food sciences, chemistry, environment, forests and forestry
products, materials, medicine and health, meteorology, physics, social sciences

and humanities, Telecommunications, Information Science and Technology
(TIST), transport and urban civil engineering.
For more information, the reader may visit the Web site
http://www.cost.esf.org/index.php.
COST Action 272: Packet-Oriented Service Delivery via Satellite
http://www.tesa.prd.fr/cost272/
This COST Action ended in the rst half of 2005 and was entirely
devoted to study aspects related to packet transmission via satellite. The
main objectives of COST Action 272 were the identication of key require-
ments, analysis, performance comparison, architectural design and protocol
specication of packet-oriented satellite communication systems, with a clear
focus on Internet-type system concepts, applications and protocols/techniques
across the various layers. This Action rstly assessed the interesting satellite-
specic market segments and came up with a clearly focused set of reference
scenarios (global/regional, GEO/non-GEO, broadcast/multicast/interactive,
QoS/best-eort, all-IP/hybrid, etc.) as a basis for further research and de-
velopment work, also providing some interesting technical solutions. COST
Action 272 was the continuation of COST Action 253 (Service Ecient
Network Interconnection via Satellite) [65] and the starting point for the
SatNEx Consortium, which elaborated the SatNEx NoE proposal.
2.6.7 The ISI Initiative
http://www.isi-initiative.eu.org/
The Integral Satcom Initiative (ISI) is an open platform, started in 2005,

whose membership embraces all relevant and interested private and public
stakeholders. ISI collaborates and cooperates with the European Commission,
the European Space Agency (ESA), the EU and ESA Member States and
Associated States, the National Space Agencies, International Organizations,
user Fora, and other European technology platforms. ISI fosters international
cooperation under a global perspective. The ISI technology platform brings
together for the rst time in a unied, industry-led forum all research and
technology aspects related to satellite communications, including mobile,
broadband, and broadcasting applications. The purpose is to foster and
develop the entire industrial sector, to maximize the value of European
research and technology development, and to contribute to EU and ESA
policies. The document in [66] species the Strategic Research Agenda of
the ISI technology platform. It addresses the overall development of satellite
communications and satellite broadcasting in Europe till about year 2020.
In doing so, it shows that satellite communications and broadcasting has
60 Erina Ferro
strategic relevance for Europe, and identies medium and long term strategic
objectives. Key research themes of ISI are cited in [66]; among them, RRM
research topics are addressed in various points of the ISI research vision.
In particular: (i ) cross-layer design of RRM techniques, with cross-layer
information coming from adaptive physical layer and QoS requirements from
upper layers, to achieve optimum performance of mobile broadband mul-
timedia satellite services, is one of the key research items; (ii ) advanced
RRM techniques can provide optimum use of the scarce spectrum resource
and contribute to lowering the level of electromagnetic radiation in the
hybrid terrestrial/satellite network environment; (iii ) novel RRM protocols
are considered, which include Medium Access Control (MAC) and Usage
Parameter Control (UPC) mechanisms for the QoS provision under fairness
constraints.
2.7 Conclusions
The goal of RRM is to optimize capacity utilization and QoS in satellite links,
in the presence of trac ows generated by services with dierent require-
ments. The best results are obtained with the cooperation of the protocols
operating at dierent architectural layers, i.e., through a cross-layer approach,
while maintaining the principle of layer separation. A possible grouping of
the RRM techniques in the literature can be: frequency/time/space resource
allocation schemes, power allocation and control schemes, and call admission
control and handover algorithms. For each of these groups, this Chapter
reviews the current results in the literature, even if the survey is far from
being exhaustive.
Some ongoing research projects in Europe that consider the RRM problem
are cited, and the reader is encouraged to visit their Web sites for further in-
formation. Among these projects, the SatNEx Network of Excellence deserves
special attention. It combines the research activities of 22 European institu-
tions, with proved excellence in satellite communications. The realization of
this book has been made possible due to the SatNEx support.
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3
QoS REQUIREMENTS FOR MULTIMEDIA
SERVICES
Editors: Jose Ignacio Moreno Novella1 , Francisco Javier Gonz no2

alez Casta
Contributors: Rafael Asorey Cacheda2 , Daniel Castro Garca3 , Antonio

Cuevas1 , Francisco Javier Gonz no2 , Javier Herrero S
alez Casta anchez3 ,
Georgios Koltsidas , Vincenzo Mancuso , Jose Ignacio Moreno Novella1 ,
4 5
Seounghoon Oh6 , Antonio Pant` o7
1
2
UVI - Universidad de Vigo, Spain
3
INFOGLOBAL, Spain
4
5
UToV - Universit`a degli Studi di Roma Tor Vergata, Italy
6
RWTH - Rheinisch -Westfalische Technische Hochschule Aachen, Germany
7
CNIT - University of Catania, Italy
3.1 Introduction
Internet development and an ever-increasing demand for bandwidth are

boosting the market for satellite solutions. Technological progress leading to
new satellite capabilities and the availability of bandwidth at lower cost is
68 Jose Ignacio Moreno Novella, Francisco Javier Gonz
alez Casta
no
enabling this growing role of satellites in the Internet world. Satellite solutions
are being used for both broadcast/multicast applications and point-to-point
services. End-user access combines multicast and point-to-point services while
content distribution to the edge of the Internet (i.e., to service providers
points-of-presence serving access local loops) is a true multicast application.
Geostationary Earth Orbit (GEO) satellites and Low Earth Orbit (LEO)
constellations essentially play a complementary role, in order to provide this
complete range of services. Due to the large amount of capacity they provide
and their low-latency characteristics, LEO systems are very well suited for
point-to-point high-quality services while GEO solutions are very ecient for
both broadcast/multicast oerings and access services including a signicant
percentage of multicast data. To support the dierent services it is important
to consider their Quality of Service (QoS) requirements.
This Chapter mainly describes QoS requirements for multimedia services
based on international standards. Section 3.2 shows a classication of ap-
plications according to error and delay tolerance, as well as performance
characterization of traditional and multimedia applications. This work is
based on the ITU G.1010 [1] standard that has been adopted by other
standardization bodies like 3GPP. Section 3.3 presents main QoS support
models over IP networks, while Section 3.4 shows main concepts for the
transmission of multimedia and broadcast services over satellite networks.
Finally, Section 3.5 presents experimental results of application performance
over a real platform; the main interest here is to present QoS results on
classical and emerging applications.
3.2 Services QoS requirements
Nowadays it is very important to support QoS in telecommunication systems,

considering the requirements that should be met when a service is provided.
This task should take into consideration that a user is not interested in the
way a particular service is provided, but in the service quality level he/she
nally delectates.
QoS refers to the capability of a telecommunication system to provide
better service to selected trac over heterogeneous networks (technologies or
domains). The primary goal of QoS is to provide priority, including dedicated
bandwidth, controlled jitter and latency (required by some real-time and
interactive trac), and improved loss characteristics. Moreover, it is important
to assure that providing priority for one or more ows does not cause the
failure of other ows. On intuitive level, QoS represents a certain type of
requirements to be guaranteed to the users (e.g., how fast data can be
transferred, how much the receiver has to wait, how correct the received data
is likely to be, how much data is likely to be lost, etc.).
QoS requirements for multimedia trac have been covered by dierent
standardization groups, like ITU, ETSI or 3GPP. The main work provided by
Chapter 3: QoS REQUIREMENTS FOR MULTIMEDIA SERVICES 69
ITU is in Recommendations Y.1541 [2], F.700 [3], and G.1010 [1]. Applications
have been classied in eight groups, according to the error tolerance and delay,
as summarized in Figure 3.1 [1],[4].
Fig. 3.1: End-user QoS categories mapping. This gure is reproduced with the kind
permission of ITU.
Referring to the above Figure, it is possible to consider the following values

on the ordinate axis for what concerns the error rates:
Error tolerant applications

Conversational voice/video Frame Erasure Rate (FER) < 3%
Voice/video messaging FER < 3%
Streaming audio/video FER < 1%
Fax Bit Error Rate (BER) < 106
Error intolerant applications
Information loss = 0.
The ETSI Broadband Satellite Multimedia (BSM) [5] working group pro-
vides technical reports and standards establishing a framework to specify QoS
requirements for broadband satellite networks based on the Internet protocol
suite. These standards (following those developed in ETSI and other bodies)
identify how Internet quality-related standards can be adapted, translated or
made transparent to satellite transmission protocols and equipment. Some of
the results of this standardization work have been the denition of the protocol
stack architecture shown in Chapter 1 (Section 1.5), where lower layers depend
on satellite system implementation (satellite-dependent layers) and higher
layers are those typical of the Internet protocol stack (satellite-independent
layers).
alez Casta
no
The trac classes established by BSM are based on ITU-T, Tiphon,

3GPP, and UMTS decisions, with adaptation to the satellite environment.
In particular, the BSM standards deal with variable link layer conditions,
high asymmetry and higher delay that are characteristics of satellite networks.
The aim is to enable the satellite network and the Internet Service Provider
(ISP) to ensure acceptable QoS levels and to relate these issues to the BSM
architecture for broadband systems.
In UMTS and, by extension, in satellite networks, four basic service classes
(layer 7) are dened [4]: conversational, streaming, interactive and background.
It is interesting to note that there is no strict one-to-one mapping between
these service classes and the namesake trac classes (layer 2) [6]: an interactive
application can very well use a bearer of the conversational trac class, if
the application/service or the user has tight requirements on delay. In the
following sub-Sections the performance requirements for all four service classes
are investigated from the user perspective.
Note that the delay values in the Tables of the following sub-Sections
represent one-way delay (i.e., from originating entity to terminating entity).
3.2.1 Performance requirements for conversational services
The most common service in this category is real-time conversation, such

as telephony speech. Voice over IP (VoIP) and video conferencing also
belong to this category, with increasing relevance as the Internet is rapidly
evolving. This is the only class whose characteristics are strictly determined
by human perception (senses). Thus, this scheme has the most stringent QoS
requirements: the transfer time should be low and, at the same time, the
temporal relation of information entities of the stream should be preserved.
The limit for acceptable transfer delay is very strict (failure to provide low
transfer delays will result in unacceptable lack of quality). However, there
are loose requirements on FER, due to the human perception. For real-time
conversation, the fundamental QoS characteristics are:
Preserving the temporal relation of information entities in the same
stream;
Conversational pattern (stringent and low delay).
Some application examples based on conversational services are: con-
versational voice, videophone, interactive games, two-way control telemetry
and Telnet. Table 3.1 summarizes these applications providing the explicit
requirements for each of them [1],[4].
Conversational voice
Audio transfer delay requirements [3] depend on the level of interactivity of

end-users. To preclude diculties related to the dynamics of voice communi-
cations, ITU-T Recommendation G.114 species the following general limits
Medium Application Degree of Data Key performance parameters

symmetry rate and target values
End-to- Delay Information
end variation loss
one-way within a
delay cell
Audio Conversational Two-way 4-25 < 150 ms < 1 ms < 3% FER

voice kbit/s preferred
< 400 ms
limit
Video Videophone Two-way 32- < 150 ms < 1% FER
384 preferred
kbit/s < 400
ms limit
Lip-
synch:
< 100 ms
Data Telemetry- Two-way < 28.8 < 250 ms NA Zero
two-way kbit/s
control
Data Interactive Two-way < 250 ms NA Zero
games
Data Telnet Two-way < 250 ms NA Zero
(asymmetric)
Table 3.1: End-user performance expectations - conversational services.
for one-way transmission delay (assuming that echo control has been applied)
[7]:
0 to 150 ms: preferred range (below 30 ms the user does not notice any
delay at all, whereas above 100 ms the user does not notice delay if echo
cancellation is provided and there are no distortions in the link)
150 to 400 ms: acceptable range (but with increasing degradation)
Above 400 ms: unacceptable range
We should remember here that there are three types of satellite systems:
LEO, MEO and GEO. Due to their dierent distance to Earths surface, the
propagation delay for the transmitted signal (from Earth to the satellite and
back to Earth) varies from 10 ms to 250 ms (see Section 1.2). This means
that for LEO and MEO satellite systems the preferred range described above
is achievable. However, a GEO system cannot achieve an end-to-end delay
below 250 ms. This means that, according to the satellite system used, the
network designer should be very careful when selecting operational modes.
Other classes have looser requirements and they may be supported by GEO
alez Casta
no
satellites.
The human ear is highly intolerant to short-term delay variation (jitter ),
so it should be kept really low. It has been suggested that 1 ms is an adequate
limit. However, the human ear is tolerant to moderate distortion of the speech
signal. An acceptable performance is typically obtained with FER up to 3%.
Finally, a connection for a conversation normally requires the allocation of
symmetrical communication resources.
Videophone
Videophone requires a full-duplex system, carrying both video and audio, and
it is intended for a conversational environment. Therefore, the same delay
requirements of conversational voice will apply, i.e., no echo and minimal
eect on conversational dynamics, with the added requirement that audio
and video must be synchronized within certain limits to provide lip-synch
(i.e., synchronization of the speakers lips with the words the end-user hears).
In fact, it will be dicult to meet these requirements, due to the long delays
incurred in video codecs. Human eye is tolerant to some information loss,
so that some degree of packet loss is acceptable. It is expected that high
performance video codecs will provide acceptable video quality with FER up
to about 1%. In satellite networks, the same considerations for conversational
voice hold in this case.
Interactive games
Interactive games are games that use the network to interact with other users
or systems. Requirements for interactive games are very dependent on the
specic game considered in terms of bandwidth and delay. Many interactive
games try to exchange high volumes of data, but demand very short delays,
and a delay of 250 ms is reasonable.
Two-way control telemetry
Telemetry is a technology that allows the remote measurement, operation and

reporting of information of interest. Two-way control telemetry is included
here as an example of a data service that does require real-time conversational
performance. Two-way control implies very tight limits on allowable delay and
a value of 250 ms is proposed, but a key dierence with voice and video services
is that information loss cannot be tolerated. It is well known that the satellite
channel is error-prone and in order to achieve zero information loss we need
sophisticate error control techniques to ensure it. Delay is a relative issue for
this class of trac. As far as a satellite network can meet the deadlines that
a particular telemetry service imposes, it can support that service.
Telnet
Telnet (TELetype NETwork ) is a network protocol used on the Internet or
local area network connections. In this context, Telnet refers to the program
that provides the client part of the protocol. It allows a remote server access.
Due to the interactivity of the program, Telnet needs a low delay to allow
a user perception of interactivity. This application is included here with a
requirement for a low delay in order to provide back instantaneous character
echoes. By extension we could consider in the same service/application group
any remote access applications like rlogin (remote login) or ssh (secure shell ).
3.2.2 Performance requirements for interactive services

This second class comprises interactive services (i.e., a human or a machine
request on-line data from a remote server). It is characterized by the request-
response pattern of the end-user. An entity at the destination is usually
expecting a response message within a certain period of time. The Round
Trip propagation Delay (RTD) time is therefore one of the key attributes.
Another characteristic is that the content of the packets must be transparently
transferred (with a low BER). The resulting overall requirement for this
communication scheme is to support interactive non-real-time services with
low RTD.
For interactive trac, the fundamental QoS characteristics are:
The request-response pattern;
Preserving payload content.
Some examples of this service type are: voice messaging and dictation,
data, Web-browsing, high-priority transaction services (e-commerce) and
e-mail (server access). The corresponding requirements are summarized in
Table 3.2 [4].
Voice messaging and dictation

The requirements for information loss are essentially the same as for conver-
sational voice, but, on the contrary, there is more tolerance to delay since
there is no direct conversation involved. Therefore, the main task becomes to
determine the delay that can be tolerated between the user, issuing a command
to replay a voice message, and the actual start of the audio. There is no precise
data on this, but a delay in the order of a few seconds is considered to be
reasonable for this application.
Web-browsing
The main performance factor is the visualization response time, after a Web
page has been requested. A value of 2-4 s per page is proposed. However, a
decrease up to a target of 0.5 s would be desirable.
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One-way Delay Information
delay variation loss
Audio Voice Primarily 4-13 <1s < 1 ms < 3% FER

messaging one-way kbit/s (playback)
<2s
(record)
Data Web-browsing Primarily <4 NA Zero
- HTML one-way s/page
Data Transaction Two-way <4s NA Zero
services - high
priority e.g.,
e-commerce,
ATM
Data E-mail (server Primarily <4s NA Zero
access) one-way
Table 3.2: End-user performance expectatives - interactive services.
3.2.3 Performance requirements for streaming services
This service class is mainly unidirectional with high continuous utilization

(few idle/silent periods) and low time variation between information entities
within a ow. However, there is no strict limit for delay and delay variation,
since the stream is normally aligned at the destination. Additionally, there is
no strict upper limit for the packet loss rate.
For real-time streams, the fundamental QoS characteristics are:
Unidirectional continuous stream;

Preserving time relation (variation) between information entities of the
stream.
The resulting overall requirement for this communication scheme is to

support real-time streaming services with continuous unidirectional data
ows. Table 3.3 details some application examples and the corresponding
limitations [4].
Note that Figure 3.1, Table 3.1, Table 3.2 and Table 3.3 derive from
3GPP TS 22.105 [4]. 3GPPT M TSs and TRs are the property of ARIB, ATIS,
ETSI, CCSA, TTA and TTC who jointly own the copyright in them. They
are subject to further modications and are therefore provided as is for
information purposes only. Further use is strictly prohibited.

Start-up Transport Packet
delay delay loss at
variation session
layer
Audio Speech, mixed Primarily 5-128 < 10 s <2s < 1%

speech and one-way kbit/s Packet
music, medium loss ratio
and high
quality music
Video Movie clips, Primarily 20- < 10 s <2s < 2%
surveillance, one-way 384 Packet
real-time video kbit/s loss ratio
Data Bulk data Primarily < 384 < 10 s NA Zero
transfer/ one-way kbit/s
retrieval,
layout and
synchronization
information
Data Still image Primarily < 10 s NA Zero
one-way
Table 3.3: End-user performance expectations - streaming services.
Audio streaming
Audio streaming is expected to provide better quality than conventional

telephony, thus the packet loss requirements will be correspondingly tighter.
However, there are no conversational elements involved and the delay require-
ments can be relaxed.
One-way video
The main distinguishing feature of one-way video is the absence of conversa-

tional elements. Therefore, the delay requirements will be not so stringent.
Still image
Regarding still images, the human eye is tolerant to information loss. However,
single bit errors can cause large disturbances in still image formats. Therefore,
it is generally expected that there will be zero errors in the transmission of
still images. Delay requirements are low.
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3.2.4 Performance requirements for background

services-applications
This service class applies when the end-user, typically a computer, sends and
receives data les in background. It is a classical data communication scheme
where the destination is not expecting data within a certain deadline. Hence,
the propagation delay (like that of satellite systems) is not that important
in this case. However, error control is very important, since errors should be
kept to very low levels (in the satellite scenario such feature calls for adequate
coding protection and retransmission schemes).
For background trac, the fundamental QoS characteristics are:
The destination is not expecting data before a certain deadline;

Preserving payload content.
The resulting overall requirement for this communication scheme is to

support non-real time services without any special requirement on delay. A
background application has no delay constraint. In principle, an essentially
error-free delivered information is the only requirement for applications in
this category. However, there is still a delay constraint, since data is eectively
useless if it is received too late. Examples of these applications are: fax, low
priority transaction services, e-mail (server to server), Short Message Service
(SMS), download of databases and measurement records.
Fax
Fax is not normally considered a real-time communication. Nevertheless, there

is an expectation that a fax transmission will take less than 30 s.
Low priority transaction services
An example in this category is SMS. An acceptable delivery delay is 30 s.

Table 3.4 compares the applications on the basis of the service class and the
associated delay requirement [8].
3.3 IP QoS frameworks/models
Many factors inuence the user-perceived quality of a telecommunication ser-

vice, from codecs employed to the performance of the network. The constraints
and requirements have been presented in the previous Section 3.2. In this
Chapter we will analyze the mechanisms designed for IP networks to achieve
QoS. This Section addresses the IP layer and as such we keep it very general,
so that the satellite network can be one of the possible scenarios.
It is well known that IP networks were not designed to provide any
Service Conversational Interactive Streaming Background

class (delay 1 s) (delay 1 s) (delay < 10 s) (delay > 10 s)
Error Conversational Voice Streaming Fax
tolerant voice and video messaging audio and video
Error Telnet interactive e-commerce FTP, still e-mail arrival
intolerant games Web browsing image, paging notication
Table 3.4: Application examples in terms of QoS.
This table is reproduced from Radio Resource Management across Multiple Protocol Layers in
Satellite Networks: A Tutorial Overview, P. Barsocchi, N. Celandroni, F. Davoli, E. Ferro, G.
Giambene, F. Casta no, A. Gotta, J. I. Moreno, P. Todorova, International Journal of Satellite
Communications and Networking, Vol. 23, No. 5, pp. 265305, September/October 2005. ISSN:
15442-0973. 2005.
c Copyright John Wiley & Sons Limited. Reproduced with permission.
QoS guarantees. However, the applications traditionally using IP as a com-

munication technology could perfectly cope with that lack; telephony or
iterative applications over IP (that are nowadays beginning to be used) need
transport networks with very strict QoS support. Such mechanisms vary from
100% guarantee solutions (employing techniques that can be assimilated to
virtual circuit creation/provisioning) to other solutions not providing 100%
guarantees. The over-provisioning approach is also considered but, of course, it
cannot be applied in scarce-bandwidth radio access networks. Besides, oering
dierent qualities for the data transport service will create new opportunities
for providing several quality levels at dierent prices. We can conclude that,
in the future, the IP data transport will be QoS-enabled.
The way to provide QoS in IP networks has been discussed for a long
time. The most accepted solutions are IETFs IntServ [9] and DiServ
[10]: both IntServ and DiServ endow the routers with QoS mechanisms,
such as queuing, scheduling and shaping, as illustrated in Figure 3.2. These
mechanisms are implemented in the router interfaces. The main dierence
between IntServ and DiServ lies in the level of detail used by the classiers
and in the need to keep state information.
The IntServ model provides end-to-end QoS guarantees by reserving
per-ow resources (normally using the RSVP protocol [11]) in all the nodes
along the path. While this architecture provides excellent QoS guarantees,
it has scalability problems in the network core because of per-ow state
maintenance and per-ow operation in routers. It is worth noting that RSVP
is not the only IP reservation protocol, but RSVP is by far the most accepted
one and has become an integral part of IntServ networks. There exist even
some commercial RSVP-enabled routers.
RSVP identies a communication session by the combination of destina-
tion address, transport-layer protocol type, and destination port number. In
IPv6 those two last parameters may be replaced by the ow label. RSVP
is used to reserve resources in the routers along the path between the sender
and the receiver(s). RSVP also allows freeing these resources when they are no
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Fig. 3.2: QoS mechanisms in a router interface.
longer needed. Normally these reservations are to be policed and it is common

to have an entity termed bandwidth broker (or, also, QoS broker) that takes
the policy decision and communicates it to the routers. This entity will be
studied later in this Section.
The primary messages used by RSVP are the Path message, which
originates from the trac sender, and the Reservation message, which
originates from the trac receiver(s). They are used in the resource reservation
process. RSVP can also explicitly shut down the QoS sessions using RSVP
teardown messages. Teardown messages can be initiated by an application
in an end-system (sender or receiver) or a router as the result of state time-
out. RSVP supports two types of teardown messages: path-teardown and
reservation-request teardown. Path-teardown messages delete the path state
(deleting the reservation state), travel toward all receivers downstream from
the point of initiation, and are routed like path messages. Reservation-request
teardown messages delete the reservation state, travel towards all matching
senders upstream from the point of teardown initiation, and are routed like
corresponding reservation-request messages.
On the other hand, DiServ requires no per-ow control in the core
network and, consequently, routers do not maintain any per-ow state and
operation; no reservation protocol exists. As a result, DiServ is relatively
scalable in the forwarding/data plane, but oers no strict QoS guarantees.
The criterion to classify the packets in core routers relies on the DiServ Code
Point (DSCP) eld in the packet header [12]. DSCP denes three classes of
priority:
Best Eort (BE): to provide the service in the same way as in the current
Internet, where there are no QoS guarantees, IETF recommends that the
DSCP value should be 000000 (bin).
Assured Forwarding (AF): The AF group contains four independent classes,
each with three dierent drop precedence values in the queues. There is no
specied algorithm for each value, but the dropping probabilities must
be increasing and the packets must be marked with AF DSCP value and
must arrive to the destination in the proper order. In case of congestion,
the dropping probability depends on the drop precedence value.
Expedited Forwarding (EF): EF is designed as the best group. It should
provide very small drop probability, latency and jitter. That is the reason
why this service is sometimes regarded as a Virtual Leased Line (VLL).
This Per-Hop Behavior (PHB) is predestined to handle real-time applica-
tions like video streams. When EF packets enter a DiServ router, they
should be handled in very short queues and quickly serviced to maintain
lower latency, packet loss, and jitter. Throughput of the EF ow should
be limited to the value that can be handled by each node. It is necessary
to avoid the situation where the queue could overow and cause ow
degradation. IETF recommends that the EF class should be marked with
the DSCP value 101110 (bin).
Routers should allocate enough resources for the high priority DSCPs,
while the lower ones or the classical BE trac (DSCP 0) may use spare
resources. DiServ networks require access control in the edge routers, so that
only authorized users can inject packets with high priority DSCPs. Access
control is enforced by the shapers. Depending on the type of edge routers,
this access control can take place in dierent levels of detail. For instance, in
edge routers connecting the core network to the users (Access Routers, ARs)
this control follows a per-user and per-ow basis, since ARs will handle a
small trac load. However, for edge routers connecting the core network to
the Internet or other domains, this access control can only proceed at a very
rough level of detail.
Besides the QoS-enabled routers, another entity called QoS broker [13]
is used to control and to manage the network. This entity, for scalability
reasons, can be replicated in the network; moreover, the network can be
hierarchically divided into several areas, as proposed in [14]. In a simplied
way, the QoS broker manages and monitors the network resources in one
particular domain of operation. It also monitors the edges for incoming and
outgoing resource reservations/utilization. The information thereby acquired
is used in conjunction with the policy system information to take admission
control decisions and recongurations and to convey them to the routers. A
QoS broker is then an entity that takes Service Admission Control decisions
and performs network device conguration, according to a set of conditions
imposed by the network administration entities (e.g., Authentication, Autho-
rization and Accounting, AAA, System) with the goal of achieving end-to-end
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QoS, also over dierent networks. The QoS broker may also be responsible for
managing inter-domain communications with neighbor QoS brokers, so that
QoS-enabled transport services are implemented in a coordinated way across
various domains.
Since IntServ requires resource reservation, it is the most evident scenario
to integrate a QoS broker. In IntServ a RSVP enabled router may consult the
QoS broker (using the Common Open Policy Service, COPS, protocol) about
the decision to take when receiving RSVP path or reservation messages. The
decision taken by the QoS broker is normally conveyed in a COPS message
and then enforced by the router. In DiServ, the edge routers need to perform
admission control and may also outsource the decision to the QoS broker. This
process can take place when the DiServ access router detects a new trac;
the level of detail to dene new trac may vary, as we just explained. QoS
brokers functionally can go beyond taking policing decisions; generally they
are also in charge of managing the network. The actual role of the QoS broker
may adapt to the dierent scenarios and business models. For instance in the
scenario described in Section 3.4.1, the recovery provider may consult a QoS
broker before gathering data from the content providers and sending it to the
satellite so that this broadcasts it.
Many existing approaches combine IntServ and DiServ: IntServ in the
access part of the network and DiServ in the core network. Of course,
solutions based on other paradigms also exist and are even complementary
to these ones. For example, [15] proposes new routing schemes over high
availability networks.
3.4 Broadcast and multicast services
In addition to DVB-S broadcast, satellite IP multicast for content distribution

to the edge of the Internet and to corporate sites has numerous advantages
over terrestrial technology. Satellites oer highly regular broadband data
streams and a single transmission from a central operation center can be
delivered to a high number of receiving sites. In addition to reducing costs,
the single long hop of the satellite link replaces all the small hops of terrestrial
content distribution and bypasses bottlenecks, thus improving QoS in many
applications. Thus, satellite multicast for content distribution and satellite
content delivery to mobile terminals (either broadcast or multicast) are
interesting working areas. Clearly, reception is mainly possible when the
satellite is in direct line-of-sight or attenuation is low. Hence, complementary
terrestrial repeaters enhance the architecture by retransmitting the satellite
signal.
When only a satellite signal is present (i.e., no terrestrial repeaters), satel-
lite broadcasting systems may use time diversity to enhance availability. This
technique broadcasts the same content twice, so that the two transmissions
are uncorrelated with respect to mobile reception blockages. The receiver is
able to combine them to provide seamless reception.

In the case of satellite broadcasting to mobile terminals using mobile
communication modulations, the client could switch between two content
sources with dierent QoS levels: satellite (or terrestrial-repeated satellite) and
terrestrial wireless networks (when neither satellite nor terrestrial repeaters
are available). This handover between physically dierent access interfaces is
problematic for example in the case of UMTS and WiFi (again, the latter
would provide a higher QoS level, at least in terms of regularity, if a satellite
gateway is present).
When terminals support dual network access, e.g., satellite and terrestrial
(WiFi, UMTS, etc.) links, it is quite critical to select the appropriate network
for each application depending on both the available resources and the kind of
application involved. In general, interface selection can be network-initiated
or terminal-initiated. In the rst case, the network operator decides the
appropriate access network for each application, whereas in the second case
the terminal will decide the best path. All these procedures must be performed
during application initialization as well as during handovers, and must con-
sider available access technologies, user prole (SLA, user requirements, etc.),
and QoS capabilities depending on the available resources. In the case of
multicast and broadcast services, terminal-initiated interface selection seems
the natural approach, since it would be too dicult for a network operator to
select individually optimum interfaces for the large user populations involved.
Satellites have traditionally served point-to-point communications (such
as intercontinental telephony circuits) and unidirectional TV broadcast. Very
Small Aperture Terminals, VSATs (i.e., narrowband data terminals in trans-
actional mode), appeared in the 90s. With some exceptions, the medium
access technology at that time neither allowed broadband service provision
nor massive terminal deployment, but 10-to-100 units at most. On the other
hand, equipment manufacturers developed proprietary platforms that could
not interoperate. A high terminal/service cost kept related services within
corporate markets, beyond the possibilities of SMEs. This situation has
radically changed in the last six years, due to technological advances such
as multiple access protocols. On one hand, VSAT terminal manufacturers
(Hughes, Gilat [16], etc.) have developed fully bidirectional equipment (still
proprietary) to provide broadband services to large user communities and, in
some cases (Starband [17], DirectWay), at an acceptable cost even for residen-
tial users. On the other hand, a bit of new manufactures oer interoperable
equipment based on the DVB standard, i.e., specically, MPE (Multi Protocol
Encapsulation) and RCS (Return Channel via Satellite).
The advent in 1997 of the MPE standard for DVB IP data encapsu-
lation implied that equipment manufacturers should no longer supply the
whole communication chain thanks to interoperability. Traditional head-end
manufacturers began to include IP data insertion equipment in their cat-
alogues (Thomcast [18], Divicom, Rohde & Schwarz, etc.), and some new
ones completely centered their eorts in this direction (Logic Innovations,
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Tandberg [19], etc.). In general, they did not provide user terminals, due to
the deep dierences between professional and user markets in terms of quality
goals, sales support, etc. For this reason, many PC peripheral manufacturers
entered the competition with DVB-S boards and boxes (Adaptec, Terratec
[20], Technotrend, etc.).
MPE stimulated the entrance of satellite IP services into the mass market.
For applications requiring interactivity (bidirectionality), the services initially
relied on auxiliary terrestrial technologies for the return channel, wired
(POTS, ISDN or Frame Relay) or wireless ones (GSM, GPRS or similar).
There was a clear lack of a satellite technology to eliminate this terrestrial
dependence. In 1999, the DVB-RCS standard covered this gap. Despite of
some initial interoperation problems (usually leading to the election of the
same supplier for the whole communications chain), the standard has matured
in the last years. Several operators have selected it (Satlynx, Hispasat [21],
etc.).
In the last two years the new protocol DOCSIS for Satellite (or DOCSIS-S)
is emerging as an alternative to DVB-RCS, based on the well-known DOCSIS
standard for cable networks and mostly promoted by American vendors
and providers (Viasat [22] and WildBlue [23]). Compared with DVB-RCS,
DOCSIS-S exploits the economies of scale of silicon designs for cable infras-
tructure, and takes advantage of a huge selection of Operations and Business
Support Systems platforms from the cable market. However, DOCSIS-S is still
a vendor-promoted protocol, not a real standard; thus interoperability and
availability of suppliers are an issue.
These new protocols enable new multimedia application scenarios based
on multicast and broadcast distribution. One of these applications is distance
learning with or without interactivity. In it, a teacher provides a lesson to a
number of remote students using multicast video and audio streaming and
additional aids such as a digital blackboard, slides, etc. When interactivity
(return channel) is available, students may send questions to the teacher either
by chat or by their own microphone and webcam, so that the other students
may follow the question and the response. In this case, because of the delay
induced by the satellite itself (500 ms for a GEO system), the media access
protocol for the return channel (100 - 300 ms) and the video codecs (100 -
1000 ms), a voice handshake similar to a walkie-talkie must be implemented
in order for the teacher and the student not to interfere. Also, when there
is a large audience, the application must provide specic controls so that
the teacher may act as moderator, granting or denying participation to the
students. At present, distance learning systems (Centra [24]) and services
(Hughes [25], Gilat [16]) are commercially available and widely deployed.
Another common multicast application not requiring real-time operation
that largely benets from a return channel when available is massive content
distribution, where a central station delivers common multimedia contents to
a large population of remote clients (with a reception acknowledge mechanism
when interactivity is provided). The typical data losses and corruptions are
avoided by a) adding redundant information to the data to be transmitted at

the application level by means of convolutional coding, polynomial protection
and interleaving, and b) implementing a return channel so that each remote
client may inform the central station about the missing parts of the media
content after receiving them and correcting the errors. Then, the central
station re-sends those pieces of data grouped in overlapped parts, to avoid
repeating the same datagrams. Massive content distribution software solutions
are available from Kencast [26] and Tandberg [19], among others.
The DVB-RCS standard enables other innovative application scenarios
for satellite content delivery to mobile terminals, such as Delayed Real-Time
(DRT) services with QoS support for GEO satellite distribution systems. We
describe them in the next sub-Section.
3.4.1 Delayed real-time service over GEO satellite distribution

systems
The distribution of multimedia contents via satellite, even though it is one of

the very rst services envisioned by the satellite communication community,
still represents a hot topic for satellite networks. There are many types of
multimedia communication services; in this sub-Section, we address the class
of DRT services, whose importance arises in the eld of QoS-aware real-time
communications.
DRT services fall in the category of streaming services whose requirements
are discussed in sub-Section 3.2.3. DRT services have been conceived as an
extension of unidirectional real-time broadcast and multicast services. So far,
there are no standard architectures to support DRT, but diverse applications
have been proposed in order to cope with given QoS requirements by means
of specic application layer mechanisms. Instead of limiting DRT support to a
mere application layer implementation, this Section presents an architecture
that exploits both application and transport layer features. The proposed
architecture assumes that DVB-RCS is deployed over a GEO satellite system.
Nonetheless, it can be easily extended and adapted to any other layer 2
protocol stack suitable for broadcast and multicast applications, allowing
customers to interact in real-time with the multimedia distribution farm (e.g.,
WiMAX or UMTS technologies).
A DRT service recovers from data losses and corruptions by using a
buer and, in turn, by introducing an articial delay at the beginning of the
play-out phase. A real-time return channel is fundamental, since the receiver
must initiate a data recovery procedure after a data loss has been detected.
In that case, additional resources can be invocated over the distribution
channel, if available. The maximum possible duration for each recovery phase
is determined by the length of the adopted buer, and can be modulated by
the choice of the codec (or codecs) for the multimedia streaming.
It is worth noting that multiple retransmissions could be requested at the
same time by dierent users (e.g., by users belonging to the same multicast
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group), and, therefore, dierent retransmissions could partially overlap. Ac-

cordingly, retransmissions are executed in multicast and are requested through
dynamically joining and pruning the multicast retransmission group. As a
consequence, it is possible to design an architecture where a legacy satellite
broadcast service is endowed with a specic multicast recovery algorithm able
to mitigate the impact of network/satellite disruptions. This is the case of
link failures due to user mobility and related shadowing eects. The reference
scenario (Figure 3.3) is composed of three main elements:
The Content Provider (we assume to have N content providers in the
network);
The Recovery Service Provider (just one in the network);
The users (specically, N groups of users, one group for each active content
provider).
Fig. 3.3: DRT service architecture.
The Content Providers are the primary sources for video applications,
i.e., they generate the real-time data. We can suppose that a content provider
is located just before the satellite hop or, more generally, that the Internet
spreads between them.
The Recovery Service Provider consists of a streaming proxy that has
access to satellite resources and manages the retransmission priority. In fact,
retransmission requests can be listed according to a priority that is related to
the time constraints of the recovery phase, but also to the type of service
and the customer class the service pertains to. It is worth noting that
retransmission requests can be rearranged in time by the proxy, based on

a metric that quanties the importance of a data segment for a requesting
customer, so that a simple FIFO scheduling of retransmissions is far to be
optimal in terms of fairness, throughput and user satisfaction degree.
The user is actually to be considered as a set of customers (Group 1, Group
2, ... , Group N ) located behind the satellite link, whose applications share
some common bandwidth resources. Optimizing the usage of those resources
is one of the goals of the envisaged architecture.
3.4.2 Scenario characterization and results
Every content provider sends a multimedia stream over the satellite link
using a guaranteed bandwidth. According to Figure 3.3, there are N content
providers and, therefore, N statically allocated channels. Data are sent to the
streaming application after a playout delay (e.g., D seconds). Each receiver
uses a local proxy buer to store at most D seconds of streaming data,
i.e., data to be played within D seconds. This elastic buer , that empties
at constant rate and lls at variable rate, permits to continue the playout
during the satellite channel outage, as long as sucient information has
been previously stored in the buer. When a channel outage happens, the
receiver (i.e., the proxy located at the receiver group) leaves a blank space
in the application buer and, when the channel is again available, sends a
retransmission request to a Recovery Service Provider (RSP), in order to ll
the hole in the elastic buer. All the retransmissions use a shared channel, e.g.,
the (N +1)-th channel. We propose that, in this recovery channel, content
providers retransmit the packets using a transport protocol with the Additive
Increase Multiplicative Decrease (AIMD) scheme [27],[28]. In particular, the
number of packets a sender can put on the network is limited by a congestion
window (cwnd ) that is managed as follows:
Slowly (additively) increase the cwnd size as long as there is no congestion.

Typically, the cwnd is increased by one packet for each window sent
without a packet drop (in practice, cwnd = cwnd + /cwnd as each ACK
returns, with = 1).
Quickly (multiplicatively) decrease the cwnd size as soon as congestion is
detected. Typically, cwnd is halved for each window containing a packet
loss (cwnd = /cwnd, with = 0.5).
In this way, the available bandwidth is fairly shared. After receiving a

retransmission request, the RSP (which acts like a proxy for on-demand
services) classies the request according to the run-time estimated urgency.
The urgency is calculated from the information requested and the time
available for recovery purposes. Correspondingly, every user communicates
a time interval and two timestamps conveyed by the retransmission request:
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T, [t0 , t1 ] (3.1)
where t0 is the time when the broadcast connection became unavailable

for the requesting receiver, t1 is the time when the data to be retransmitted
should be used by the multimedia player, and T is the data window that
is requested, i.e., the room to be lled in the receiver buer, in playout
seconds.
The RSP assigns a proper bandwidth to each retransmission, which is
calculated from the corresponding urgency. The policy that determines the
urgency of a request is based on both the dierence (t1 -tcurrent ) and T,
i.e., the intervals available to start and to complete the recovery procedure.
This means that the urgency of a retransmission may change during the
retransmission itself, so that bandwidth assignments have to be dynamically
adjusted. Possibly, a policy function might run on the retransmissions codec,
trying to accommodate multiple requests in the same channel.
Once the codec has been selected for a retransmission, the amount B of
data to be sent is determined, and the following formula is used to compute
the AIMD transmission parameters and :
B = r(, ) (t1 tcurrent ) (3.2)
where B is the amount of data to send at time t1 and r is the rate to be

achieved by means of an opportune choice of and .
A formula is shown in [29] that correlates the AIMD mean sending rate r
with the control parameters, and , the loss rate p, the mean Round Trip
Time, RTT, the mean timeout value, T0 , and the number b of packets each
ACK acknowledges:
1
r(, ) = (3.3)
T D, + T O,
where:

2b(1 )
T D, = RT T p (3.4)
(1 + )

(1 2 )b
T O, = T0 min 1, 3 p p(1 + 32p2 ) . (3.5)
2
Thus, from the bandwidth value, the proxy calculates and parameters
of the AIMD transport scheme, which will be communicated to every content
provider that has to retransmit data.
Here we modeled the link with a good-bad process with exponentially dis-
tributed permanence times for both good and bad states. Real-time broadcast
applications are always on, with a xed bandwidth usage. Also the bandwidth
available for retransmission is xed and guaranteed by the distribution sys-
tems, and the playout delay of each receiving application is the same for
all users. Furthermore, we represent each multicast group with a single user
that acts as the worst-case user, so that the good-bad process actually refers
to the time distribution of periods in which link failure occurs or not, for
an entire multicast group. This assumption simplies the simulative analysis
while preserving the correctness of results; in fact, in our system, overlapping
retransmission requests sum and turn into a single multicast retransmission.
Finally, no codec adaptation has been considered.
As for the transport protocol, we have tested UDP-like retransmissions
(the evaluation of TCP and AIMD-like protocols will be considered in a future
study). However, preliminary results obtained with UDP, justify the study of
connected transport protocols to enhance system performance.
As a reference, let us consider a scenario with N = 10 Content Providers
generating an aggregate of 20 Mbit/s (each Content Provider generates at a
xed, but dierent rate of about 2 Mbit/s, to avoid synchronization eects),
and a 6 Mbit/s bandwidth is guaranteed for recovery. The playout delay of
users is 20 seconds, and the transport protocol is UDP. The average duration
of the bad state of each link has been set to 5 s; we have obtained the results by
changing the average duration of the good state and by collecting simulation
results over 200000 seconds.
Figure 3.4 shows the aggregate amount of retransmitted data when the
adopted retransmission priority is proportional to the bandwidth of the
real-time stream. Curves are normalized to the aggregate number of bytes
requested by users. The lower curve in the Figure represents data retransmit-
ted for retransmissions that the system was able to complete. It is clear that
a great number of retransmissions is stopped due to lack of resources as soon
as the link error probability exceeds 0.2. Furthermore, for error probability
greater that 0.1, the number of unrecoverable bytes increases (due to outage
periods longer that the playout delay, which are now more frequent).
For the same scenario, Figure 3.5 depicts the aggregate delivered data
and the amount of data lost due to link failures during the retransmission
procedure. Lost data are normalized to retransmitted data and not to re-
quested data, to give a correct measure of the needs of a connected transport
protocol during the recovery procedure. Note that system performance is not
satisfactory even with values of the link failure probability as small as 0.1,
which is not so much for users.
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Fig. 3.4: Retransmitted data using a retransmission priority proportional to the

required bandwidth.
Fig. 3.5: Delivered and lost retransmitted data using a retransmission priority
proportional to the required bandwidth.
3.5 Experimental results on QoS
Many of the application QoS requirement studies have been done in current-
day Internet networks, for instance many of the considerations shown in
Section 3.2. The aim of this Section is to describe the work carried out in
a Next-Generation Network (NGN) prototype to characterize the application
QoS requirements in such a kind of network. Results refer to real experiments
on application behavior.
The test bed was an all IPv6 network; Figure 3.6 depicts the network
architecture. Two access technologies, one wired (Ethernet) and one wireless
(IEEE 802.11), where employed. This can represent a subset of all the access
technologies a future network operator may oer to its customers. Users,
employing the appropriate devices could connect to any of the two networks. In
the test bed, wireless connectivity is assured using commercial SMC WLAN
cards with prism driver. Satellite links were not available in our test bed due
to the complexity and high costs in using these links for experiments. We
however believe that the obtained results may provide good insights also for
general networks (including satellite links) in particular for what concerns
the characterization of application behavior in NGNs with features such as
mobility or QoS, using IP as convergence layer.
Fig. 3.6: NGN prototype test bed.
Our network was divided in 2 parts: (i ) an access part where the users
connect to via either Ethernet or WLAN (i.e., WiFi); (ii ) the core network.
The latter is connected to the 6 bone (IPv6 Internet) via an Edge Router
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and to the access part and the users terminals via the Access Routers. The
core network hosts two servers supporting dierent functionalities of NGNs.
These functionalities include aspects that should be present in next-generation
commercial mobile networks, such as user authentication and accounting;
mobility and QoS management were also controlled by these servers. All the
nodes (including the routers) are general-purpose machines (Pentium III and
IV PCs). All run Red Hat 7.2 with Linux-2.4.16 kernels. More details about
the test bed can be found in [30].
QoS is based on DiServ with access control. This access control is
performed on the Access Routers on a per ow and per user basis. The Access
Router outsources the admission decision to the QoS broker, an entity located
in the core network able to take this decision and congure the routers with
appropriate parameters.
The test bed here described is composed of general-purposes machines and
it is just a mere representation of what a next network infrastructure may be,
but we believe that the results obtained in it can provide us early and valuable
hints about the applications specic QoS requirements when using NGNs.
We performed on-site real measurements of end-user performance percep-
tion and application characterization under dierent situations that can be
present in NGNs, as detailed in [30] and [31].
The tested applications correspond to conversational services and interac-
tive services. All of them were IPv6 applications. Conversational services were
provided by Robust Audio Tool (RAT) for conversational voice and Quake 2
and Tetrisnet for games. Again, for interactive services we employed RAT
(for audio streaming) and VideoLan for video streaming. Conversational and
interactive services characterization was already described in Section 3.2; the
added value of this Section is to show experimental studies obtained in an
NGN prototype and check the dierences.
Two kinds of tests were performed: the rst was intended to characterize
application behavior in terms of bandwidth needs (including burstiness); the
second one experimented with user tolerance to delay, jitter and packet loss.
We will show and analyze the results; the tests methodology is further detailed
in [30].
For the rst type of tests, ethereal [32], a network analyzer software,
was used to capture the packets and tcpstat was adopted to analyze the
application trac. Two parameters were evaluated: packet size and packets
per second. Mean, min, max, deviation and deviation/mean values were
calculated for those two parameters. First, the results are presented and then
some conclusions drawn. Audio stream has constant packet size and very small
variation in packet rate. For video stream we have a nearly constant packet
size and a small variation in packet rate. For conversational applications the
results are as follows:
Conversational voice presents a constant packet size, but also a high

variation in packet rate.
The Tetrisnet game generated a very low trac, but with great variation
in packet size and rate.
Quake 2 generated more trac and also had remarkable variations in
packet size and a small variation in packet rate.
As a general conclusion, interactive applications have a higher bandwidth

variation since they depend on user behavior: there is silence suppression,
thus when the user does not talk no packets are sent. Moreover, Quake 2
bandwidth consumption depends on user activity: the more it interacts the
larger the packets are, because more information needs to be sent (packets are
sent at a rather constant rate). The bandwidth of the streaming application
does not depend on user behavior, but only on the nature of scenes and
audio. Obviously, the employed codecs play a fundamental role in determining
application bandwidth consumption.
The results are as expected and similar to the ones obtained in the current
Internet. However, there are some remarkable aspects worth to mention.
For instance, mobility and overhead. Mobility in NGNs will be based on
Mobile IP (MIPv6). This means adding, to the basic IP header the IP
home address header and, also, generally the IPv6 routing header. For
conversational applications with only audio, the payload is small and, as
such, the ratio payload/overhead becomes very small. We also found NGNs
specic results when dealing with applications adaptability. In NGNs, the
users will roam between several access technologies with dierent performance
characteristics. Applications should be able to cope with this heterogeneity
adapting themselves, for instance in layered video, sending only detailed
layers when the available bandwidth is high, for instance in downlink satellite
links.
As aforementioned, the second type of tests evaluated user-perceived
quality. NIST Net [33] is the software that can alter network conditions. It
was employed to generate packet loss, delay and jitter in the test-bed network.
Since NIST Net works only on IPv4 networks and the test-bed infrastructure
was pure IPv6, a tunnel was set up. Table 3.5 presents the results. These
results were as expected: conversational applications (Tetrisnet, Quake 2, and
VoIP) have more strict requisites for delay and jitter. Tetrisnet is an exception,
since it is an interactive application, but interaction speed is rather small (in
the order of a second) so that delay requirements are very loose.
Application Packet loss (%) Delay/Direction (ms) Jitter/Direction (ms)

Audio Stream 2 > 500 100
Quake 2 15 100 150
VoIP 10 150 50
Tetrisnet 20 > 500 > 500
Table 3.5: QoS requirements as measured in the NGN prototype.

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The obtained requirements are similar to those presented in Section 3.2 for
nowadays networks. The specic aspects of NGNs can be found mainly in the
fact that network QoS is priced and tailored for the users. As such, we found
that low prole users, paying less for the transport service where much more
tolerant with their requirements. Besides, for some users, more than having
better QoS, the important aspect was the unique NGN ability of supporting
all kinds of applications and having seamless inter-technology handovers with
the capability of taking the best prot from the available access technologies.
3.6 Conclusions
This Chapter stressed on the importance of providing QoS for data transport.
Some applications have stringent QoS requirements, mainly related to delay
and jitter. Satellite networks may suer from too high delays so QoS aspects
should be considered very carefully. On the other side, satellite networks
are very well suited for multicast and broadcast transmissions as well as
for DRT services. For about 6 years now, satellite networks are also a
commercial solution for completely dierent scenarios: unicast bidirectional
services like broadband Internet access. These scenarios, requiring strong QoS
requirements, need a careful analysis and the implementation of mechanisms
to support QoS as discussed in the next Chapters of this book.
References
[1] ITU-T Recommendation G.1010: End-user multimedia QoS categories, URL:

http://www.itu-t.org.
[2] ITU-T Recommendation Y.1541: Network performance objectives for IP-based
services, URL: http://www.itu-t.org.
[3] ITU-T Recommendation F.700: Framework Recommendation for audio-
visual/multimedia services, URL: http://www.itu-t.org.
[4] 3GPP, Technical Specication Group Services and System Aspects Service
aspects; Services and Service Capabilities, TS 22.105 V6.0.0 (2002-09) (Release
6), URL: http://www.3gpp.org.
Multimedia (BSM) services and architectures; Functional architecture for IP
interworking with BSM networks, TS 102 292, V1.1.1 (2004-02).
[6] 3GPP, QoS Concept and Architecture, TS 23.107, URL:
http://www.3gpp.org.
[7] ITU-T Recommendation G.114: One-way transmission time, URL:
http://www.itu-t.org.
[8] P. Barsocchi, N. Celandroni, F. Davoli, E. Ferro, G. Giambene, F. Casta no, A.
Gotta, J. I. Moreno, P. Todorova, Radio Resource Management across Multiple
Protocol Layers in Satellite Networks: A Tutorial Overview, International
Journal of Satellite Communications and Networking, Vol. 23, No. 5, pp.
265-305, September/October 2005. ISSN: 15442-0973.
[9] R. Braden et al., Integrated Services in the Internet Architecture: an
Overview, IETF RFC 1633, June 1994.
[10] S. Blake et al., An Architecture for Dierentiated Services, IETF RFC 2475,
December 1998.
[11] R. Braden et al., Resource ReSerVation Protocol (RSVP) - Version 1
Functional Specication, IETF RFC 2205, September 1997.
[12] K. Nichols et al., A Two-Bit Dierentiated Services Architecture for the
Internet, IETF RFC 2638, July 1999.
[13] K. Nahrstedt et al., The QoS Broker, IEEE Multimedia, Vol. 2, No. 1, pp.
53-67, Spring 1995.
[14] G. Cortese et al., CADENUS: Creation and Deployment of End-User Services
in Premium IP Networks, IEEE Communication Magazine, Vol. 41, No. 1, pp.
54-60, January 2003.
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[15] G. Schollmeier et al., Providing Sustainable QoS in Next-Generation

Networks, IEEE Communications Magazine, Vol. 42, No. 6, pp. 102-107, June
2004.
[16] Gilat Satellite Networks, URL: www.gilat.com.
[17] StarBand, URL: http://www.starband.com/.
[18] Thales Broadcast and Multimedia, URL: http://www.thomcast.com/.
[19] Tandberg, URL: http://www.tandbergtv.com.
[20] TerraTec Electronic GmbH, URL: http://www.terratec.net/.
[21] Hispasat, URL: http://www.hispasat.com.
[22] Viasat, URL: http://www.viasat.com.
[23] Wildblue, URL: http://www.wildblue.com.
[24] Centra, URL: http://www.saba.com/centra-saba/.
[25] Hughes, URL: http://www.hughes.com.
[26] Kencast, URL: http://www.kencast.com.
[27] E. Altman, C. Barakat, V. Manuel Ramos, Analysis of AIMD Protocols over
Paths with Variable Delay, INFOCOM 2004.
[28] L. Cai, X. Shen, J. W. Mark, J. Pan, A QoS-Aware AIMD Protocol for
Time-Sensitive Applications in Wireless/Wired Networks, in Proc. of IEEE
Infocom05, Miami, Florida, March 13-17, 2005.
[29] Y. R. Yang, S. S. Lam, General AIMD Congestion Control, University of
Texas, Tech. Rep. TR-2000-09, May 2000.
[30] P. Serrano et al., Medida y analisis del tr
aco multimedia en redes m
oviles de
cuarta generacion, Telecom, I+D 2004, Madrid.
[31] A. Cuevas et al., Usability and Evaluation of a Deployed 4G Network
Prototype, Journal of Communications and Networks (ISSN: 1229-2370), Vol.
7, No. 2, pp. 222-230, June 2005.
[32] Ethereal: A Network Protocol Analyzer, URL: http://www.ethereal.com/.
[33] NIST Net Home Page, URL: http://snad.ncsl.nist.gov/itg/nistnet/.
4
CROSS-LAYER APPROACHES FOR
RESOURCE MANAGEMENT

Editor: Mara Angeles Vazquez Castro1
Contributors: Franco Davoli2 , Erina Ferro3 , Giovanni Giambene4 , Petia

Todorova5 , Mara Angeles Vazquez Castro1 , Fausto Vieira1
1
UAB - Universitat Autonoma de Barcelona, Spain
2
3
4
5
4.1 Introduction
The enormous advantages of physical layer adaptivity for adequate operation
of wireless systems over widely-varying channel conditions have been widely
proved. However, an optimal adaptation strategy for a given set of resource
constraints requires a joint optimization across layers. Such a cross-layer
optimization is becoming a new paradigm for wireless system design, which
can be extraordinarily complex as the number of optimization parameters and
layers grows.
In this Chapter, we present a comprehensive literature survey of existing
cross-layer design approaches for resource management optimization in order
to draw some preliminary conclusions on adaptive satellite systems.
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Mara Angeles V
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4.2 Literature survey on cross-layer methods
Fade Mitigation Techniques (FMT) allow for adaptation to the dynamics

of the physical system, thus introducing a new concept in system design,
no longer based on worst-case behavior. Three dierent FMT types can be
distinguished (see for instance [1]), each of them introducing a diverse degree
and nature of adaptivity: power control techniques, diversity techniques and
adaptive waveform techniques.
A conventional protocol stack employs independent design of protocol
layers, thus precluding adaptation of the system to changing conditions. Cross-
layer optimization oers a new paradigm for the design of next-generation
wireless networks. As satellite-based systems evolve towards Internet-centric
networks, system adaptivity poses new challenges; for example, dynamic
resource management to provide the dierent QoS requirements and Service
Level Agreements (SLAs), suitable for multimedia.
Cross-layer methods provide a natural solution to the challenges of adap-
tation to both system dynamics and the demands of highly dynamic appli-
cations. In order to optimize the overall performance, the joint adaptation of
several layers must be coordinated, requiring a new cross-layer framework
to be designed and standardized. It is important to realize that dierent
communities have somewhat diverse perspectives on cross-layer optimization.
For instance, the networking community has proposed developing protocols
and mechanisms to adapt the network to the applications. Conversely, the
video community has suggested adaptation of the source coding to the
network, since Shannons separation theorem does not apply to general
time-varying channels, or to systems with a complexity or delay constraint.
At the satellite-dependent layers (i.e., physical and MAC layers), there are
proposals to adapt the radio resource management to pre-dened trac
proles and to changing propagation conditions. In general, cross-layer design
involves interactions among ve key protocol layers: application layer (includ-
ing presentation and session), transport layer, network layer, link (MAC) and
physical layer.
A cross-layer approach requires the introduction of new control functions
in the protocol stack in order to enable interactions between non-adjacent
protocol layers. This is in itself an important topic of research and one that
is currently not well understood in the general case. Initial solutions are
therefore likely to be oriented for ad hoc optimizations for specic protocol
stacks and may be suited to only a small number of system scenarios. Once
the approaches are well understood, future work may seek to generalize the
primitives and control exchanges.
In designing a cross-layer architecture for satellite networks, care must
be taken to consider the implications and the principle of layer separation.
In particular, it is important to dene the extent to which parameters at a
lower (e.g., physical) layer should inuence control strategies at higher layers
(e.g., network QoS, transport reliability, application data format) [2]. This
Chapter 4: CROSS-LAYER APPROACHES 97
may be dependent on the specic environment and on the type of control

exerted on the system. Separation principles (which are also related to time
scales) may be adopted in adaptive hierarchical control systems, whereby
tighter (regulatory control) actions are taken at lower layers, and their eect
is perceived through aggregate parameters. However, especially in satellite
systems, the presence of protocol enhancing proxies with specic protocol
stacks may mitigate the potential negative eects of cross-layer interactions
on the network as a whole.
The cross-layer protocol design entails a protocol stack optimization on the
basis of novel interactions even between non-adjacent protocol layers. Due to
the specicity of the optimization process, the cross-layer design should be
suitably tailored for each examined protocol stack and systems scenario. In
particular, among these scenarios, we may consider two most signicant cases:
(i ) DVB-S/-RCS (or DVB-S2) -based systems for GEO-based broadband
communications; (ii ) S-UMTS systems for GEO or non-GEO-based commu-
nications to mobile users.
In the following paragraphs, a preliminary literature survey is provided in
order to illustrate the available cross-layer methods. The dierent proposed
cross-layer approaches have been categorized according to the layers or layered
functionalities that are jointly optimized.
Joint PHY/MAC optimization
In [3], the authors provide a cross-layer optimized design of the MAC layer
under Rayleigh fading, based on a Markov chain formulation. System in-
formation and physical layer measurements are jointly considered with the
intention of maximizing the overall throughput. In [4], a discussion on protocol
harmonization for MAC and physical layer for IEEE 802.11 is addressed.
The authors investigate the eects of packet length, transmit power and
bit-error rate. Their results show that minimum energy is consumed for an
optimal transmission power, which is proportional to the packet length. In
[5], the joint eects of nite length queuing at MAC layer and adaptive
coding and modulation are analyzed. The performance gain is quantied
when applying cross-layer design to maximize throughput. In [6], the authors
describe the ow of information between PHY and MAC layers in order to save
power and to improve overall performance via an adaptive distributed MAC
(uplink) protocol. Several authors propose link layer adaptation to reduce the
transmission errors based on current channel conditions. In [7], around 50%
improvement in goodput and 20% improvement in transmission range is shown
to be obtained by using the optimal Maximum Transfer Unit (MTU) for a
particular BER. In [8], it is shown that an 18-25% throughput gain may be
obtained by increasing the frame length, depending on radio conditions. In [9],
the authors focus on the cross-layer optimization of the scheduling policies to
assure queuing stability. In [10], the issue of jointly optimal energy allocation
and admission control for communication satellites in Earth orbit (LEO, MEO
98
Mara Angeles V
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and GEO) is addressed. Using a dynamic programming approach, an optimal

policy is derived.
In general, information about channel conditions can be used to adapt
the coding or schedule transmission [11]-[13]. In [14], several levels of adap-
tation are proposed within each layer, fast and slow ones. The adaptation
also covers the hardware layer. In [15], the authors propose a cross-layer
design approach using perfect prediction-based wireless channel conditions to
improve the performance of a multicast packet scheduler over satellite network
environments in the downlink transmission. In [16], cross-layer methods are
used to improve the eciency of reliable multicast services supported by GEO
satellites. The reliability issue has to be carefully taken into account, since
satellite resources are expensive and link quality degrades signicantly during
adverse weather conditions. This paper proposes to remove at low layers,
most of packet discarding, but introduces an additional protection for protocol
headers. Moreover, at transport level erasure coding is used in combination
with a hybrid-ARQ protocol. Such approach allows that applications (like
massive le transfers) requiring full reliability are less demanding in terms of
network resources.
Joint PHY/MAC/APP optimization
A coordinated cross-layer adaptation can be considered to meet QoS demands

from the application layer. In [17], a mechanism is proposed to map QoS
levels of scalable video to the QoS levels of the transmission, both being time-
varying. Scheduling policies are derived allowing QoS mapping interaction
between the video coder and the transmission module. In [18], a cross-layer
framework for WLAN QoS support is proposed. The authors show that QoS at
MAC layer can be optimized by taking advantage from layers 4-7 information.
In [19], a joint cross-layer design for QoS content delivery is proposed. The
authors derive a QoS-aware scheduler and power adaptation scheme at both
uplink and downlink MAC layer to coordinate the behavior of the lower layers
for an ecient utilization of resources. They show that the cross-layer design
provides a good scheme for wireless QoS content delivery. In [20], power saving
is proposed by using feedback from the application about delay sensitivity.
Moreover, information about the type of coding used by a video-application
could be used by the frame scheduler at the network interface to save power
[21].
In a similar context, the problem of QoS mapping between adjacent layers
has been recently treated in [22],[23]. Rather than considering specically the
network and the MAC layers, the problem is posed in a more general setting,
as dened by the ETSI Broadband Satellite Multimedia (BSM) protocol
architecture [24],[25], at the Satellite Independent - Service Access Point
(SI-SAP). Specically, the interworking between the Satellite-Independent
(SI) and Satellite-Dependent (SD) architectural components is considered by
taking into account both the change in encapsulation format and the trac
aggregation (in the passage from SI to SD queues). In the presence of IP

DiServ queues at layer 3, the problem consists in dynamically assigning the
bandwidth (service rate) to each SD queue, so that the performance required
in the SI IP-based SLA is guaranteed. By considering a uid model and the loss
volume as the performance indicator of interest, the Innitesimal Perturbation
Analysis (IPA) technique of Cassandras et al. [26] is applied. Assuming that
the SI layer is properly congured, in order to satisfy the requirements (i.e.,
the IP buers do not constitute a bottleneck for QoS) the MAC resource
allocation is performed to maintain on-line the equalization between the loss
volumes at the network layer and at the MAC layer. In doing so, the allocation
is dynamically adapted, to follow both trac and fading variations. More
details on this scheme are provided in Section 8.4.
Joint optimization of layers involving transport layer
The transport layer is in charge of establishing end-to-end network connec-

tions. Transport protocols like TCP interpret large delays and packet losses,
typical of a wireless channel, as a congestion event, thus aecting the TCP
performance.
In [27], it is shown that increasing MAC level retransmissions, in order
to avoid TCP retransmissions, decreases the power consumption. In [28] and
[29], TCP windows are optimized according to the application priority. The
bandwidth assignment problem for long-lived TCP connections in a faded
satellite environment is addressed in [30], where cross-layer optimization
approaches between physical and transport layers are presented. Another
example of physical-transport cross-layer approach can be found in [31], where
the authors demonstrate that it is possible to obtain a better performance for
TCP connections by jointly choosing the bit error rate and the information
bit-rate of satellite links that maximize the goodput of a single TCP connec-
tion, without touching the TCP stack.
In [32], an innovative resource allocation algorithm, based on a cross-layer
interaction between TCP and MAC layers is proposed for a DVB-RCS
scenario. Such an algorithm aims to synchronize the requests of resources
with the TCP transmission window trend. The obtained results show that
the scheme permits to reduce the delay, to increase the utilization of air
interface resources, and to achieve a fair sharing of resources among competing
ows. This approach calls for a TCP-driven Dynamic Bandwidth and Resource
Allocation (DBRA) to be operated at layer 2 so as to reduce the queuing delay
(layer 2) and congestion phenomena (with timeout expirations) [33]. More
details on these techniques are shown in Section 9.4.
In split scenarios [34], the end-to-end TCP semantics is broken. The
satellite link is isolated by the terrestrial segment and interconnecting routers
(Performance Enhancing Proxies, PEPs) are used that close the TCP ow.
PEPs are typically implemented at transport or application layer. Examples
of transport layer PEPs are TCP spoong and TCP connection-split proxies.
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Mara Angeles V
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In both PEP types, the goal is to shield high-latency or lossy satellite network
segments from the rest of the network, in a transparent way to applications.
A critical issue in PEP is the design of buers and related management
rules and sizes. Interesting proposals envisage the adoption of Active Queue
Management (AQM) at the MAC layer for improving the TCP performance.
In AQM, when the router determines that the bandwidth is fully utilized,
packets are dropped even when the queue is not full in order to reduce the
data injection rate of the TCP sender [35].
In [36], experimental quantitative performance metrics can be found; they
are obtained by using H.264 and UDP-Lite for the next-generation transport
of IP multimedia. A cross-layer technique is proposed that features partial
checksum coverage for the packet header allowing the application to signal
implicitly the link CRC coverage. The sending end-host implicitly signals (i.e.,
without explicit control messages) by using a modied transport header, such
as UDP-Lite. This work discusses the architectural implications for enhancing
performance of a wireless and/or satellite environment.
Joint optimization of layers involving call admission control
Reference [37] presents an overview of high-speed mobile satellite commu-

nication systems, the technologies adopted or planned for deployments, and
the challenges. Various physical channel models for characterizing the mobile
satellite systems are presented. The most prominent technologies used in
the physical layer, such as coding and modulation schemes, multiple-access
techniques, diversity combining, etc., are discussed in the scenario of satellite
systems. What is interesting in our context is the overview of cross-layer
design methods employed in satellite systems, in particular those that involve
joint network and physical layer optimizations, or joint MAC and physical
layer optimizations. Specically in the GEO satellite environment, dierent
forms of parametric Call Admission Control (CAC) strategies have been
proposed, among others, in [38],[39], and [40], which are all based on a
cross-layer optimization. In [38], where the presence of both Variable Bit
Rate (VBR) MPEG connections and Available Bit Rate (ABR) data has been
considered, CAC is exerted with the goal of keeping the probability that the
bandwidth dedicated to VBR exceed a given value below a predetermined
threshold. A bandwidth expansion factor, whose value is adaptively adjusted
on the basis of measurements, is used to account for statistical multiplexing
eects in VBR trac. FEC and MPEG coding rate adjustments are other
corrective actions taken to cope with trac and channel variations. The
approach taken in [39] and [40] considers real-time Reserved Bandwidth (RB)
and Best Eort (BE) trac; however, no rate adjustment derived from
application-level coding is assumed to be available for RB ows. Adaptive
cross-layer bandwidth partitions are derived per station, based on stationary
performance indexes, such as the call blocking probability for RB connections
and the loss probability for data packets, which are recomputed at each
signicant change in fading or trac intensities. The control architecture has

a hierarchical structure, where CAC tasks are delegated to local controllers
at the stations, and uplink capacity partitions for the Earth stations are
adaptively determined by a Master Control Station (MCS). Owing to the
dynamic fade changes, the bandwidth assigned to an Earth station may be
temporarily insucient to carry on the currently ongoing number of RB
connections; since inelastic trac is considered, in such cases one or more
ongoing calls would be dropped. However, reallocations of the bandwidth
partitions upon detection of signicant changes in trac intensities and fading
classes do help in reducing the probability of this event. As regards the
MCS, the bandwidth allocation is formulated as an optimization problem in a
discrete setting (with the assignment granularity determined by the Minimum
Bandwidth Unit, MBU); if the performance index is a separable function of
the station parameters (e.g., a sum of terms, each depending only on the
bandwidth to be assigned to a station), the problem can be numerically solved
by applying dynamic programming over the stations [39],[40], possibly in a
form that may greatly reduce the search space, by exploiting the presence of
constraints.
It is worth noting that these model-based approaches can be by-passed
by using a uid approximation and by treating the bandwidth partitions
as continuous variables. A gradient descent technique can be adopted, in
conjunction with IPA for gradient estimation [27],[28]. The advantage of
these methodologies is that they are measurement-based and they require
neither the knowledge of any functional form of the performance index nor
any characterization of the trac sources.
A cross-layer radio resource management problem involving network and
MAC layers has been extensively considered in [29],[41], and [42]. In particular,
Dynamic Capacity Allocation (DCA) is applied, by computing bandwidth
requests for each Earth stations DiServ queue, which are passed to a
centralized scheduler, typically residing in an MCS. The latter assigns the
bandwidth proportionally to the requests received. The requests are computed
on the basis of queuing models, capturing both Short Range Dependent (SRD)
and Long Range Dependent (LRD) behaviors, and by using as QoS metric the
probability of the length of each service queue to exceed a given threshold,
depending on the service; this probability must be kept below a specied
value, beyond which the station is considered in outage. The scheduling of the
MAC queues must be such that this constraint is maintained for the IP-level
queues [i.e., those corresponding to Expedited Forwarding (EF), Assured
Forwarding (AF) and Best Eort (BE) services within a given Earth station].
The remaining capacity is assigned on a free basis, according to Combined
Free/Demand Assignment Multiple Access (CF/DAMA). Only trac is taken
into account (fading variations are not considered), but, as noted in [29], the
eect of fade countermeasures might be included as a reduction in the available
uplink bandwidth.
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Concluding comments
From this literature review, some general conclusions can be drawn as follows:
Little work has been published to date on cross-layer optimization in the

satellite context.
Most of the cross-layer optimizations proposed in the terrestrial wireless
realm involve physical layer and MAC layer. After these two layers, the
application layer is also widely considered. TCP is a particular case in
the sense that very dierent alternatives have been explored in order to
optimize the TCP protocol itself, especially over satellite channels.
Two main system performance parameters are optimized: QoS or service
dierentiation, in particular harmonization of QoS across layers, and
throughput. A special attention is also paid to energy saving, which may
not be directly applicable to a satellite scenario.
A wide variety of methodologies are presented and therefore no mature
general methodology seems to be available. Moreover, every published
work seems to follow an ad-hoc cross-layer methodology for the particular
case to be optimized.
4.3 The need of a cross-layer air interface design
The ISO/OSI reference model and the Internet protocol suite are based on a
layering paradigm. The target of the ISO/OSI reference model was to dene an
open system so that dierent network elements can interwork independently
of manufacturers. The OSI protocol stack entails 7 dierent abstraction levels,
addressing separately communication tasks. Each protocol solves a specic
problem by using the services provided by modules below it and giving a new
service to upper layers. The main interest here is on IP-based scenarios. The
Internet protocol stack is slightly modied with respect to the ISO/OSI one
and entails 4 layers, as depicted in Figure 4.1.
Fig. 4.1: Current view of the Internet protocol stack.

Standardization bodies dene the dierent protocols that a system can

use to exchange information. The implementation of interfaces is left free to
manufactures, provided that they support the primitives that determine the
service.
The disadvantages of the strict layered approach can be detailed as follows:
The needs of a service provided by the communication system to its users

are dened at the top-level. The hierarchy and the overall performance of
the system is however build upon the bottom-level.
The bottom level does not communicate directly, but through all higher
layers with the top-level. Information is lost during this layer-by-layer top-
down conversion.
Layers are independently optimized.
The challenging characteristics of satellite communications are:

Dynamically-varying channel characteristics; both slow and fast variability
are present in a satellite scenario depending on whether mobile or xed
users are considered;
Similar to terrestrial mobile channel, the satellite mobile channel lacks of
reliability (need of countermeasures: coding, retransmissions, modulation
techniques, diversity, etc.);
Strong inuence of intra-system interference levels;
Bandwidth shortage and need of supporting broadband applications;
necessity of managing the bandwidth in an ecient way;
QoS support for multimedia trac classes;
Interoperability among dierent wireless networks (2.5G, 3G, 4G, WiFi,
WiMAX, satellite, etc.).
A strict modularity and layer independence may lead to non-optimal
performance in IP-based next-generation satellite communication systems.
Furthermore, the growth of heterogeneous networks entails the need of adap-
tive actions. Finally, since both radio resources and power are strongly
constrained, a system optimization is needed. In this framework, a better
adaptation to system dynamics and trac demands can be attained by
employing a cross-layer approach with interactions even between non-adjacent
protocol layers.
Without a cross-layer design in the air interface we can expect a loss of
system eciency according to some typical problems outlined below.
IP packets lost due to errors induced by the wireless channel are interpreted
as signals of congestion at the TCP level, thus lowering the bit-rate
(congestion window). A long time is needed to recover (in terms of TCP
goodput) after a loss event especially when multiple losses occur that cause
a TCP timeout.
Radio resources can be also allocated to mobile users that have bad channel
conditions.
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Intra-system and inter-system hando procedures can take a too long time
that leads to connection interruption or higher layer protocol timeouts.
System eciency is an important task in satellite communications where

radio resources are costly and scarcely available. System eciency is needed
for allowing a mass-market diusion of satellite services. Whereas, QoS
support is the mandatory aspect requested by end-users who do not care
about resource utilization, but expect a good service. Resource utilization
and QoS support are typically conicting needs; for instance, the best QoS
condition for delay-intolerant trac is to have a high amount of available
resources, thus contrasting with system eciency. These conicting needs can
be solved by means of a suitable cross-layer system design and by exploiting
the multiplexing eect. In particular, the dierent layers of the OSI protocol
stack should be jointly optimized or dynamically jointly adapted to nd the
best trade-o between resource utilization eciency and QoS provision.
The idea behind cross-layer design is that we can obtain substantial gains
in performance and eciency by jointly optimizing the behavior of dierent
layers. For example, source compression at the application layer can improve
with knowledge of the transmission rate being used at the link layer. Moreover,
the network layer can gain by looking both up and down the stack in order
to obtain route diversity and multilink routing, where the routing algorithm
might add redundant links if link layer provides an unreliable channel or if
QoS constraints from the application layer are particularly tight. Satellite
communication systems optimization calls for a vertical design of the air
interface protocol stack.
The cross-layer approach requires new interfaces across the layers, which
exchange control information beyond the standard ISO/OSI structure to
improve the interactions among layers. Cross-layer interfaces can be within,
between or beyond adjacent abstraction layers. Although interfaces between
adjacent layers are in general preferable, there can be the need for ecient
and direct interaction between non-adjacent layers; in general, a layer should
be aware of the other layers of the protocol stack. Cross-layer information can
be exchanged from higher to lower layers (top-down approach) or from lower
to higher layers (bottom-up approach).
In the classical OSI stack, the exchange of information between adjacent
layers is performed through send and receive primitives. In a classical
layered approach, non-adjacent layers can communicate only involving in-
termediate layers. The novelty of the cross-layer approach is to allow the
exchange of control information (signaling) among non-adjacent layers [43].
For instance, a get function can be used by higher layer protocols to acquire
the internal state of lower layer protocols; moreover, a set function can
be adopted by higher layer protocols to change the state of lower layer
protocols. Dierent solutions have been proposed to support the cross-layer
exchange of signaling information; an interesting method has emerged from
the following papers [44]-[46] where a global coordinator of the dierent
layers is considered allowing to acquire the internal state information from

the dierent protocols to store it in a shared memory and to set the state
of the protocols to be adaptable to dierent events (see Figure 4.2a). The
global coordinator may reside in the MAC (i.e., MAC-centric approach), in
the application layer (i.e., application-centric approach) or being an external
entity. It should be noted that in a slowly-varying scenario, such as for example
the interactive broadband satellite channel with stationary users, the MAC
layer could control adaptability (coordinating cross-layer interactions) in an
optimal way [47]; this is the case of the MAC-centric approach presented in
Figure 4.2b.
Fig. 4.2: (a) Possible cross-layer air interface based on a global coordinator; (b)
Possible MAC-centric cross-layer air interface.
4.4 Cross-layer design: requirements depending on the

satellite scenario
4.4.1 Broadband satellite scenario requirements (DVB-S/S2)
Next-generation multimedia broadband satellite networks require the devel-

opment of key technologies to increase the capacity and eciency as well as
to decrease the total cost for the end-user. Such requirements call for very
high throughput, exibility, multi-beam processing and system adaptivity.
Role of Ka band: Current bent-pipe Ku band satellites create diculties
to develop protable multimedia satellite models. The current deployment
of Ku band spot-beams and frequency re-use will probably be eective
for a near-term business model. However, spot-beam coverage, in conjunc-
tion with Ka band frequency, can be extremely advantageous. Satellite
transponders operating at Ka band frequency permit to achieve a higher
G/T and, therefore, higher return channel burst rates. With lower power
levels, the price of the terminal signicantly decreases. The launch of
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additional Ka band capacity will greatly aect the multimedia satellite

market and will probably lead to more successful models and protability.
Role of DVB-S2: Typical Ku band broadcasting links are designed with
a clear-sky margin of 4 to 6 dB and a service availability target of about
99% of the worst month (or 99.6% of the average year). Since the rain
attenuation curves are very steep in the region 99% to 99.9% of the time,
many dBs of the transmitted satellite power are useful, in a given receiving
location, only for some ten minutes per year. Unfortunately, this waste of
satellite power/capacity cannot be easily avoided for broadcasting services,
where millions of users, spread over very large geographical areas, receive
the same contents at the same time. However, this design methodology
devised for broadcasting systems is not optimal for unicast networks. In
fact, the point-to-point nature of link connections allows exploiting space
and time variability of end-user channel conditions for increasing average
system throughput. This is achieved by Adaptive Coding and Modulation
(ACM) format to best match the user SNIR, thus making the received data
rate location- and time-dependent. The inclusion of advanced coding and
modulation schemes has been the rst objective of the DVB-S2 working
group. In particular, ACM has been considered as a powerful tool to
increase system capacity, allowing for better utilization of transponder
resources and hence providing additional gain with respect to current
DVB-S systems. Therefore, ACM is included as normative in DVB-S2
for the interactive application area and optional for Digital Satellite News
Gathering (DSNG) and professional services. The standardization of the
use of ACM by the DVB-S2 standard, introduces therefore an adaptive
physical layer, which calls for the development of optimum adaptive
resource management strategies to exploit fully ACM potentialities.
Applications requirements: The requirement of increasing bi-directional
data rates so that multimedia broadband satellite solutions can be closer to
the specications of terrestrial networks is undoubtedly a core need for any
DVB-based or DOCSIS-based network due to the rise in video and large le
transfers in enterprises. Future broadband satellite networks should aim to
create more symmetry between forward and return links due to a perceived
future demand for symmetric applications such as videoconferencing or
interactive e-learning. Moreover, satellite solutions must include features
and functionalities similar to a terrestrial solution in order to integrate into
and coexist with current enterprise infrastructures.
In order to meet application requirements especially of future satellites

that implement adaptive physical layer (DVB-S2), a logic reasoning is that
cross-layer design is essential to exploit fully new technologies potentialities
instead of loosing them by constraining the design to the conventional protocol
stack with independent layers.
In what follows, per-layer-based requirements for cross-layer design of
broadband satellite systems are presented from the layer 2 perspective.
Physical layer requirements: The DVB-S2 ACM modulator operates

at constant symbol rate, since the downlink carrier bandwidth is as-
sumed constant. A sequence of physical layer frames TDM multiplexed
is transmitted. Each frame transports a coded block and adopts a uniform
modulation format. However, when ACM is implemented, coding scheme
and modulation format may change frame-by-frame. Via a return channel,
individual Satellite Terminals (STs) provide to the Gateway (GW) infor-
mation on the channel status, by signaling the SNIR and the most ecient
modulation and coding scheme the ST can support. The ST indications are
taken into account by the GW in coding and modulating the data packets
addressed to each ST. It is then apparent that the resource management
functionalities shall be aware of the physical layer adaptation in order to
follow the time variability of capacity.
Network layer requirements: IP-layer QoS provision should be ade-
quately mapped to layer 2 radio resource management protocols. Adequate
attention should be also paid to both IntServ and DiServ approaches.
Dierent multimedia trac should be provided either with reserved ca-
pacity or capacity on demand and QoS guarantees. AF, EF and BE
trac ows of the DiServ scheme should have an adequate mapping
at layer 2. Suitable layer 2 intelligence should be able to perform this
important task. In case the broadband satellite sub-network is used as
a stand-alone end-to-end network, where the end-to-end QoS can be
controlled, a practical solution may be to apply guaranteed QoS to the
access network. The implementation of this hybrid solution still needs to
be investigated since it requires end-to-end network coordination.
Transport layer requirements: resource management schemes may
account for the specic transport layer trac characteristics, such as TCP,
UDP and multicast/broadcast. Note that in this scenario (i.e., broadband
satellite communications for xed users) a memoryless channel has to be
considered that causes random packet losses, impacting the performance
of the transport layer. Few examples are provided below.
The ECN (Explicit Congestion Notication) signaling for TCP trac
could be exploited at layer 2 to modify some trac shaping functions
or policing schemes.
The TCP congestion window (estimating the network congestion level)
could be used at layer 2 to adaptively reserve capacity for TCP-based
trac; such approach could improve the QoS experienced for TCP-
based applications and could also improve the multiplexing eciency
of such trac ows (throughput). Note that the congestion window be-
havior plays a fundamental role in TCP-based satellite communications
due to the very high round-trip propagation delays.
Application layer requirements: dierent trac types (e.g., real-time
trac and non-real-time trac) should have specic SLAs and a moni-
toring action should be jointly performed with layer 2 in order to modify
adaptively the service priority.
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4.4.2 Mobile satellite scenario requirements (S-UMTS)
The mobile user scenario adds specic criticalities in the management of

resources due to the dynamically changing propagation conditions. Such
circumstances made even more crucial the need of cross-layer protocol design.
The management of air interface resources (layer 2) must be improved to
exploit dynamically updated information exchanged with all the other layers
and, in particular, OSI layers 1, 3, 4 and 7. In fact, the congestion of the
scarcely available satellite air interface resources as well as the congestion of
the related xed network are too critical aspects that must be taken into due
account when designing the air interface protocol stack and, in particular,
layer 2 resource management protocols.
Focusing on cross-layer information available at layer 2, we can consider
the following contributions coming from other (even non-adjacent) layers:
Physical layer requirements: radio channel conditions should be con-
tinuously estimated. In particular, signal strength, BER or PER estima-
tions should be made available to implement multi-mode (i.e., modulation
and coding) adaptivity and the selection of appropriate formats and
priority levels at layer 2. These capabilities are supported by a possible
satellite extension of the High Speed Downlink Packet Access (HSDPA)
standard, as discussed in Chapter 5.
Network layer requirements: in the IP trac management, user
mobility should be adequately taken into account. Hence, layer 2 protocol
should provide a prioritized management for trac coming from uses that
incur in handover phases (this may be very important and time-critical
in the presence of non-GEO satellites). In addition to this, mechanisms
for IP-layer QoS provision should be adequately mapped to layer 2 radio
resource management protocols, as already described in the previous
sub-Section (see requirements for network layer in sub-Section 4.4.1).
Transport layer requirements: resource management schemes should
be improved to account for the suitable rules for specic transport layer
trac, such as TCP, UDP and multicast/broadcast. Note that in this
scenario correlated packet losses are experienced that may aect the
transport layer behavior; typically, a multi-state channel model (e.g.,
good/bad model) should be considered. For details on requirements, please
refer also to the related part in sub-Section 4.4.1.
Application layer requirements: dierent trac types (e.g., real-time
trac and non-real-time trac) should have specic SLAs and a moni-
toring action should be jointly performed with layer 2 in order to modify
adaptively the service priority.
4.4.3 LEO satellite scenario requirements
LEO satellite networks are deployed as an enhancement to terrestrial wireless

networks in order to provide broadband services to users regardless of their
location. They provide signicant benets including wide area coverage,

unique broadcast capability, ability to meet dierent QoS requirements, the
possibility to communicate with hand-held devices and low access cost. At
the same time, these networks present protocol designers with an array of
important challenges, including handover procedures, mobility and location
management.
Two broadband transport technologies, ATM (Asynchronous Transfer
Mode) and IP, are proposed for future broadband LEO satellite networks. In
the recent literature most publications are oriented towards the ATM-based
LEO satellite scenario. For these reasons, such scenario is described in details
later on.
In case of IP-based LEO satellite networks, with IP-routing implemented
on board, the satellite network can seamlessly integrate with the terrestrial
Internet. Another advantage is the IP QoS support without any required
interworking with terrestrial IP QoS mechanisms. Multicast application pro-
vision is also well supported by using on-board router. However, routing in
mobile satellite IP networks is considered a complex issue, because, one cannot
simply use terrestrial Internet routing for on-board routing. The mobile IPv6
protocol, enhanced to support paging and handover, has to be implemented
on-board.
ATM is a basic transport mechanism for Broadband Integrated Services
Digital Network (B-ISDN), broadband Internet access and other technologies.
ATM provides high transmission rates, bandwidth-on-demand, compatibility
with previous existing protocols and guaranteed QoS. ATM-based LEO satel-
lite networks are expected to support a wide range of multimedia services and
applications and to provide their users with appropriate QoS based on the
strong end-to-end QoS mechanisms oered by the ATM technology. However,
the limited bandwidth of the satellite channel, satellite rotation around the
Earth and the mobility of end-users make QoS provisioning and mobility
management a challenging task. The following list provides a description of
the requirements to support QoS in ATM-based LEO satellite systems.
Common LEO system requirements: The main resources in LEO

networks are the satellite radio bandwidth and the buer capacity of the
on-board ATM switches. Because the total link capacity has to be divided
among several carriers, and given the limited buer capacity of the ATM
switch, advanced resource reservation cross-layer mechanisms have to be
developed. They have to ensure fair bandwidth sharing and provide users
with the negotiated QoS guarantees as end-users roam in the system. At
the same time, the network and the end-systems have to be protected
from congestion. One of the most important QoS parameters for LEO
satellite networks is the Call Dropping Probability (CDP), quantifying the
likelihood that an on-going connection will be forcedly terminated due
to an unsuccessful handover attempt. Moreover, Call Blocking Probability
(CBP) quanties the chance that a new call request is denied entry into
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the system for lack of available resources [48]. Cross-layering is aimed to

optimize bandwidth allocation, and to provide for low CDP for reliable
handovers and acceptable CBP for new calls, while maintaining high
resource utilization.
ATM layer requirements: ATM-based LEO satellite networks should
be able to meet dierent QoS requirements at the ATM layer. These
requirements are stated in terms of the objective values of the network
performance parameters, as specied in ITU-R Recommendation S.1420
[49]. Some of the QoS parameters may be oered on a per-connection basis
and are negotiated between the end-system and the network. Other QoS
parameters cannot be negotiated.
MAC layer requirements: The most important resource management
function is bandwidth allocation. The main constraint is the bandwidth
available to all users on the satellite uplink. Unlike a xed ATM network,
the satellite can only control the bandwidth in the downlink towards the
end-system. Thus, dynamic bandwidth allocation should be developed in
order to meet QoS guarantees for the various Virtual Channels (VCs),
as dened in the trac contracts. Moreover, it is necessary to ensure
the utilization of the unused bandwidth by connections with no explicit
guarantees, as a BE service. Additionally, the MAC protocol should
provide support for the ATM service categories. Only a QoS-aware MAC
protocol is able to comply with the QoS requirements of dierent ATM
service categories and the ATM signaling. MAC for ATM via satellite
is also faced with the fact that an ATM cell does not have a dedicated
eld for the service parameters. In ATM, the service parameters of a
connection are announced to the ATM switches along with the VPI/VCI
value during the connection setup. Thus, the service parameters of the
ATM cells belonging to a certain connection can be identied only through
its VPI/VCI value. Consequently, the MAC layer needs some kind of
lookup table with the service parameters of the ATM connections and the
corresponding VPI/VCI values, if QoS of dierent ATM service categories
has to be supported. This determines a special design of the protocol stack
[50].
Network layer requirements: The most important resource manage-
ment function is CAC. The CAC algorithm operates at the call level in
the network. It denes the procedure performed by the network during the
call set-up phase to determine if the connection request can be accepted
without infringing on existing commitments. If the request exceeds the
available bandwidth, the role of the CAC is to deny the connection. In
this case, we say that the connection is blocked. CAC schemes should be
improved and mapped to layer 2 radio resource management protocols. A
detailed analysis of CAC schemes is provided in Chapter 6.
4.5 Conclusions
In this Chapter we have provided a comprehensive literature review of existing

cross-layer design approaches. From the literature review and taking into
consideration the particular characteristics of the satellite scenario, a set of
requirements has been identied for resource management with cross-layer
design. These requirements have been shown to be dierent for the dierent
scenarios from broadband to mobile and from GEO-based to LEO-based
systems. The need of a cross-layer air interface design has been discussed
and a couple of possible cross-layer architectures presented.
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Part II
Cross-Layer Techniques for

Satellite-Dependent Layers
5
ACCESS SCHEMES AND PACKET
SCHEDULING TECHNIQUES
Editors: Giovanni Giambene1 , Cristina P

arraga Niebla2 , Victor Y. H. Kueh3
Contributors: Kostantinos Avgeropoulos4 , Wei Koong Chai3 , Giovanni

Giambene1 , Samuele Giannetti1 , Du Hongfei3 , Victor Y. H. Kueh3 , Cristina
arraga Niebla2 , Veronica Pasqualetti1 , Aduwati Sali3 , Orestis Tsigkas4
P
1
2
DLR - German Aerospace Center, Institute of Communications and Naviga-
tion, Wessling, Germany
3
UniS - Centre for Communication Systems Research, University of Surrey,
UK
4
5.1 Introduction
The dual objectives of achieving ecient satellite resource utilization and

acceptable user QoS levels require a consistent, controllable and exible Radio
Resource Management (RRM) scheme. The interest is here in managing
packet data trac of multimedia nature in mobile satellite systems. Complex-
ity is added by the presence of multimedia trac classes with dierentiated
QoS requirements and for the dynamically-varying channel conditions with
(possible) consequent adaptations at the physical layer.
The MAC layer is the place in the protocol stack where RRM techniques
operate. In fact, the achievable resource utilization eciency and the resulting
120 Giovanni Giambene, Cristina P
arraga Niebla, Victor Y. H. Kueh
QoS are governed by MAC protocols that are used in the uplink case to
manage the transmissions of dispersed terminals to an Earth station through
the satellite and that are also employed in downlink to schedule the dierent
transmissions from the Earth station to the terminals. Hence, the two essential
components of the MAC layer are: access protocols and scheduling techniques.
These are also the main targets of this Chapter.
The studies carried out in this Chapter are related to Scenario 1 for what
concerns S-UMTS (see Chapter 1, Section 1.4); however, the last part of this
Chapter refers to a TDMA-like air interface.
5.2 Uplink: access schemes
Since early 1960s, satellite access protocols have attracted the attention
of various researchers. These protocols control the access of a station to
the transmission medium. For terrestrial networks, where the transmission
medium could be a coaxial cable or a twisted pair, several MAC protocols
such as Ethernet, Token Rings and Token Buses have matured. However, these
protocols are not suitable for satellite networks. Although the functionalities
required and the users QoS requirements are similar, the design of a satellite
access protocol is more complicated and restrictive due to its operating
environment. In brief, there are ve reasons why many access protocols
designed for terrestrial networks are not suitable for satellite ones [1]:
The long propagation delay constrains the performance of access protocols.

Satellite and terrestrial links have very dierent characteristics.
Hardware modications to controllers used in space are almost impossible
and hence, satellite access protocols need a simple control mechanism.
In contrast with terrestrial networks where topological changes are slow,
satellite networks are characterized by topological changes and network
recongurability in case of failures is mandatory.
Power limitation in satellite networks is much stringent and therefore, the
use of buer memory, transponder capacity and processing power are more
restrictive.
In the access protocol design phase, there are several factors to be taken
into account. One of them is the type of applications that would traverse
the satellite network. The trac pattern the satellite network is expected
to support is also a main input to the design process. As new network
technologies and applications emerge, access protocols also evolve accordingly.
Generically, there are ve main access protocol categories:
Fixed Assignment (FA),

Random Access (RA),
Fixed rate demand-assignment,
Variable rate demand-assignment and
Chapter 5: ACCESS SCHEMES AND PACKET SCHEDULING TECH. 121
Free assignment.
Fixed assignment protocols were the initial access protocols being used in
commercial systems. However, because they were inecient, newer proposals
were demand-assignment protocols. The main application at that period was
telephony and thus, xed demand-assignment was proposed. Later, the need
to support packet-switched data network has led to the introduction of random
access protocols to satellite networks in early 1970s. Although improvements
for the protocols in this class have been proposed for satellite, their low
upper bound utilization has encouraged researchers to seek for alternatives.
The result is the use of variable demand-assignment protocols. Based on
the buer state, users compute and send resource requests. The requested
resource will be allocated for a nite period, usually in terms of a number
of frames. With the increasing need to support multimedia trac, the access
protocol has to be able to manage trac ows (i.e., trac classes) with distinct
QoS requirements. As a response, hybrid protocols have been proposed,
combining diverse resource allocation mechanisms for dierent trac types.
For instance, to support real-time inelastic trac, xed demand assignment
coupled with additional admission control could be used while for elastic data,
a combination of variable demand-assignment and free assignment (e.g., a sort
of round-robin allocation) could be the right choice.
In the following sub-Sections we examine random access protocols for
S-UMTS. We begin by describing the current proposals for random access
in S-UMTS and continue with an overview of PRMA-like schemes. Finally,
we examine how PRMA can be adopted by S-UMTS and which cross-layer
approach can be adopted to optimize the access protocol performance.
5.2.1 Random access in UMTS and application to S-UMTS
The S-UMTS air interface is currently dened by ETSI in technical spec-

ications 101.851-1 to 101.851-4 [2]-[5]. These specications do not dene
the type of satellite system (GEO or non-GEO) to be used, although the
focus is towards GEO systems. Attention is given however to the consistency
between the terrestrial and the satellite part of the system in terms of air
interface design. For this reason, the general design and channel structure of
the satellite air interface follows that of T-UMTS, modied appropriately in
order to accommodate the special characteristics of satellite communications
(long delay, Doppler eect, propagation loss, etc.). Table 5.1 below presents
the physical channels used in S-UMTS and describes how these are mapped
to transport channels, which in turn provide services to the higher layers.
The only common uplink physical channel available in S-UMTS is the
Physical Random Access Channel (PRACH), which is mapped one-to-one to
the Random Access Channel (RACH) at the transport level. In one cell, several
RACHs/PRACHs can be congured. The Physical Common Packet Channel
(PCPCH), the other common uplink channel available in T-UMTS, is not
Physical Channel Direction Description

DPDCH Both Carries the DCH transport channel
DPCCH Both Layer 1 control information for DPDCH
PRACH Uplink Carries the RACH transport channel
P-CPICH Downlink Phase reference for downlink channels
S-CPICH Downlink Phase reference for dedicated downlink channels
P-CCPCH Downlink Carries the BCH transport channel
S-CCPCH Downlink Carries the FACH and PCH transport channels
SCH Downlink Synchronization (spot search)
AICH Downlink Acquisition indicators (random access results)
PICH Downlink Paging indicators
MICH Downlink MBMS indicators
Physical Channels
DPDCH Dedicated Physical Data Channel
DPCCH Dedicated Physical Control Channel
PRACH Physical Random Access Channel
P-CPICH Primary Common Pilot Channel
S-CPICH Secondary Common Pilot Channel
P-CCPCH Primary Common Control Physical Channel
S-CCPCH Secondary Common Control Physical Channel
SCH Synchronization Channel
AICH Acquisition Indicator Channel
PICH Paging Indicator Channel
MICH MBMS Indicator Channel
Transport Channels
DCH Dedicated Channel
RACH Random Access Channel
BCH Broadcast Channel
FACH Forward Access Channel
PCH Paging Channel
Table 5.1: Transport and physical channels.
supported in the satellite air interface. RACH is characterized by open-loop

power control and a collision risk in every transmission. RACH is crucial
for the operation of the UMTS air interface, since it is used not only for
initial channel access to the network (e.g., call origination, paging response
and registration messages), but also for sending short data bursts (e.g., Short
Message Service, SMS), as investigated in the following simulative study.
In 3GPP specications [6], the PRACH transmission is based on a Slotted-
ALOHA (S-ALOHA) approach with fast acquisition indication. The User
Equipment (UE) can start the random access procedure at the beginning
of a number of a well-dened time intervals, called access slots, by sending
a preamble burst, as detailed below. There are 15 access slots on a 2-frame

structure (totally, 20 ms duration) and they are interspaced by 5120 chips.
The PRACH transmission consists of two parts: preamble and data mes-
sage. Since, the preamble can be transmitted one or several times (due to
possible collisions), we can arm that the structure of the random access burst
is composed by one or several preambles and one message (see Figure 5.1). The
random access procedure to transmit the preamble is dened in [2]-[5],[7]. The
preamble part has a length of 4096 chips and the message part has a length
of 10 or 20 ms. Only when the preamble is successfully detected the UE can
transmit the message part with a power related to the detected preamble and
with a channelization code corresponding to the signature selected to transmit
the preamble.
Fig. 5.1: Structure of the message transmission on RACH.
To construct the preamble, the UE uses two components: the preamble

scrambling code (there are 8192 such codes available) and the preamble
signature code (16 signatures to choose from, obtained as a repetition of a
Hadamard codeword). These codes, sequences of chips with values +1 or 1,
are combined to determine the complex preamble transmission code. More
details can be found in [4].
The 10 ms message is split into 15 slots, each of 2560 chips (each slot of
these has half duration with respect to access slots). The message consists
of two parts: the data part and the control part, which are transmitted
simultaneously (see Figure 5.2) using dierent channelization (spreading)
codes that both depend on the signature used to construct the preamble part.
The control part has a Spreading Factor (SF) of 256 and the data part can
have dierent spreading factors in the set {32, 64, 128, 256}. The content of
the data bits depends on the higher layers. The 8 pilot bits of the control
part are used to support channel estimation for coherent detection and the
Transport Format Combination Indication (TFCI) bits are used to indicate
the spreading factor and the format of the data part.
Access Service Class (ASC) represents a certain PRACH partition (i.e.,
sub-channels and signature codes, as explained below) and an associated
access persistency value (i.e., a probabilistic check to determine whether a
preamble transmission can be attempted in the current access frame). There
Fig. 5.2: Structure of the RACH message part (slots are here shorter than the
access slots; in this case, a slot contains 2560 chips so that 15 slots correspond to 10
ms).
are 8 ASCs, numbered from 0 (highest access priority) to 7 (the lowest access
priority) [8]. ASC 0 shall be used for emergency calls. A PRACH sub-channel
denes a sub-set of the access slots. There are a total of 12 sub-channels.
Typically, every 8 frames the allocation pattern of the dierent access slots
to the dierent sub-channels repeats. The higher layers communicate to the
physical layer the available signatures and sub-channel groups for each ASC.
There are at most 16 PRACH channels per cell; each of them corresponds
to a dierent preamble scrambling code. On a given access slot of a PRACH,
up to 16 simultaneous transmissions are possible by using distinct (orthogonal)
signatures codes. A PRACH channel is dened by the following parameters:
preamble scrambling code, spreading factor for data part, available signatures
for each ASC, available sub-channels (i.e., slots) for each ASC and power
control information. Available sub-channels and signature codes are broadcast
through the BCH channel. When there is data to be transmitted, the UE
performs PRACH selection randomly. Then, MAC selects the appropriate
ASC for the trac type to be managed. Consequently, an access slot and a
signature are randomly selected among those available for the selected ASC.
In the PRACH access mechanism, the main dierence with respect to the
classical S-ALOHA system is that, besides the time of the transmission, the
UE also randomly chooses the signature and the scrambling code that will be
used to transmit the preamble.
Once the preamble is sent, the UE waits for an acquisition indication
(a sort of acknowledgment message) sent by the Node-B on the Acquisition
Indicator Channel (AICH), a downlink physical channel that is received in
the entire cell or part of the cell in case of sectorization. This transmission
may fail for various reasons (interference from other terminals, fading, etc.).
If an acquisition indication is not received by the time the UE response
timer expires (pa ), the UE schedules a new transmission attempt on the
ASC resources. Note that this timer must be set to a value greater than
the estimated round trip delay. In the GEO satellite scenario, this timer can
be set to either 280 or 560 ms (the actual selection is made by upper layer
procedures) depending on the fact that the satellite is regenerating or not [2].
The system can provide dynamic persistency by publishing a dynamic
persistency value through the Broadcast Channel (BCH). This value should
be determined on the basis of an estimate of the number of contending UEs.
The ow chart in Figure 5.3 describes the random access protocol on the
RACH channel. For further details the interested reader should refer to 3GPP
specications [5].
The message transmission is performed with a scrambling code that is
one-to-one mapped to the scrambling code used for the preamble.
The remainder of this sub-Section is devoted to the performance evalu-
ation of RACH in a GEO bent-pipe scenario. A C++ simulator has been
implemented with a slightly simplied access procedure with respect to that
in Figure 5.3 (i.e., no power ramping has been considered; only one PRACH
has been simulated). We refer to a GEO bent-pipe satellite scenario, where the
Node-B that manages the RACH protocol is on the Earth: the UE exchanges
messages with the Node-B via the GEO satellite. In this study the Earth
station provides a feedback to the UE about its transmission attempts. Hence,
there is a round-trip propagation delay of about 560 ms to know the outcome
of this transmission (pa timer has been set accounting for such propagation
delay).
In order to evaluate whether the access attempt has been successful or not,
we have to consider collision events and the uplink interference conditions
typical of CDMA transmissions. An access (i.e., preamble transmission) is
considered successful if the following conditions are fullled [9]:
1. No other UE selects the same access slot and the same signature code on
the same PRACH (otherwise there is a collision event; the capture eect
is not considered in this case).
2. The received Signal-to-Interference Ratio (SIR) at the satellite exceeds a
given threshold, SIRt .
The above SIR issues (point 2) can be taken into account in the access
phase by assuming a maximum number of transmissions (MaxUE) that can
be tolerated in the same access slot for interference reasons. Hence, when
there are n concurrent access attempts with n > MaxUE, there is a too high
interference level (i.e., SIR < SIRt ) so that all n transmission attempts (using
dierent signature codes) are unsuccessful. We can consider that MaxUE is
1
proportional to SIR t
.
The simulator numerical settings are detailed below:
Fig. 5.3: Random access process on the PRACH channel (PRC, Power Ramping
Control, denotes a mechanism to increase the transmission power of the access burst
in subsequent attempts).
We have considered a GEO bent-pipe satellite scenario with round-trip

propagation delay of 560 ms.
Only one PRACH has been simulated (i.e., one scrambling code is used).
We consider two dierent cases for the interference conditions concerning
the preamble transmission: MaxUE = 6 (mild interference conditions) and
MaxUE = 3 (severe interference conditions). More appropriate MaxUE
values could be determined with a complex analysis of the interference con-
ditions deriving from the simultaneous transmissions of dierent preambles
on the same access slot with dierent signature codes and the same
scrambling code. Such a study is beyond the scope of this work.
All the signature codes can be used by an ASC. While, dierent ASCs
are distinguished by a dierent set of used sub-channels. In particular, the
numbers of sub-channels distributed among the ASCs are as follows: ASC0
= 8, ASC1 = 3, and ASC2 = 1. Hence, the highest-priority ASC0 has a
greater number of resources (i.e., sub-channels), thus guaranteeing lower
collision and interference probabilities. Note that in this study, all the 12
sub-channels are used.
We refer here to a case with persistency probability equal to one: the trans-
mission of the preamble is soon attempted or reattempted by randomly
select the resources.
A source (i.e., UE) generating a message does not generate another
message until the previous one has been transmitted. Hence, a source can
be in the OFF state (waiting for the generation of a new message) or in
the ON state (waiting for message transmission).
There are 10 sources per ASC. The OFF state sojourn time is exponentially
distributed with mean message arrival rate denoted with . As soon as
the source leaves the OFF state, a procedure is started to transmit a 10
ms message.
After the successful transmission of the preamble, message transmission
requests are served according to the priority order of the related ASC. A
virtual message transmission queue corresponds to a PRACH (messages
from ASC0 are prioritized with respect to ASC1, etc.). These message
transmissions use a suitably shifted scrambling code with respect to the
scrambling code of the preamble transmission that also combines this
code with a signature code. We neglect interference between simultaneous
message and preamble transmissions related to the same PRACH. Hence,
preamble transmissions and message transmissions use separated resource
spaces. Of course the message part can be received at the Node-B with
errors according to a certain Frame Error Rate (FER) value.
Simulation runs have a duration of 500 s.
We evaluate through simulations both the mean preamble delay (from

the arrival of the message for the S-RACH transmission to the instant when
the terminal receives the acknowledgment -AICH message- that the random
access is successful) and the mean message delay (from the instant when
the AICH message is received to the instant when the message transmission
completes). The total mean message delay (from message arrival to message
transmission) is the sum of the two above mean delay components. Results are
shown in Figure 5.4 considering both the cases MaxUE = 6 and MaxUE =
3. The ideal preamble delay (lower bound) only contains a frame duration
and a round trip delay. As expected, the mean preamble delay increases
with the mean arrival rate and reduces with the MaxUE value. Moreover,
the mean preamble delay increases from ASC0 to ASC1 and to ASC2 (i.e.,
the higher priority ASC0 permits to achieve lower mean preamble delay
values). As expected, the mean message delay increases with the mean arrival
rate and is practically insensitive to the MaxUE value variation (the

message transmission on PRACH can be described as a simple queuing system
-M/D/1-like queue with state-dependent arrival rate- with no interference
with preamble transmissions, as previously assumed). Moreover, the mean
message delay for ASC0 is lower than that for ASC1 that, in turn, is lower
than that for ASC2.
Note that for all the ASCs, the mean preamble values are not so dierent,
thus proving the robustness of the preamble access protocol: the time-code
space is a suciently wide resource space also for the ASCs with lower number
of assigned sub-channels. The random access scheme for preamble, based on
dierent sub-channels and signature codes, has an intrinsic stability since
it uses a form of special capture eect due to the codes. In addition to
this, the mechanism that a source in the ON state cannot generate a new
message, allows reducing the load of random preamble attempts and the load
of messages to be transmitted on the PRACH virtual queue. This mechanism
further provides stability to both the random access phase and the subsequent
message transmission queue.
As a nal consideration, we may note that these results prove that the total
message delay is high in a GEO bent-pipe scenario. A possible improvement
has been proposed for the GEO satellite case in [9] where the message
transmission immediately follows the preamble transmission.
Fig. 5.4: PRACH performance in the presence of trac on three ASCs with
dierently allocated resources and two cases for MaxUE values.
5.2.2 The Packet Reservation Multiple Access (PRMA) protocol
PRMA is a random access mechanism based on TDMA and S-ALOHA. Since

its initial proposal in [10] it has attracted the attention of both research
community and industry, especially because of its eciency when handling
real-time trac. PRMA can be viewed as a Dynamic TDMA (D-TDMA)
protocol where time slots are allocated to the users on demand. It is targeted
mostly for voice and data trac [11]-[14].
PRMA voice User Terminals (UTs) use speech activity detectors so that
the channel is accessed only when there is voice trac to be sent. This is
important, since a voice channel is active for less than 50% of the time in a
telephony dialog, which means that allocating slots statically for the entire
call would waste resources.
As in all time division mechanisms, a PRMA carrier is divided into time
slots of duration Ts that are grouped into frames of duration Tf = Ts N. Each
slot has two states: available and reserved.
The gure below shows the state diagram for a UT in the simplest case
where a UT is allowed to reserve only one slot at a time [more complex state
diagrams result when more than one reserved slot per UT is allowed in a
frame: N 1 or 2(N 1) states are added, depending on the mechanism used
to reserve additional slots]. The UT starts in the silent state. When a talkspurt
begins, the UT moves to the contending state where it attempts to reserve
one slot in order to transmit the voice data. Random access transmissions are
only allowed in available slots and occur according to a permission probability
scheme. We assume that a UT monitors the state of the slots (using a
downlink control channel) and therefore knows which of them are available.
If a transmission is correctly received by the base station (no collisions), the
transmitting UT is notied via a downlink control channel (this channel is
often broadcast and can be used by UTs for slot state monitoring). In this case,
the UT moves to the active state and the slot becomes reserved. This means
that only the reserving UT is allowed to transmit in that slot in subsequent
frames. When the talkspurt ends, the UT releases the slot by sending a special
signal and moves again to the silent state while the slot becomes available. If
the random access burst is not correctly received, usually due to a collision
with other UTs that transmit their random access burst in the same slot,
the base station informs the UTs that a collision has occurred and the UT
remains in the contending state and schedules a retransmission attempt. The
UT behavior in the access phase is depicted in the diagram in Figure 5.5.
During the access phase, a packet can be dropped (front-end clipping
phenomenon): if the voice packet transmission delay D (i.e., the time between
the packet generation and the packet successful transmission) exceeds a
certain limit (30-40 ms), Dmax , the packet is dropped and the UT will
attempt to transmit the next one following the same procedure. Of course, the
probability that a packet gets dropped is an important performance parameter
and must be kept very low (lower than 1%) for guaranteeing a good voice
Fig. 5.5: State diagram of the PRMA protocol.
quality.
As shown in [10], PRMA outperforms the classical S-ALOHA protocol in
terms of packet dropping probability and is therefore more preferable. It is
also exible enough to accommodate data and voice trac. Moreover, there
have been proposals where a UT can reserve more than one slot per frame
to accommodate more demanding real-time trac. There are certain issues
however that are critical for PRMA performance, some of them are even more
important in the case where it is used for satellite systems. These issues are:
Frame and slot duration, channel bandwidth and voice codecs. In our
discussion above, we mentioned that in order to transmit a talkspurt the
UT reserves one slot per frame. This assumes that the channel bandwidth,
the slot and frame durations and the codec used must be coordinated in
order to receive the required voice quality at the receiver. This means that
if the channel bit-rate is Rc and the codec voice bit-rate is Rs , then the
maximum number of slots per frame is
Tf R c
Nmax = (5.1)
Rs Tf + Lh
where Lh is the length of each packet header.
Scheduling retransmissions and resolving collisions. We can assume that
as soon as a talkspurt begins, the UT selects the next available slot to
transmit the random access burst in order to make a reservation as soon as
possible. If there is a collision and all UTs that participated in the collision
select another available slot in deterministic manner (e.g., they all select
the next available slot), then they will enter a collision deadlock since all of
them will select exactly the same slot to transmit. To avoid such deadlocks,
a probabilistic collision resolution mechanism must be employed. In the
simplest case, each UT may decide to transmit with a probability p, known
as permission probability or persistence. Selecting an appropriate value

for this probability is vital in order to achieve a fast collision resolution
mechanism and to guarantee a stable protocol behavior.
Contending-to-active state transition time. Obviously, there is a strict
requirement for the time it takes for a UT to move from the contending
to the active state. If this time exceeds the packet deadline, then the rst
packet of the talkspurt must be dropped reducing the voice quality at the
receiver (front-end clipping phenomenon). The exact limit depends on the
codec used, but it usually amounts to a few tens of milliseconds.
Round Trip propagation Delay (RTD). As already discussed, the base
station is responsible for transmitting the results of a random access
attempt to UTs. This means that after a UT transmits its random
access burst, it must wait and listen for the result of its transmission
(success or failure) on the downlink channel for a duration at least equal
to RTD. In other implementations, the base station does not reply after
a failed transmission and the UTs assume that they failed after not
receiving a response within a given timeout. This means that in every
transmission (or retransmission) the round trip delay is directly added
to the contention phase. While this is not an issue in terrestrial systems
with very small RTD values, it is quite critical for satellite systems. To
cope with this problem, a modied PRMA protocol, called PRMA with
Hindering States (PRMA-HS) has been proposed in [11]. In this PRMA
version, the UT employs a more aggressive behavior in the contending state
by continuously reattempting random access transmissions during RTD,
without stopping for waiting the base station reply. It has been proved
that while this approach increases the contention load with possibly useless
re-transmissions, it still outperforms the classical PRMA scheme in mobile
satellite systems.
Available slots versus collision probability. In the classical PRMA protocol,
we described above, the number of available slots (i.e., the number of
unreserved slots that are available for contention) is variable. This means
that as more slots become reserved the probability that two or more UTs
transmit their random access bursts in the same available slot (collision
probability) increases. There are cases where this phenomenon is not
desirable. Therefore, there have been proposals in which a separate channel
is used for contention (for example, this channel may simply consist of
a certain amount of minislots in a reserved portion of the frame, thus
signicantly reducing the variations on the collision probability. There is
obviously a trade-o here, as these contention-dedicated resources may
cause a waste of bandwidth.
5.2.3 Adopting PRMA-like schemes in S-UMTS
GEO systems cannot adopt PRMA since their long RTD (max 280 ms in
the case of a regenerating satellite; max 560 ms for a bent-pipe satellite)
makes the state transition time from the contending to the active state to
exceed the limit posed by the codec (i.e., the voice packet deadline, typically
of few tens of ms). For these systems a simple reservation scheme may be
used where a reservation per call is made. In LEO systems, however, RTD is
much smaller (between 5 and 30 ms) and PRMA techniques are applicable.
A feasibility study for the adoption of PRMA in the LEO case is made in
[11]-[14], including the selection of the permission probability p and the frame
duration Tf .
It should be noted that there is a substantial dierence between the
S-UMTS air interface and the air interface assumed by classical PRMA.
PRMA relies solely on time division, whereas S-UMTS can be characterized
as a hybrid CDMA/TDMA system. Therefore, CDMA/TDMA variations of
PRMA must be considered such as the one proposed in [15], where UTs select
a code in addition to a time slot in order to transmit their access bursts in a
very similar fashion as we previously described.
The ow chart shown below in Figure 5.6 is an example of how S-UMTS
channels can be used in order to adopt a PRMA-based scheme. We assume
that a UT will require to use a Dedicated Channel (DCH) consisting of
one Dedicated Control Channel (DCCH) and one or more Dedicated Trac
Channels (DTCHs) (DCCH and DTCH are logical channels) depending on the
upper layer requirements. These requirements can be stated in the message
part of the RACH burst. If the burst is not received properly, the UT schedules
a retransmission using the permission probability p, which is announced by
the system using the BCH channel, as specied in [5]. Note also that by using
dierent channels for contention (RACH) and data transmission (DCH) we
keep the collision probability constant and independent of the already assigned
DCH channels. This separation of contention and data channels constitutes a
substantial dierence from classical PRMA schemes.
Note that in the presence of dierent trac classes sharing the same RACH
access channel, dierent permission probability values should be used to take
into account the trac urgency and other priority requirements.
As a conclusion, we may observe that S-UMTS as well as T-UMTS can
adopt PRMA-like schemes without dramatic alterations to the air interface,
since the already-available transport channels can be utilized by higher layers
to implement PRMA. Due to this, variations of PRMA, such as the PRMA-HS
mentioned earlier, can be also adopted in S-UMTS in order to improve the
overall system performance. Anyway, it should be reminded that PRMA may
only be used in LEO satellite systems.
5.2.4 Stability analysis of access protocols
The behavior of S-ALOHA-like protocols, such as the protocol used for

PRACH or the PRMA-like variants used for S-UMTS, calls for a suitable
design of the access protocol parameters.
As for the PRACH access protocol, 3GPP MAC specications do not
Fig. 5.6: PRMA-like access protocol (voice trac).
provide a specic scheme to determine explicitly the ASC conguration for

dierent trac classes. However, access control could be coordinated by the
satellite Earth station in order to dene dynamically the access characteristics.
This should be implemented by means of the feedback BCH signal. The studies
made in [13],[16],[17] could be exploited to optimize both the access delay (for
the dierent trac classes) and the energy consumption during the access
phase.
Also PRMA-like protocols need suitable settings for the control parame-
ters. In particular, the permission probabilities can be used to modify the
backlog period (after a contention) or to refrain a terminal from attempting
transmissions. The problem is that a too aggressive protocol may lead to
protocol bi-stability (i.e., too many collisions occur so that the throughput
of correctly carried out requests goes to zero). This is a critical problem
especially when many UTs contend for the same (access) resources [13],[17]. It
is therefore important to adopt an explicit cross-layer scheme that dynamically
adjusts transmission probability values on the basis of dierent aspects, such

as the characteristics of each trac class, the radio channel behavior and
trac load conditions. Analytical studies as those carried out in [13] can
provide the appropriate framework for the cross-layer (adaptive) design of the
access protocol parameters.
5.3 Downlink: scheduling techniques

5.3.1 Survey of scheduling techniques
The nature of the scheduling mechanisms employed on network links greatly

impacts the QoS levels that can be provided by a network. The basic
function of the scheduler is to arbitrate between packets that are ready for
transmission on the link. Based on the scheduling algorithm, as well as the
trac characteristics of the ows multiplexed on the link, certain performance
measures can be obtained. These can then be used by the network to provide
end-to-end QoS guarantees.
First In First Out (FIFO) is not only the simplest scheduling policy, but
also the most widely deployed one in the Internet today. As its name suggests,
FIFO (or else First Come First Served, FCFS) serves packets according to
their arrival order. This scheduling policy does not provide any guarantees to
end-users.
Fixed priority mechanisms between two or more classes aim to provide
the lowest possible delay for the highest priority class. The link multiplexer
maintains a separate queue for each priority. The scheduler sends the data
from the highest priority class before sending data for the next class. A packet
in a lower priority queue is served only if all the higher priority queues are
empty. As each queue is served in an FCFS manner, xed priority schedulers
are almost as simple as the FCFS scheduler with the added complexity
of having to maintain queues. While this scheduling policy oers service
dierentiation, care should be taken in order not to starve lower priority
classes. Moreover, it should be noted that xed priority mechanisms do not
readily allow end-to-end performance guarantees to be provided on a per-class
basis.
Weighted Round Robin (WRR) [18] aims to give a weighted access to
the available bandwidth to each class, ensuring a minimum allocation and
distribution. The scheduler services each class in a round-robin manner
according to the weights. If one or more classes are not using their full
allocation, the unused capacity is distributed to the other classes according to
their weights. A class can achieve a lower eective delay by giving it a higher
weighting than the trac level it is carrying.
Class-Based Queuing (CBQ) or Hierarchical Link Sharing (HLS) [19] is a
more general term for any mechanism that is based on the class. Each class is
associated with a portion of the link bandwidth and one of the goals of CBQ
is to guarantee roughly this bandwidth to the trac belonging to the class.

Excess bandwidth is shared in a fair way among the other classes. There is no
requirement to use the same scheduling policy at all levels of a link sharing
hierarchy.
Generalized Processor Sharing (GPS) [20] is an idealized uid discipline
with a number of very desirable properties, such as the provision of minimum
service guarantees to each class and fair resource sharing among the classes.
End-to-end guarantees on a per-class basis can be provided if the trac
characteristics of the classes are known. Due to its powerful properties, GPS
has become the reference for an entire class of GPS-related packet-scheduling
disciplines, and relatively low cost implementations have started reaching the
market. Weighted Fair Queuing (WFQ) [21] and its variants similarly aim
to distribute available bandwidth over a number of weighted classes by using
a combination of weighting and timing information to select which queue
has to be served. The weighting eectively controls the ratio of bandwidth
distribution between classes under congestion. However, it has been shown [22]
that the tight coupling between rate and delay under GPS in the deterministic
setting leads to sub-optimal performance and reduced network utilization.
The Earliest Deadline First (EDF) is a dynamic priority scheduler, with
an innite number of priorities. The priority of each packet is given by its
deadline. EDF has been proven to be optimal [23] in the sense that, if a
set of tasks is schedulable under any scheduling discipline, then the set is
schedulable under EDF as well. EDF scheduling in conjunction with per-class
trac shaping permits the provision of end-to-end delay guarantees.
The Service Curve-based Earliest Deadline rst policy (SCED) [24] is
based on service curves, which serve as a general measure for characterizing
the service provided to a user. Rather than characterizing service by a single
number, such as minimum bandwidth or maximum delay, service curves
provide a wide spectrum of service characterization, specifying the service by
means of a function. It is shown that the SCED policy has greater capability
to support end-to-end delay-bound requirements than other known scheduling
policies.
Scheduling techniques for wireless systems
The approaches presented above are designed according to specic goals

in terms of fairness and service requirements, without taking into account
the transmission media. At present, the success of wireless networks pushes
towards the design of scheduling techniques that, not only are aware of the
characteristics of the transmission channel, but might also take some prot of
this knowledge to achieve better performance.
Wireless systems are characterized by time-varying and location-dependent
link states conditioned by interference, fading and shadowing. As a result,
wireless channels are error-prone. This aspect has been considered in the
literature from dierent perspectives in order to design scheduling techniques
suited for wireless environments.

A rst approach consists in the emulation of an error-free channel by
deferring transmissions of user terminals experiencing bad channel conditions,
and compensating them when their channels are again in a good state. Among
the users in a good channel state, a scheduler suited for wired systems is
typically considered. Examples of this approach are the Idealized Wireless Fair
Queuing, the Channel Condition-Independent Fair Queuing, the Server-Based
Fairness Approach and the Wireless Fair Service scheduler. These techniques
are described below referring to a channel with BAD and GOOD states,
interpreting them as error channel and error-free channel, respectively.
The Idealized Wireless Fair Queuing (IWFQ) simulates an error-free
channel by applying a compensation model on top of the WFQ scheduler [25].
A start tag and a nish tag are associated to each packet, as for WFQ. The
ows are serviced according to increasing service tags of the ows perceiving
error-free channels. The compensation model operates as follows: if a ow
receives service in one round, its service tag is increased by a factor l (lead
bound); furthermore, for each round that a ow experiences a BAD channel,
its service tag is decreased by a factor bi (lag bound). This way, ows, which
are in error channel during some time, are able to capture the resources as
soon as they experience error-free channel, since their service tag is very low.
The drawback is that leading ows, i.e., those with higher service tags, might
be starved for long periods and therefore QoS bounds cannot be guaranteed
and the service degradation is abrupt.
Similar to IWFQ, the Channel Condition-Independent Fair Queuing (CIF-
Q) simulates an error-free channel by applying a compensation model on
top of the Stochastic Fairness Queuing (STFQ, proposed in [26] as an
enhancement of WRR). The compensation model applied here avoids abrupt
service degradation. A lag parameter l is assigned to each ow, which is
positive if the ow is lagging and negative if the ow is leading. In principle,
ows are scheduled according to STFQ; however, if a ow i in error channel
has allocated resources, the scheduler looks for other backlogged ows that
perceive an error-free channel. If a ow j is found that fullls this requisite,
the ow i gives way to ow j, and their lag parameters are updated: li is
incremented and lj is decreased [25]. Hence, ow i still receives a fraction of
its service, yielding to a graceful service degradation.
In Server-Based Fairness Approach (SBFA), a specic amount of transmis-
sion bandwidth is reserved for compensation purposes only. This is achieved
by creating a virtual ow called Long-Term Fairness Server (LTFS) that will
be used to manage the compensation. If a ow cannot be served because it
experiences BAD channel conditions, the corresponding packet is queued in
the LTFS. The scheduler treats the LTFS ow the same way as any other ow
for the channel allocation. The share of bandwidth corresponding to LTFS
is determined by a weight relative to the total bandwidth (as in a WRR
approach). Since the lag of a ow is not bounded and the packets in the LTFS
ow are served according to a FIFO policy, no packet delay bounds can be
guaranteed.
Applying the Wireless Fair Service scheduler, each ow i has a lead bound
of li,max and a lag bound of bi,max . Each leading ow relinquishes a portion of
its lead li /li,max for lagging ows. On the other hand, each lagging ow gets a
fraction
of the aggregated relinquished resources that is proportional to its lag:
bi / iS bi , where S is the set of backlogged ows [25]. In practice, the leading
ows free their resources in proportion to their lead and those resources are
fairly distributed among the lagging ows. This approach achieves fairness, as
well as delay and bandwidth guarantees.
The above scheduling techniques assume a simplied two-state channel
model representing an error state and an error-free state. A more realistic
model is to consider that each channel state is associated with a certain error
probability, which allows for more exibility in scheduling decisions. Based
on this assumption, several scheduling techniques have been proposed in the
literature, driven by the comparison of the channel quality level experienced
by the user terminals having backlogged packets. Detailed examples of these
techniques are reported below referring to the UMTS scenario.
Packet scheduling in UMTS
In case of CDMA cellular systems, the resources are the bandwidth, the codes,
the RLC buers at the RNC node and the UE, and the transmit power.
In UMTS, the packet scheduler works in close-cooperation with the other
resource management functions, in particular the admission control and the
load (congestion) control entities [27]. Scheduling is part of the congestion
control function, namely it is a form of reactive resource management, as
opposed to the proactive characteristic of admission control. The packet
scheduler can decide the allocated bit-rates and the length of the allocation
among users. In W-CDMA, this can be done in several ways, in a code or time
division manner or power scheduling-based.
In the code division approach, a large number of users can have a low
bit-rate channel available simultaneously. When the number of users wanting
capacity increases, the bit-rate, which can be allocated to a single user,
decreases. In time division scheduling, the capacity is given to one user or
only to a few users at each time instant. A user can have a very high bit-rate,
but can use it only very briey. When the number of users increases in the time
division approach, each user has to wait longer for transmission. Power-based
scheduling may be employed in response to the condition of the radio link
between sender and receiver. If the power devoted to a code is kept xed, the
possible supported rate for a given transmission quality (interpreted in this
context in terms of Eb /I0 ) increases for GOOD and decreases for BAD channel
conditions. Likewise, if the information rate is kept constant, maintaining the
same transmission quality is obtained by employing two dierent levels of
transmit power (i.e., the concept of power control).
The most common packet schedulers for UMTS are described below. The
Maximum C/I Scheduler serves in each resource allocation interval the ow

with the best Carrier-to-Interference ratio (C/I ) [28]. This approach is unfair,
provided that ows corresponding to users located in the coverage edge have
in general poor C/I performance and are starved in general, experiencing
uncontrolled long delays.
The C/I Proportional Scheduler (C/I PS) also serves the ow with the best
channel quality among backlogged ows. The main dierence to the Maximum
C/I Scheduler is that once a ow gets the resource, it is served until its queue
is empty. This method does not guarantee fairness and QoS; it just maximizes
channel eciency and in turn network throughput. Furthermore, users with
poor channel conditions might remain in a waiting status for a long time,
experiencing very high delays.
Finally, more advanced scheduling techniques have been proposed, that
operate on the basis of trade-o criteria (throughput vs. fairness) and even
prot from developments in the elds of digital modulation and forward error
correction. Examples of these approaches are the Proportional Fair scheduler
and an enhanced version of it, named Exponential Rule scheduler.
The Proportional Fair (PF) scheduling algorithm has been originally
developed to oer an appealing trade-o between user fairness and cell
capacity in terrestrial High Speed Downlink Packet Access (HSDPA) as
well as in CDMA/HDR [29]-[31] (see also the following sub-Section). With
this approach, the server retrieves information about the instant quality of
the downlink channel (Channel Quality Indicator, CQI). According to this
CQI measure, the server calculates for each ow in every scheduling round
the Relative Channel Quality Index (RCQI) that is a trade-o measure
between the maximum throughput that the ow can achieve (according to
the modulation and coding rate it can aord) and the service it got in the
past. The scheduler serves in each resource allocation interval the ow with
the highest RCQI value. The maximum achievable throughput by a ow is
determined by the highest modulation and highest coding rate that can be
applied according to the experienced channel conditions, reported in the CQI.
The RCQI parameter provides a trade-o between channel eciency and
fairness, avoiding that users with good enough channel are starved due to
the presence of users with better channel conditions. However, delay bounds
cannot be guaranteed with this approach.
The Exponential Rule scheduler introduces enhancements to the PF scheme
that aim at balancing the weighted delay of all backlogged ows when the
dierences of weighted queue delay among users become signicant [31]. This
is achieved by adding a multiplicative exponential parameter to the RCQI
metric. The exponential function is dependent on the weighted instantaneous
delay compared to the cumulative delay. If a signicant increase on the delay
is detected, the function gets a high value (due to its exponential prole) that
increases the nal value of the RCQI metric, thus giving high priority to that
user in front of the others. In addition to the trade-o between fairness and
transmission eciency, this scheduling technique provides also guarantees in
terms of delay bounds.

Although satellite systems can be considered as a specic case of wireless
systems, additional eects might have impact on the scheduling performance,
such as the propagation delay and channel state dynamics dierent from the
terrestrial case. These issues are considered in the following sub-Sections.
5.3.2 Scheduling techniques for HSDPA via satellite
Overview on terrestrial HSDPA
HSDPA is a step beyond the W-CDMA air interface, in order to improve the
performance of downlink multimedia data trac according to the increasing
demand for high bit-rate data services. For that purpose, the main targets of
HSDPA are to increase user peak data rates, to guarantee QoS and to improve
spectral eciency for downlink asymmetrical and bursty packet data services,
supporting a mixture of applications with dierent QoS requirements [28].
The HSDPA concept is based on an evolution of the Downlink Shared
Channel (DSCH), denoted as High Speed-DSCH (HS-DSCH). DSCH time-
multiplexes the dierent users and is characterized by a fast channel recon-
guration time and a packet scheduling procedure, which is very ecient
for bursty and high data rate trac in comparison with DCH. HS-DSCH
introduces several adaptations and control mechanisms that enhance peak
data rates, and spectral eciency for bursty downlink trac.
The HS-DSCH structure is based on a Transmission Time Interval (TTI)
whose duration is selected on the basis of the type of trac and the amount
of users supported (in the order of 2 ms). In comparison with the typically
longer TTIs of W-CDMA (10, 20 or 40 ms), the shorter TTI in HSDPA allows
for lower delays between packets, multiple retransmissions, faster channel
adaptation and minimal wasted bandwidth.
Two fundamental CDMA features are disabled in HS-DSCH, i.e., fast
power control and Variable Spreading Factor (VSF), being replaced by other
features such as Adaptive Coding and Modulation (ACM), multi-code opera-
tion, Fast L1 hybrid ARQ (FL1-HARQ) and xed spreading factor equal to 16
[28]. The xed spreading factor allows the allocation of 15 codes in each TTI
(the 16th code is used for signaling purposes) that can be assigned to either
the same UE to enhance its peak data rate or several UEs code-multiplexed
in the same TTI.
Furthermore, in order to achieve low delays in the link control, the MAC
layer functionality corresponding to HS-DSCH (namely MAC-hs) is placed in
the Node-B (instead of the RNC, where the MAC layer functionality corre-
sponding to DSCH is typically located). This solution allows the scheduler to
work with the most recent channel information, so that it is able to adapt
the modulation scheme and coding rate to better match the current channel
conditions experienced by the UE. However, this solution introduces some
changes in the interface protocol architecture, as depicted in Figure 5.7 [32].
Fig. 5.7: Interface protocol architecture of HSDPA.
The adaptability of HS-DSCH to physical channel conditions is based

on the selection of a coding rate, a modulation scheme, and the number of
allocated codes to the scheduled UE in each TTI. In particular, the HS-DSCH
encoding scheme is based on the Release99 rate-1/3 turbo encoding, but adds
rate matching with puncturing and repetition to obtain a high resolution on
the eective code rate, ranging approximately from 1/6 to 1. To facilitate
very high peak data rates, the HSDPA concept has added 16QAM on top of
the existing QPSK modulation available in Release99. A modulation scheme
and code rate combination is denoted as Transport Format and Resource
Combination (TFRC). Under very good channel conditions, the selection of
highly ecient TFRCs combined with the allocation of several orthogonal
codes to the scheduled UE (multi-codes operation), allow the UE to receive
theoretically up to 10 Mbit/s [28]. However, this might be constrained by the
UE capabilities, due to the limitation of receiving several parallel codes [33].
The packet scheduler can be considered as the central entity of the HSDPA
design. In the HSDPA protocol stack architecture, the packet scheduler is
located in the MAC-hs at the Node-B. The tasks corresponding to the MAC-hs
layer in the UE and in the Node-B are summarized in Table 5.2.
According to a certain packet scheduling algorithm, the HS-DSCH trans-
port channel is mapped onto a pool of physical channels, High Speed Physical
Downlink Shared Channels (HS-PDSCHs), to be shared among all the HSDPA
users in a time-multiplexed way.
The scheduler governs the distribution of the available radio resources
in the cell among the UEs, i.e., it selects which UE is scheduled in the next
TTI and which settings should be used (TFRC and number of parallel codes),
supported by the link adaptation functionality. The scheduler relies on channel
state information sent from each UE in order to perform its functions. The UE
MAC-hs in UE MAC-hs in Node-B

Generation of ACK and NACK MAC PDU ow control
responses to received packets
Routing of packets to the correct Scheduling and priority handling
reordering queue based on
queue identier
Reordering of PDUs Request of retransmission if
NACK received
Removing of MAC-hs header Selection of appropriate transport
and padding bits format and resource combination
Table 5.2: MAC-hs functions in UE and Node-B.
is requested by the RNC to send periodically a specic CQI on the uplink High
Speed Dedicated Physical Control Channel (HS-DPCCH). The periodicity is
selected from the set {2, 4, 8, 10, 20, 40, 80, 160} ms. The CQI provides the
following information related to the currently experienced channel conditions
by the UE [7]:
TFRC mode (most ecient modulation scheme and coding rate that can
be used);
Maximum number of parallel codes that can be used by the UE;
Specication of a transport block size (i.e., the transport layer PDU) for
which the UE would be able to receive data with a guaranteed FER lower
than or equal to 10%, after rst transmission.
There are dierent CQI tables for several UE categories. Table 5.3 shows
an example [34]. If CQI indicates that the quality is degrading, the scheduler
can choose a less ambitious TFRC that will cope better with the poor channel
conditions.
Implications of the satellite component in HSDPA
The HSDPA concept and architecture have been designed for terrestrial
environments. In a satellite scenario, the allowed complexity on board of
the satellite, the selected constellation (LEO, MEO, GEO) and a dierent
propagation environment condition the applicability of the HSDPA concept
as it is dened and the feasibility of the promised peak data rates.
One of the major advantages of HSDPA with respect to the W-CDMA
interface is the location of the scheduling function at the Node-B, allowing
for shorter delays and better adaptability to time-varying channel conditions.
However, the location of the dierent network entities, such as Node-B or
RNC, is not uniquely determined in a satellite-based UMTS system. Depend-
ing on the available complexity on the satellite, part of the functionalities
typically located at the Node-B or at the RNC in a UMTS network can be
CQI Modulation Number of Bits per TTI

value and coding codes used (transport
per TTI block size)
1 1 137
2 QPSK 1/3 1 173
3 (on each code 1 233
4 960 bits are 1 317
5 sent in a TTI) 1 377
6 1 461
7 2 650
8 2 792
9 2 931
10 3 1262
11 3 1483
12 3 1742
13 4 2279
14 4 2583
15 5 3319
16 5 3565
17 16QAM 1/3 5 4189
18 (on each code 5 4664
19 1920 bits are 5 5287
20 sent in a TTI) 5 5887
21 5 6554
22 5 7168
23 7 9719
24 8 11418
25 10 14411
26 12 17237
27 15 21754
28 15 23370
29 15 24222
30 15 25558
Table 5.3: Example of CQI mapping in transport block size for TTI = 2 ms
(terrestrial standard); the highlighted CQIs are those considered for simulations
referring to a GOOD/BAD channel model.
executed on board or not. In the case of a bent-pipe satellite, all medium access
control mechanisms must be located at the Gateway station or the network
control center. In any case, the large distances involved in a satellite system
disable the HSDPA capabilities of fast retransmissions and quick adaptation to
physical channel variations, thus scaling the performance that link adaptation
mechanisms can achieve.
In GEO satellite systems, retransmissions take too long time. Therefore,
FER upper bounds should be adequately much lower in order to reduce
statistically the number of required packet retransmissions.

Furthermore, the behavior of the channel is not comparable to the ter-
restrial mobile channel: deeper and longer fades are expected in the satellite
case, in contrast to the fast fades of the terrestrial mobile channel.
All the issues discussed above condition the performance of the packet
scheduler, which is the main entity of the HSDPA concept. With the purpose
of testing the performance of dierent scheduling techniques in a simplied
satellite-HSDPA scenario, the following assumptions have been made:
A multi-beam GEO bent-pipe satellite has been considered.
All Radio Access Network (RAN) functionalities corresponding to the
network part are located at a Gateway station, as can be observed in
Figure 5.8.
The propagation delay between Gateway station and UE is approximately
280 ms, i.e., round-trip propagation delay is about 560 ms.
Each UE performs channel estimation. The result is sent back (in the form
of CQI) to the Gateway station.
CQI information transmission interval is extended to 40 ms in order to save
power at the UEs (this is not that critical, considering that the impact of
round-trip delay in the acquisition of channel state information should be
more dominant than this larger periodicity).
During the time interval between two CQI updates, channel conditions are
considered constant by the scheduler for a given UE.
The TTI duration of the terrestrial HSDPA is kept also in the satellite
case in order to reduce packet delays and to have ne scheduling time
granularity.
A GOOD/BAD Markovian channel model is considered at the physical
layer for the sake of simplicity. Accordingly, one CQI value from Table
5.3 is selected for each channel state: CQI = 15 for the BAD state and
CQI = 25 for the GOOD state. Note that the channel variation dynamics
in a satellite environment are slow; in particular, a mean GOOD (BAD)
sojourn time of 6 s (2 s) has been considered.
Code-multiplexing of dierent users in the same TTI is not applied in this
simplied study, i.e., only one UE is served in each TTI. According to this
assumption, the task of the scheduler is to select the UE to be served in
each TTI. The service got by the scheduled UE depends on the transport
block size determined by the CQI currently supported by the UE (see
Table 5.3).
On the basis of the assumptions above, the channel state information that
the UE transmits to the Gateway station is outdated when received at the
Gateway due to the high propagation delay. To cope with this, either higher
margins in the selection of the CQI value to be sent shall be considered or
delay compensation strategies shall be applied that permit to predict what
will be the channel evolution by the time that the CQI information reaches
the Gateway. For the interested reader, some delay compensation techniques
Fig. 5.8: S-HSDPA network architecture.
are proposed in [35] for Ku and Ka band satellite links.

If no countermeasures are adopted, channel state transitions cause temp-
oral misalignments between the current channel state and the considered CQI
by the Gateway station. In particular, in the presence of a transition from
BAD to GOOD, the system uses a more conservative mode than necessary
for 560 ms plus maximum 40 ms (1 ). This does not aect the service quality
(i.e., no packet losses are caused), but the resource utilization is not optimal.
On the other hand, in the presence of a transition from GOOD to BAD,
the system does not adequately protect transmissions during 560 ms plus
maximum 40 ms, so that the transmitted data during this period is lost due
to channel impairments with high probability. For the sake of simplicity, it
is assumed that FER = 1 during misalignment periods from GOOD to BAD
channel states.
The trac scenarios may also aect the resource utilization performance
achieved by any scheduling technique. If the trac generated by the scheduled
UE in the next TTI is not sucient to ll the transport block assigned for
the transmission, part of the capacity remains unused and certain ineciency
is experimented.
PHY-aware scheduling approaches for HSDPA over satellite
The design of suitable scheduling techniques for HSDPA-like transmissions in

a satellite environment must consider the several degrees of freedom imposed
1
Depending on the CQI transmission timing with respect to the current state
transition (GOOD to BAD or vice versa), the delay to receive a packet with an
updated TFRC ranges from 560 to 600 ms.
by the HSDPA interface in addition to the specic characteristics of satellite

links. HSDPA has a sort of hybrid TDM/CDM air interface, where packet
scheduling can be done in two dimensions: time and code. The code dimension
allows for two avors of resource management strategies: code-multiplexing
and multi-code operation. By means of code-multiplexing, several UEs can be
scheduled in the same TTI, thus enhancing the resource utilization. Using the
multi-code operation, the throughput of one UE can be improved on a TTI
basis by allocating several codes to it.
Based on the CQI information periodically sent by each UE, the scheduler
can nd out the achievable throughput by each UE in the next TTI by checking
a look-up table like Table 5.3; this scheme is considered here like an explicit
cross-layer technique that envisages the dynamic interaction of physical and
MAC layer. The throughput achievable by each UE is determined by the
most ecient applicable modulation and coding rate, the maximum number
of codes that can be allocated to the UE and the transport block size that can
be used. It should be noted that the eective code rate must be calculated
taking into account the Cyclic Redundancy Check (CRC) code of 24 bits that
is added to each transport block before encoding and additional puncturing
and repetition, which yield a number of physical layer bits equal to:
960 bits number of assigned codes, if QPSK modulation is used;

1920 bits number of assigned codes, if 16QAM modulation is used.
The eective code rate is given by:
24 + (transport block size)

Code rate (CQI) = (5.2)
bcode (number of codes)
where bcode = 960 bits for QPSK modes and bcode = 1920 bits for 16QAM
modes.
For the method adopted in HSDPA to pass from transport to physical
layer, the interested reader should refer to [36].
The availability of channel state information and the relation between
channel state and achievable throughput by a UE adds new dimensions
for optimization to the scheduling problem. Typically, a scheduler manages
the share of resources among ows accessing the media according to some
fairness or QoS criterion. However, in a system that supports ACM and
code-multiplexing on top of time-multiplexing (and multi-code operation),
the scheduler operation becomes even more complex. Several approaches
can be adopted in the design of scheduling techniques, depending on the
optimization goals. We consider here some of those schemes already introduced
in sub-Section 5.3.1.
A rst approach is to ignore the additional degrees of freedom of HSDPA
and to schedule the backlogged trac according to either a fairness criterion
or driven by QoS constraints. In this case, algorithms such as EDF can be
applied. However, this approach does not exploit the exibility of HSDPA to
use eciently the available resources, since the channel state corresponding
to each ow is transparent to the scheduler.
A second approach is to maximize the transmission eciency by scheduling
the ows that can achieve the highest throughput in the current TTI, i.e.,
those ows that are associated to better channel conditions, which is the
strategy applied by the opportunistic scheduler [37]. However, this approach
does not guarantee QoS, since those UEs with worse channel conditions shall
be blocked for long periods, even if their channel state is good enough for
transmission.
A third approach is to schedule the ows according to a hybrid criterion
that combines fairness and transmission eciency maximization in a trade-o
manner. This concept has been proposed for scheduling in HSDPA in terres-
trial environments under the name of PF scheme [30],[31] (see sub-Section
5.3.1).
A scheduler has been considered to manage downlink transmissions (HS-
DSCH) that is in the Node-B (Earth Station) according to the architecture in
Figure 5.8. In particular, the scheduler is at MAC-hs level and it is assumed to
have dierent queues for the dierent UEs. Each queue (at IP level) contains
the multimedia trac corresponding to one UE. Suitable priority indexes are
considered to serve these queues; these indexes are related to either the EDF
scheduler or the PF scheme. In what follows, the performance achieved by
these schedulers are compared in the presence of video streaming and Web
trac [38],[39]. The assumptions previously made (see the previous part on
Implications of the satellite component in HSDPA) are considered for this
simulation study.
EDF scheduler
This scheduling technique, described in sub-Section 5.3.1, serves packets

according to their urgency. The EDF scheme is quite appropriate for the
management of real-time trac ows that are characterized by deadlines.
Such scheme requires the dynamic management of the buer for each trac
class when packets with dierent deadline values have to be served.
To implement the EDF criterion it has been considered that the priority
index for the generic k -th UE in the current n-th TTI interval, Pk [n], is given
by the ratio between the transmission delay of its oldest IP packet, dk [n], and
the packet deadline, Tdeadline :
dk [n]
Pk [n] = k = 1, 2, ... , N (5.3)
Tdeadline
where N denotes the number of UEs per spot-beam.
The above priority index does not permit to prioritize the real-time video
trac with respect to the interactive Web trac. This approach could degrade
the video performance in the presence of signicant Web trac load. To cope
with this, a dierentiation in the priority index in equation (5.3) is needed.
In particular, equation (5.3) is used for video trac so that a video IP-packet
has an increasing priority up to (almost) 1 when the packet is close to its
deadline and risks to be dropped. Moreover, a modied priority index is used
for Web IP-packets that saturates to 0.9 when these packets are close to (or
exceed) their virtual deadline:

Dk [n]
Pk [n] = min 0.9, k = 1, 2, ... , N. (5.4)
Tdeadline
Hence, very urgent video packets will be served with highest priority than
any Web packet. In what follows, the scheme where the priority index (5.3) is
used for both video and Web trac ows will be denoted as EDF; whereas, the
name Prioritized-EDF (P-EDF) is applied to the scheme where the priority
index (5.3) is used for video ows and the priority index (5.4) is adopted for
Web trac ows.
PF scheduler
This strategy serves the UE with largest RCQI, which represents the ratio
between the maximum data rate currently supported by each UE (according
to its CQI and by using a look-up table like Table 5.3) and the average
service that the UE got in the past, according to a suitable sliding window.
On the basis of [30], the RCQI value corresponding to the k -th UE can be
computed as follows.
Rk [n]
RCQIk [n] = k = 1, 2, ... , N (5.5)
Tk [n]
where n is related to the time measured in TTI units, Rk [n] is the bit-rate
supported by the k -th UE in the n-th TTI interval (depending on its current
CQI) and Tk [n] represents the average throughput achieved by the k -th UE
up to the present TTI (according to a dened memory length).
Rk [n] and Tk [n] can be computed as follows [30]:

Bk [n]
Rk [n] = min CQIk [n], (5.6)
TTI

1 1
Tk [n] = 1 {Bk [n] > 0} Tk [n 1] + Rk [n 1] (5.7)
Nk Nk
where CQIk [n] denotes the maximum bit-rate supported by the k -th UE
at the current time, calculated as the throughput that is allowed by the CQI
in the next TTI interval (according to a look-up table like Table 5.3). Bk [n]
represents the amount of data waiting for transmission in the Node-B buer
of the k -th UE at current time; {Bk [n] > 0} is either 1 or 0 depending on
whether the Boolean expression is right or not. Nk represents the memory

of the averaging lter (which has been set to 1000 TTI units), and Rk [n1]
denotes the bit-rate used for the transmission to the UE during the (n1)-th
scheduling interval. It is assumed that Tk [1] = CQIk [1].
According to the assumptions made on the GOOD/BAD channel (i.e.,
CQI = 15 for the BAD state and CQI = 25 for the GOOD state) and on the
basis of Table 5.3, we have the corresponding bit-rate capacities:
CQIk [n] = Rbad = 3319 bits/TTI 1.6 Mbit/s in the BAD state;
CQIk [n] = Rgood = 14411 bits/TTI 7.2 Mbit/s in the GOOD state.
Note that in the PF case an explicit cross-layer scheme has to be adopted

since scheduling takes into account the dynamic variation of the radio channel
conditions for the UEs.
The software simulator presented in [38],[39] has been used to evaluate the
performance of S-HSDPA transmissions, using both EDF and PF techniques
in order to manage video streaming and Web trac downlink ows.
S-HSDPA performance: simulation results
In order to evaluate the performance of S-HSDPA transmissions when using

EDF, P-EDF and PF schedulers, the following metrics have been considered:
Eciency in the utilization of radio resources, ;

Percentage of IP-video packets lost due to deadline expiration, Pdrop ;
Percentage of IP packets lost due to GOOD-to-BAD channel misalignment,
Ploss channel (without considering packet retransmissions);
Mean delay for the transmission of an IP-Web packet, DelayW eb .
Let C denote the mean capacity considering the GOOD/BAD channel

previously described [39] and the related CQI values associations in Table 5.3.
Hence, the resource utilization eciency (<1) can be measured as follows:
Mean aggregated transmitted bit-rate

= . (5.8)
C
Pdrop is obtained as the ratio between the number of IP-video packets
that are lost due to deadline expiration (the deadline has been set to 150
ms) and the number of generated IP-video packets. Ploss channel is computed
as the ratio of the number of IP packets that are lost at the receiver due to
the GOOD-to-BAD channel misalignment (considering both video and Web
trac together) and the number of transmitted IP packets.

In the following graphs, the above dierent performance metrics are
plotted as a function of the system load, , that is dened as:
Mean aggregated generated bit-rate

= [Erl] . (5.9)
C
Of course, due to the fact that not all the generated bits are
transmitted (some of them may be dropped due to deadline expiration in
the case of video trac).
The following simulation results have been obtained considering an equal
number of video and Web trac sources; both video and Web sources pro-
duce the same mean bit-rate (variable parameter) according to the formulas
detailed in [39]. Each simulation run corresponds to 4104 s. Moreover, we
have used Tdeadline = 150 ms for video packets and Tdeadline = 2 s for Web
packets (2 ).
Figure 5.9 shows the Pdrop behavior as a function of for PF, EDF and
P-EDF schedulers, when the total number of trac sources is equal to 30. All
these scheduling schemes employ the physical layer adaptability, but PF and
EDF achieve extremely poor Pdrop performance since they do not include a
strategy to provide a strong prioritization of video trac with respect to the
Web one. Whereas, P-EDF attains a low Pdrop value that permits to fulll
the Pdrop requirement ( 1%) up to about equal to 1 Erlang.
Figure 5.10 shows the DelayW eb behavior as a function of for PF, EDF
and P-EDF schedulers in a scenario where the total number of trac sources
is equal to 30. As expected, EDF and PF schemes allow the lowest DelayW eb
values; the high DelayW eb values with the P-EDF scheme are due to the strong
prioritization of video trac that entails higher transmission delays for Web
trac.
In the following graphs, the performance comparison is focused on P-EDF
and PF techniques. Figure 5.11 presents the comparison of as a function of
for PF and P-EDF in a scenario where the total number of trac sources is
equal to 30. It can be observed that P-EDF allows a better eciency than PF
since it permits to achieve a lower Pdrop value. Finally, Figure 5.12 provides
Ploss channel results as a function of for PF and P-EDF for cases with total
number of trac sources equal to 30. The obtained results show that all the
Ploss channel values are quite close and around 7%, a limit loss value that could
be still tolerated by some recent video codecs, such as H.263 used in UMTS.
It should be noted that the PF scheduler in some way may provide a
more frequent service to UEs in the GOOD state than P-EDF. Hence, with
PF, it could be more probable to schedule a UE that is changing its state
from GOOD to BAD (thus incurring in packet losses due to channel state
2
Video packets exceeding the deadline are dropped; while, Web packets exceeding
their deadline are sent anyway since they are related to interactive trac.
Fig. 5.9: S-HSDPA results in terms of Pdrop for video packets.
Fig. 5.10: S-HSDPA results in terms of DelayW eb for Web packets.

Fig. 5.11: Resource utilization comparison as a function of system load for P-EDF
and PF schemes.
Fig. 5.12: Ploss channel as a function of system load for P-EDF and PF schemes.
misalignment). These are the reasons why PF is characterized by a higher

Ploss channel value than P-EDF. Moreover, the higher the number of UEs, the
higher the probability to schedule a UE when a transition occurs from GOOD
to BAD.
In conclusion, the simulation results reported here prove that S-HSDPA is
feasible, provided that suitable scheduling functions and trac ow prioriti-
zation are employed.
5.3.3 Scheduling techniques for broadcast and multicast services

in S-UMTS
With the increasing use of high-bandwidth applications in 3G mobile systems,

especially with a large number of users receiving the same high data rate
services, ecient information distribution is essential. Thus, broadcast and
multicast are techniques to decrease the amount of transmitted data within
the network and to use resources more eciently. In particular, broad-
cast/multicast is a method for transmitting datagrams from a single source
to several destinations. Due to the broadcast nature of the radio channel, this
method is ecient for sessions sharing the same (or even common) contents. If
the nature of the oered service lends itself to spatial and temporal bundling
of the demands into one transmission, the benet of multicast and broadcast is
that data are sent just once by the network and transmitted to users, located
in the same cell, over a single common channel without clogging up the air
interface with multiple transmissions of the same data, as caused by multiple
usage of unicast sessions.
Due to the broadcast nature and ubiquitous coverage, satellite systems
may become a very ecient complement to terrestrial mobile networks, remov-
ing their asymmetric load and providing them with far more point-to-point
equivalent capacity for far less investment cost.
Design requirements
Requirements of broadcast and multicast services delivery and impact on

packet scheduler design
Even though the broadcast and multicast delivery mode is able to give
many benets for certain application areas such as inherently non-interactive
applications, e.g., video/audio streaming and le downloading applications in
the presence of a high user density (stadiums, trade shows, etc.), there are
still many challenging issues to be solved such as the resource management
for providing the QoS constraints with the same conditions for all members
in the same group.
UMTS allows a user or an application to negotiate the characteristics of
the service at connection set-up. The network may check whether sucient
resources are available, and returns the results to the application, which can
accept or deny the connection request according to a CAC scheme. After

admission of the connection request, the network should keep the performance
of the connection as contracted. This is also the case of broadcast and
multicast users. By admitting the connection request, the access network
has to make a choice for the type of the radio access bearer taking into
account several conditions, like attributes of the requested service, number
of group members in the cell, current load conditions etc. In contrast to
unicast, i.e., point-to-point service delivery, the network has to select the type
of the transport channel (i.e., a common channel or a dedicated one). For
instance, if there is only one multicast member in the cell, it is not worth to
use a common channel since a common channel needs additionally a return
link dedicated channel for maintaining the quality of the connection, i.e.,
measurement control/report, power control and the error correction due to
its unidirectional nature. In other words, the usage of a common channel
is not always more eective than that of dedicated channels. Therefore,
well-dened criteria for selecting the transport channel type among others
(e.g., the minimum number of members in the multicast group, momentary
load condition, current/predictable channel condition, QoS constraints of the
session and so on) are necessary in order to utilize optimally system capacity.
Since the number of members in a multicast session can be dynamically
changing, there should be another criterion for the appropriate timing when
a Radio Access Bearer (RAB) re-assignment will be necessary. Such criterion
will certainly aect the scheduling assignment. Another issue to consider is
on the method the transmission power should be (re)assigned to reect the
group dynamics of a multicast session, since users can join or leave a multicast
group at any time. Controlling the transmission power in a UMTS network is
crucial in maximizing the capacity that the network supports. This is due to
the fact that UMTS uses the CDMA technology, which is interference-limited.
In order to get a feedback channel for the power control, several methods can
be considered, such as: use of an additional bi-directional DCH between each
multicast member and the base station (i.e., Node-B) or usage of the RACH,
as specied in UMTS.
After the assignment of a certain RAB to the multicast session, the
network should maintain the contracted performance throughout the session.
In practice, it is considerable that the network has to maintain not only this
multicast session, but also other multicast sessions as well as other unicast
sessions, which have their requirements in terms of delay, throughput, jitter,
priority and so on. Moreover, especially for the satellite network, it is also
considerable that the group members are distributed with great distances
from each other. Hence, the selection of an appropriate Transport Format
(TF) has a strong impact on the performance of connections, not only the
multicast session itself, but also on the other active sessions due the generated
interference level. According to the W-CDMA channel sharing technique, for
each TTI, we have to decide how to accommodate datagrams over channels
by choosing an optimal, or sub-optimal, TF combination, for the currently
active sessions. This TF selection has to be done dynamically according to

the changing load conditions, the number of multicast members and the radio
propagation condition. Of course, the performance experienced by the most
of group members cannot be worsened by a minor number of them.
Reference scenario and impact on packet scheduler design
The provision of multimedia services in broadcast and multicast mode has

been regarded as a key for the ecient use of the precious wireless resources,
and is currently under standardization within the Multimedia Broadcast Mul-
ticast Services (MBMS) framework [2] in 3GPP. However, serious concerns are
expressed as to whether T-UMTS can cope with the additional requirements
of MBMS delivery on top of the other point-to-point T-UMTS services due
to the spectrum limitations and very limited means to improve the spectrum
eciency. On the other hand, satellites are a promising platform for MBMS
delivery due to their unique wide area coverage capabilities.
Considering that broadcast and multicast trac ows are asymmetric
in nature, the baseline satellite system architecture under consideration is
eectively unidirectional [40], as shown in Figure 5.13. It relies on the existing
3G mobile network point-to-point (p-t-p) service capability for the return
link to manage and to control the delivered services, for example for access to
content decoding keys and retrieval of multimedia content blocks corrupted on
the satellite forward link. The space segment consists of a GEO satellite that
features a transparent payload with multiple beams. This choice provides the
desired exibility in updating/enhancing the system throughout its life and is
accompanied by reduced technology and investment risks. In build-up areas
such as in urban and indoor environments, terrestrial repeaters/gap-llers
can be introduced to enhance the signal availability. They are designed to be
smoothly co-sited with 3G base stations (i.e., Node-Bs) to prevent additional
installation costs [41].
The UE+ considered here is a multi-mode terminal (i.e., satellite and
terrestrial 2G/3G radios), with frequency band extension. It is able to perform
parallel idle mode, i.e., maintaining either GSM activity or UMTS activity
during S-MBMS reception. The basic type does not have a dedicated receiver
for S-MBMS and is then required to switch from UMTS terrestrial to satellite
reception. The hub includes 3G RAN equipment (i.e., RNC) and 3G core
network functions. It collects incoming media services from the Broadcast
Multicast-Service Center (BM-SC) and generates the W-CDMA waveform
and redirects the signal to the satellite feeder link. The BM-SC provides
functions for S-MBMS user service provisioning and delivery; for example, it
controls user access to services, authorizes and initiates bearer services within
the network, schedules and transmits MBMS data across the network.
Given that there is no real-time interaction between the user and the
satellite RAN in the considered baseline architecture, the operation of the
packet scheduler is therefore dierent than in the previous S-HSDPA case.
Fig. 5.13: S-MBMS architecture and its interworking with a terrestrial network.
The packet scheduler in the unidirectional satellite system has to decide

on allocations without knowledge of the state of individual channels, i.e.,
channel state-dependent scheduling is not possible. In any cases, even if such
information were available, it would have to be exploited in a complex way due
to the point-to-multipoint nature of the services, i.e., the decisions regarding
the scheduling of a single service data ow need to consider the state of
multiple links corresponding to all the users that have activated the service
in each (multicast) group.
The role of the packet scheduler in S-MBMS is not that dominant in
determining the system throughput as in the T-UMTS case. Nevertheless, the
scheduler is still responsible for two important tasks that are executed with a
period equal to the TTI of the radio bearers [42]:
Time multiplexing of ows with dierent QoS requirements into xed
physical channels, in a way that can satisfy these requirements.
Adjusting the transmit power of the physical channel carrying the data
ows on the basis of the required reception quality of the service (in terms
of the target FER) under the constraint that the total available power for
all the physical channels within a beam is xed.
The packet scheduling strategy can be generally conceptualized into two
steps, as described in Figure 5.14.
These two steps eectively constitute the discipline of the packet scheduler.
Functional design of packet scheduler for multicast trac

Service prioritization
In MBMS, each service is one-to-one mapped onto an MBMS point-to-
multipoint Trac Channel (MTCH), a logical channel, which is then mapped
Fig. 5.14: Packet scheduling procedure.
onto the FACH transport channel. At the physical level, the Secondary
Common Control Physical Channel (S-CCPCH) can carry one or more
FACH(s). The incoming service requests are ordered according to some
priority criterion. In selecting the respective criteria, the service attributes are
considered, which are normally mapped onto the trac handling priorities,
as dened by the UMTS QoS classes. Note that the prioritization can be
more or less dynamic; in a more dynamic prioritization, the relative priority
of the dierent channels may change in each resource allocation interval (this
is normally the TTI), depending for example on the maximum delay tolerated
by a service or the number of packets buered.
We rstly describe a semi-dynamic prioritization performed at two levels.
The rst prioritization is static: the scheduler orders the services according
to their QoS classes (streaming, background) and the type of service delivery
(streaming, hot download, cold download), i.e., streaming service MTCHs
have higher priority than hot download service MTCHs, while hot download
MTCHs have higher priority than cold download service MTCHs, with both
download type of services belonging to the background class. Essentially,
this means that an explicit cross-layer design approach has been adopted
herein, whereby the upper layer information regarding the service attributes
are signaled down to the packet scheduler. In fact, QoS attributes are regarded
as the parameters from the application layer, which are used in the scheduling
entity, so that QoS-based scheduling can be considered as a cross-layer
approach. The second level of prioritization is related to the treatment of
MTCHs featuring the same level of priority, i.e., when there are two or more
MTCHs services having the same priority level. This prioritization is more
dynamic and two alternatives can be envisaged:
The rst one is based on the rotation of the serving order of the MTCHs at
each one of the three groups (streaming, hot download, cold download)
determined from the rst prioritization level. Separate lists are maintained
for each of these groups, whereby MTCHs are served according to their
current order in the list: the MTCH at the top of the list is served rst,
then the second one, etc. When an MTCH is served, it is removed from
the head of the list and is placed at the end of it, i.e., in a round-robin
manner.
The second scheme is based on the Service Credit (SCr ) concept, which
extends the idea of tokens from the leaky bucket algorithm to CDMA
packet-switched mobile communication systems. The SCr of a service
accounts for the dierence between the actual oered bit-rate (by the
scheduler) and the requested bit-rate, i.e., the guaranteed bit-rate for this
service. Hence, a service obtaining a higher bit-rate than requested has
SCr < 0, while a service obtaining a lower bit-rate than requested has
SCr > 0. In each TTI, the SCr for a service is updated as follows:
SCr [n] = SCr [n 1] + (Guaranteed rate/T B size)

(5.10)
T ransmitted T B [n 1]
where SCr [n] is the service credit at the current TTI, n, and is measured
in number of transport blocks per TTI; SCr [n-1] is the service credit in
the previous TTI; Guaranteed rate is the number of bits per TTI that
would be transmitted at the guaranteed bit-rate; TB size is the number
of bits in the Transport Block (TB) considered, and Transmitted TB [n-1]
is the number of successfully transmitted TBs in the previous TTI.
Obviously, this dynamic prioritization scheme is directly applicable to
streaming services, which feature a guaranteed rate attribute; however,
it may be expanded to download services even if they are not explicitly
characterized by the guaranteed bit-rate attribute (see Figure 5.14).
Rather than performing service prioritization in a semi-dynamic way,
a more ecient packet scheduling algorithm performs service prioritization
dynamically, depending on the waiting time/queuing delay experienced by
packets in each MTCH/FACH at the beginning of each TTI. Resource is
then allocated to respective physical channels (i.e., S-CCPCH) according
to the priority assigned to each MTCH/FACH ow as long as their power
and load condition can be satised. This scheduling scheme is named Delay
Dierentiation Queuing (DDQ) [43]. It is worth noticing that the packet
scheduling algorithm remains under the assumption of one-to-one mapping
from logical channels (MTCHs) to transport channels (FACHs).
DDQ is not a priority queue and is based on the Hybrid Proportional
Delay (HPD) scheduling scheme [44], which is widely used in the dierentiated
service networks. It assumes that there are QoS ratios between dierent QoS
priority classes. In each TTI, the serving indexes will be calculated for each
queue. These serving indexes are obtained based on the average waiting delay
for all the packets currently in the queue, the average queuing delay for all
the packets that have left the queue before this TTI, the packet arrival rate
and the QoS priority ratio index.
The mathematical formulation of DDQ can be expressed as follows. Let
i be QoS class factor, which is essentially a time-independent parameter
designated for each queue i. Let i (n) be the average queuing/waiting delay at
current n-th allocation instant (i.e., n-th TTI) for each queue i. This measure
describes the delay states of all packets passing through the respective queue,
including both the packets which are currently in the queue and those packets
which have already left the queue. The delay index is calculated for each queue
i in each TTI as in equation (5.11):

Nq
q
Nd
d
Wi,j [n] + Wi,j [n]
j=0 j=0
i [n] = (5.11)
Nq + N d
q
where Wi,j [n] is the waiting delay for the j -th packet currently in queue i ;
d
Nq is the number of packets in the queue; Wi,j [n] is the queuing delay for the
j -th packet, which has left queue i before this TTI (i.e., current time slot n);
Nd is the number of packets that have been served and left the queue before
this TTI.
For the service ow of the FACH queue i at the current time slot (i.e.,
TTI for UMTS) n, the priority is dened as:
Pi [n] = i i [n] . (5.12)
Consequently, the serving orders are calculated and assigned to each FACH
according to (5.12) at the beginning of each TTI.
With the above approaches of semi-dynamic and dynamic service priori-
tization in mind, the dynamically changing priorities of MTCHs indicate the
serving order of FACHs and S-CCPCHs for each TTI. It must also be noted
that it is generally assumed that only services with similar characteristics and
QoS requirements are multiplexed together to the same transport channel.
Resource allocation
Once all the services to be transmitted are prioritized, the next step is the
allocation of resources to them. This phase consists of bit-rate and transmit
power assignments within the specic resource allocation interval (i.e., TTI).
The data rate assignment consists in the selection of the Transport Format
Combinations (TFCs), which directly determine the per FACH transport
block size, namely how much data from each transport channel mapped to the
physical channel will be forwarded to the physical layer in TTI. For each active
physical channel (S-CCPCH), the exact TFC is selected from the Transport
Format Combination Set (TFCS), which is passed during the admission of a
new service as well as its mapping on a specic bearer. This TFC selection
step is of paramount importance since the capacity allocated to each service
is strongly related to the QoS perceived by the end-users, and, therefore,
the selection of the TFC has to take into consideration constraints in terms
of service requirements (e.g., minimum guaranteed rate, maximum tolerated

delay) as well as system-level constraints (system load, transmit power per
beam).
As for the power allocation, the transmit power setting for the S-CCPCH
is based on the required reception quality of the active service ows mapped
to S-CCPCH, which in our case is dened in terms of the most demanding
target FER among these service ows. The calculated power is only allocated
as long as it is within the constraint of the total available power for all the
physical channels, which is xed within a beam. In the resource allocation
phase, the per S-CCPCH TFC selection and power allocation are made in
parallel.
As illustrated in Figure 5.15, the description of the DDQ packet scheduling
scheme can be summarized as follows:
Fig. 5.15: Flowchart of DDQ scheme.

For all S-CCPCHs, the packet scheduler tries to serve the MTCHs according
to the priorities dynamically allocated to them in the particular TTI. The
higher priority MTCH queues will be served ahead of the lower priority
MTCH queues. For those MTCH queues having the same priority class, the
queue with the longest packet queue will be served rst.
For each MTCH l, mapped on FACH j and on S-CCPCH i, the packet
scheduler scans the TFCS of the physical channel to nd all the dierent
TBS sizes that could be used. A sorted list of all candidate TBS sizes, in
decreasing order, is created.
The scheduler rst seeks to allocate the maximum TBS size to the rst
FACH. This is the case when the sum of data at the MTCHs queues is
greater than the maximum supported TBS size for this FACH in the
TFCS; the allocation of data (transport block) that each MTCH can
transmit is based on the priority of each MTCH mapped to this FACH,
with the highest priority channel assumed to be given the maximum
share.
Otherwise, if the sum of data from all the MTCHs queues is less than
the maximum supported TBS size for this FACH, the selected TBS
size is the minimum available in the TFCS that can serve this sum of
queued data.
For each S-CCPCH, the packet scheduler checks the power required on
the basis of the BLER requirement of the active service ow. These power
allocation decisions involve the search in lookup tables (BLER versus
Eb /Nt ) to determine the transmitted power for each S-CCPCH, satisfying
both power and load constraints.
The packet scheduler will then derive a reduced TFCS out of the initial
one for the S-CCPCH i, including only those TFCs that feature the selected
TBS size for FACH j. Further allocations in the same TTI for another
MTCH/FACH mapped on the same S-CCPCH will have to consider this
reduced TFCS. As for the power allocation, the power required to satisfy
the active service ow with the most demanding target BLER is selected, as
long as the total transmit power per beam is not exceeded; otherwise, this
service is not scheduled.
These procedures are repeated recursively until all the FACHs mapped to
each S-CCPCH are assigned.
Performance evaluation
In order to demonstrate the performance of the packet scheduling schemes

proposed for broadcast and multicast services over S-UMTS, simulations have
been carried out for a wide range of scenarios by using a simulator devel-
oped under the ns-2 environment. Specically, the DDQ packet scheduling
algorithm has been evaluated via simulations in a typical S-MBMS scenario,
and compared with the Multi-Level Priority Queuing (MLPQ) scheduling
scheme described in [42]. The main characteristics of MLPQ are that it always
processes packets starting from those non-empty queues having the highest
priority rst, with queues having the same priority served in a round-robin
fashion. As a result, packets in the lower-priority queues may suer from
a considerably longer queuing delay. Moreover, according to this scheduling
policy, there is no dierentiation made between queues with the same QoS
ranking. Therefore, this is not an ecient mechanism in dierentiated QoS
multimedia services provisioning with respect to both eciency and fairness.
Rather than prioritizing queues in a strict method, other essential QoS metrics
should also be considered in the scheduling discipline design.
The following typical scenario with 3 S-CCPCHs each of 384 kbit/s has
been simulated:
S-CCPCH 1: 64 kbit/s download (FACH 1), 256 kbit/s streaming (FACH

2), 64 kbit/s streaming (FACH 3);
S-CCPCH 2: 256 kbit/s streaming (FACH 4), 128 kbit/s streaming (FACH
5);
S-CCPCH 3: 384 kbit/s download (FACH 6).
The above scenario is summarized in Table 5.4.
S-CCPCH 1 2 3
Bit-rate [kbit/s] 384 384 384
Streaming [kbit/s] 2561; 641 2561; 1281 -
Download [kbit/s] 641 - 3841
Table 5.4: Simulation multiplexing scenario (FACHs to S-CCPCHs).
Here we assume one-to-one mapping between MTCHs to FACHs, while

multiplexing only occurs from transport channel to physical channel. There-
fore, FACHs transport channel to physical channel multiplexing scenario is
specied in the simulation as in Table 5.4.
DDQ and MLPQ performance results are compared via simulation metrics,
such as mean delay, mean jitter and channel utilization.
Analysis of delay and delay variation
As illustrated in Figure 5.16, by using the DDQ packet scheduling algorithm,

the download multimedia services (i.e., FACH 1 and FACH 6) experience
much less mean delay compared with MLPQ. It is noted that the signicant
reduction in delay of lower-class services does not result in a dramatic
performance degradation for the higher-class counterparts (i.e., FACH 2 to
FACH 5). These results demonstrate that DDQ provides the download service
the highest possible degree of utilizing those spare resources remaining after
Fig. 5.16: Mean packet delay at RLC buers for dierent packet scheduling
algorithms.
streaming applications have been serviced, given that the detrimental aect
is not posing signicant degradation on the QoS target of streaming users.
Figure 5.17 shows the mean jitter experienced by each individual service
when employing MLPQ and DDQ packet scheduling. Obviously, DDQ features
much lower jitter for both streaming service and download service than
MLPQ, especially for lower-class and lower date rate users. Since the uni-
directional streaming service in S-MBMS is quite sensitive to delay-variation
(jitter), this result proves that DDQ packet scheduling provides a way to
balance all FACH queues in order to get the minimum delay variation for
streaming services.
Analysis of channel utilization ratio
Figure 5.18 shows the average S-CCPCH physical channel utilization for both
MLPQ and DDQ. Both schedulers managed to achieve throughput values
close to the optimum. For instance, the S-CCPCH channel utilization ratios
are 97.8%, 96.2%, 85.4% respectively under MPLQ scheduling; whilst they
achieve 98.4%, 96.2%, 86.4% respectively under DDQ scheduling. Therefore,
DDQ manages to obtain a slight channel utilization improvement on those
S-CCPCHs carrying background trac.
To summarize, the DDQ algorithm achieves the following advantages over
the MPLQ scheduling scheme:
Dynamic proportional delay-driven prioritization;
Fig. 5.17: Mean packet jitter at RLC buers for dierent packet scheduling
algorithms.
Fig. 5.18: S-CCPCH utilization for MLPQ and DDQ.

Highest utilization for the background class without posing signicant

degradation on streaming class;
Signicant improvement on mean delay and mean jitter performance;
Better overall system utilization.
5.3.4 Packet scheduling with cross-layer approach
Due to the nature of wireless transmissions, satellite communications suer

from strong variations of the received signal power due to shadowing and
multipath fading. Shadowing of the satellite signal is due to obstacles in the
propagation path (buildings, trees, bridges, etc). Whereas multipath fading
occurs because the satellite signal is received not only via the direct path,
but also being reected from objects in the surrounding area. Due to dierent
propagation distances, the multipath signals may add destructively and leads
to deep fades.
Due to these variations, the most critical part in satellite communications
is the communication link between the satellite and the user terminal (i.e.,
downlink). The downlink availability could be the limiting factor for the
performance of the overall system. Thus, a scheduler employing an explicit
cross-layer technique is proposed, where signaling interactions from the phys-
ical layer are employed so that the scheduler is aware of the channel state of
the users. In this cross-layer design, a multicast packet scheduler is developed
that relies on the prediction of the wireless channel conditions to improve the
performance of downlink transmissions via satellite for a TDMA-based air
interface.
The domain architecture for the multicast service under consideration is
illustrated in Figure 5.19. The entities in the service provider will provide the
interface between RAN and external packet data networks. The scheduler is
at the Earth station and a GEO satellite relays the multicast information
to all users through Multicast Terminals (MTs) and Terminal Equipment
(TE). A reliable multicast transport protocol is assumed to guarantee delivery
and congestion control mechanism. A unicast return link will be required for
acknowledgments.
Based on a TDMA framework, the system under consideration supports
scheduled access on both downlink and uplink. Downlink capacity is organized
into xed 80 ms MF-TDMA frames that are composed of a sequence of
xed 20 ms time slots. Channel State Information (CSI) of each user, which
is the information from physical (PHY) layer, is considered in the decision
mechanism whether to transmit or not the next multicast packet. The CSI
parameter is averaged over a total of N frames to become a conditional
parameter for the next transmission. CSI parameter is updated periodically.
The update through uplink bearer is contained within a 200 kHz sub-band,
which is further divided in frequency, and time slots and each slot may contain
a burst from a user.
Fig. 5.19: Domain architecture for a TDMA-based satellite system.
It is assumed that the CSI update provides a reasonably accurate prediction

of the slowly varying elements (i.e., valid at least within one round-trip-delay)
of the channel conditions. The satellite link is modeled using the Lutzs
two-state model with variation of fade duration for open and shadowed en-
vironments [45]. The following results indicate that a signicant performance
improvement is possible by adopting a cross-layer design approach in a fading
environment.
Description of schedulers task
Prior to assigning the slot, which is the resource allocation step, the dierent
multicast services need to be prioritized. In our scheme, the prioritization is
performed at two levels. The rst prioritization is static: the scheduler orders
the services according to their QoS classes (streaming and best eort), i.e.,
streaming service is assigned higher priority than best eort. The second level
of prioritization is based upon the cross-layer information provided with CSI
for services featuring the same level of priority. This prioritization is more
dynamic and conned only for best-eort trac. The algorithm is described
as follows:
For all incoming multicast packets, the packet scheduler aims to serve the
packets according to priorities dynamically allocated to them. Streaming
trac packets have higher priority to access to time slots at all times.
For the remaining slots, if best eort trac packets arrive, the scheduler
scans the CSI intended for the multicast group. The acquisition of CSI will
be performed for each user in the intended multicast group. The update
of channel condition is acquired in every 20 ms, according to the slot and
burst denition. The channel state information is contained in the bearer

control signaling data unit. In the evaluation of schedulers performance,
we consider TDMA channels in which each frame is divided into xed
control and data sub-frames. The user channel information is updated
through bearer control signaling data unit. The data sub-frame length is
the block length which is a data transmission unit size of 125 bytes and it
ts into time slots each holding one packet.
The packet scheduler will check for estimated Eb /N0 values of each user i in
the multicast group, i . A reference Eb /N0 threshold, T , is compared with
i and the number of users satisfying this reference will be the decision
making parameter for slot allocation.
If within one slot, more than one packets of the same priority arrives, then
the scheduler will check for the packet with higher percentage of satised
users. If the number of satised users from a particular multicast group is
above a certain threshold, the slot is allocated to that packet. If not, then
the packet will be delayed and retried for the next slot, provided that the
next slot is not intended for higher priority trac.
For more details, the interested reader is referred to [46].
Performance evaluations
The following study assumes stationary users and slowly-varying channels in
satellite links where fade duration holds within one CSI update. The results are
here based on perfect channel predictions; we assume no channel estimation
loss occurs. This assumption might be impractical in a satellite environment
where the propagation delay is high, but the results with this assumption
permit to have a good indication of the eectiveness of this scheme to achieve
a high reliability multicast transmission.
In this study, two dierent scenarios have been examined. The rst one,
the single environment scenario, assumes that all users are subject to identical
channel conditions (the single environment model uses an elevation angle
= 80 and values for and calculated for urban areas with K factor of
7, where K represents the Ricean factor which is dened as the ratio of the
dominant component to the scatter contribution [45]). The proposed technique
aims at reducing the number of retransmissions that stem from bad channel
conditions. Figure 5.20 [46] exhibits some rather interesting results, where
parameter zeta is dened as follows: a packet is retransmitted only if the
percentage of users in the multicast group that experience Packet Loss Rate
(PLR) higher than a dened PLR threshold is greater than a percentage,
denoted by parameter zeta. It should be pointed out that PLR signicantly
diminishes, by endowing the multicast packet scheduler with CSI (cross-layer
approach). Moreover, PLR is hardly aected by an increase in the multicast
group size, whereas the greater the parameter zeta is, the lower PLR is.
Figure 5.21 [46] illustrates the probability that at least one user will
request retransmission versus the multicast group size. Apparently, as the
Fig. 5.20: Packet loss rate versus multicast group size. See reference [46]. Copyright
2005
c IEEE.
Fig. 5.21: Probability of at least one user in a multicast group requesting

retransmission (failure rate) versus multicast group size. See reference [46].
Copyright 2005
c IEEE.
size of the multicast group increases, so does the probability of retransmission.

Furthermore, this probability can be reduced by increasing parameter zeta.
What is more important is that the greater the size of the multicast group,
the higher the average packet delay, as illustrated in Figure 5.22. The value
of zeta used to obtain the results in Figure 5.22 is 0.9.
Fig. 5.22: Average packet delay versus target number of users (normalized) in good
channel condition.
At this point, multi-channel environments are simulated based on the

parameters in Table 5.5.
Area Rician K factor ( ) % Users

Suburban 0 dB 20 1.69 2.70 20
Urban 3 dB 20 13.90 3.06 40
Urban 7 dB 80 1.75 0.80 30
Suburban 10 dB 60 0.14 0.40 10
Table 5.5: Simulation parameters for the multi-environment scenario.
In this case, multicast users are subject to dierent channel conditions,

and the empirical models presented in [45] were deployed. A user requires
retransmission only if the dierence between ref , which is the reference
Eb /N0 from the AWGN channel model to achieve a target PLR of 102 ,
and (t), which is the Eb /N0 value of the signal received from this user, is
greater than a given Eb /N0 threshold, T :
ref (t) > T [dB]. (5.13)
Figure 5.23 [46] depicts the probability that at least one user will re-
quest retransmission versus Eb /N0 threshold. Evidently, the retransmission
probability decreases as Eb /N0 threshold increases. It should also be noted
that the retransmission probability decreases as parameter zeta increases. As
far as the average packet delay is concerned, it becomes clear from Figure
5.24 that as Eb /N0 threshold increases, the mean delay increases since the
multicast packet scheduler refrains from transmitting packets to multicast
groups typically experiencing bad channel conditions [46].
This approach has been shown to reduce unnecessary transmission of
best-eort trac and hence reduces unnecessary bandwidth usage and retrans-
mission requests. However, in achieving relatively good channel utilization for
a multicast group, higher average packet delay is expected in the cross-layer
scheduler. The average packet delay can be regulated according to the power
threshold a user is estimated to receive at the downlink transmission. It is
also important to note that this approach consumes an amount of resources
in performing the channel prediction algorithm. The accuracy of the channel
quality is highly dependent on the channel model used.
Fig. 5.23: Probability of at least one user in a multicast group requesting

retransmission (failure rate) versus Eb /N0 threshold. See reference [46]. Copyright
2005
c IEEE.
Fig. 5.24: Average packet delay versus Eb /N0 threshold. See reference [46].
Copyright 2005
c IEEE.
5.4 Conclusions
Satellite communications have a potential market in providing high downlink
bit-rate services and in supporting multicast services on broad areas of the
Earth. These are the reasons why this Chapter has focused on HSDPA
and MBMS provision via a GEO bent-pipe satellite. In both cases suitable
network architectures and radio resource management techniques have been
investigated to support such services in an appropriate and ecient way.
For HSDPA, the results showed by the Proportional Fair scheduler are
sub-optimal for mixed trac classes, since it does not provide any QoS
dierentiation among diverse applications. The study of proposed enhance-
ments to the PF scheduler to support QoS dierentiation, such as the
Exponential Rule, should be addressed in the future for the satellite case.
Furthermore, the impact of the round trip time in the acquisition of channel
state information has been shown in the form of packet losses in intervals
of misalignments between current channel state and information available at
the Gateway. In particular, the simulation results in a simplied scenario
(using a GOOD/BAD channel model) show non-negligible losses due to the
use of outdated information in the selection of the best suited TFRC for
transmission. Hence, if it is desired to reduce the number of retransmissions,
delay compensation strategies or larger margins in the selection of TFRCs
should be adopted. Furthermore, a more complex channel model should be
considered in order to take into account the channel variation dynamics typical
of S Band (S-UMTS band).
For the provision of broadcast and multicast services, it has been shown
that packet scheduling is an important element within the RRM framework.
Aiming at a more ecient provision of heterogeneous QoS-dierentiated
MBMS services over S-UMTS, novel packet scheduling algorithms have been
proposed. These algorithms take into account the impact of important per-
formance factors reecting service QoS demands in order to provide trac
dierentiation and overall system performance optimization. To tackle the
deteriorating eect of changing propagation environments in multicast trans-
missions, channel estimation can ll the void whilst obtaining the current
channel state. Statistical channel models can be used to represent channel
variations to be exploited by packet scheduler for its decisions. For trac
with strict delay bound, a negotiation between delay and channel states can be
facilitated by a cost function where a trade-o between delay and throughput
is expected.
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6
CALL ADMISSION CONTROL
Editors: Stylianos Karapantazis1 , Petia Todorova2
Contributors: Stylianos Karapantazis1 , Petia Todorova2 , Franco Davoli3 ,

Erina Ferro4
1
2
3
4
6.1 Introduction to Call Admission Control
RRM in multimedia satellite networks aims to guarantee the fair distribution

of available resources, due to the fact that the total link capacity has to be
divided among several users, as well as to fulll certain pre-negotiated QoS
requirements for the lifetime of the connection. RRM is one of the functions
that are carried out in the Data Link Layer (DLL). A general DLL protocol
stack that applies to satellite networks is depicted in Figure 6.1, while Figure
6.2 illustrates the most important RRM entities.
One of the most important resource management functions is Call Ad-
mission Control (CAC), which comprises the set of functions taken by the
satellite network during the phase of connection establishment or connection
re-negotiation to decide whether to accept or reject a users request for
a connection. A new users request can be accepted provided that there
178 Stylianos Karapantazis, Petia Todorova
are adequate network resources available to guarantee the QoS of both all
already-existing connections and the new requested one. Generally, the CAC
function results in the blocking of new calls or call dropping in the case of
ongoing calls when the bandwidth required for the connection exceeds the
available bandwidth. CAC, which turns out to be a crucial function to provide
high utilization of network resources, is network-specic and is generally
managed by the Network Control Center (NCC - recall that a description
of the NCC functions is given in Chapter 1, sub-Section 1.4.3). However, in
non-GEO satellite systems the CAC function has to be implemented on board
of the satellite as well. Nevertheless, it should be mentioned that this approach
requires satellites with on-board processing capabilities.
Fig. 6.1: A general protocol stack for the main elements of a satellite network.
Fig. 6.2: The main RRM entities.

Chapter 6: CALL ADMISSION CONTROL 179
6.2 CAC and QoS management
As noted in [1], the public data network provides a resource that could
profoundly impact on high-priority activities of society, like defense and
disaster recovery operations. Under stress, however, the public network turns
out to be a virtually unusable resource, unless suitable trac prioritization
and CAC are applied to improve its performance. CAC has been extensively
studied in the past as a general resource allocation mechanism in various
networking contexts. Ross [2] is an excellent reference for CAC mechanisms
in general, whereas reference [3] contains a recent survey on this topic in the
context of wireless networks.
In the simplest case of resource allocation, a connection is admitted simply
if resources are available at the time the connection is requested. This policy
is commonly called Complete Sharing (CS), where the only constraint on the
system is the overall system capacity. In a CS policy, connections that request
fewer resource units are more likely to be admitted (e.g., a voice connection
will more likely be admitted compared to a video connection). A CS policy
does not consider the importance of a connection when resources are allocated.
At the other extreme, in a Complete Partitioning (CP) policy, every trac
class is allocated a set of resources that can only be used by that specic
class. Other solutions are represented by Trunk Reservation (TR), where class
i may use resources in a network as long as ri units remain available [4],
and Guaranteed Minimum (GM) [5],[6], which gives each class its own small
portion of resources; once used up, classes can then attempt to use resources
from a shared pool. An Upper Limit (UL) policy was adopted in [1], and
Virtual Partitioning (VP) was proposed in [7].
As far as satellite systems are concerned, the architecture of the new
satellite systems testies the interest in ATM, IP and DVB technologies. A
general architecture of a satellite system is illustrated in Figure 6.3. An Earth
station (Gateway) is in charge of mapping ATM/IP trac originated from
terrestrial terminals over satellite connections, while the NCC performs CAC
and DBA functions. The role of the aforementioned functions is to meet the
QoS requirements of dierent service classes, i.e., delay, jitter and packet loss.
A plethora of CAC algorithms were proposed in the literature for terrestrial
ATM-based networks. Some of them require an explicit trac model, while
some others require trac parameters such as peak and average rate. A
classication of these schemes is provided in [8] along with the description
of their salient features. Nevertheless, it should be noted that while some
parameters can be easily specied (for instance, the peak rate), the actual
average rate is dicult to estimate, since the source does not know it.
Then, the user can declare an upper bound, which, however, results in
low bandwidth eciency. To cope with this issue, measurement-based CAC
methods have been proposed. In [9], the authors present a taxonomy as well
as a detailed survey of measurement-based CAC techniques. In that study,
dierent measurement-based CAC methods were compared against each other
Fig. 6.3: General architecture of a satellite system.
in the light of bandwidth eciency, Cell Loss Ratio (CLR), implementation

complexity, scalability and dependency on trac model. The authors were led
to the conclusion that those methods that are based on eective bandwidth
are the most suitable for high-speed communication systems, since they are
simple enough to be implemented in real systems, they attain high bandwidth
eciency and last, but not least, they assume fewer trac parameters. The
rationale behind this category of CAC schemes is rather simple. First, the
eective bandwidth for the aggregate connections is measured, namely the
equivalent bandwidth needs of ongoing connections. Then, a request for a new
connection is accepted provided that the requested bandwidth is smaller than
the residual bandwidth, that is, the total link bandwidth minus the eective
bandwidth.
Concerning ATM-based satellite networks, they are able to meet dierent
QoS requirements at the ATM layer [10]. These requirements are dened
in terms of objective values of the network performance parameters, as
specied in ITU-R Recommendation S.1420 [11]. Some of the QoS parameters
(Peak-to-Peak Cell Delay Variation, Max Cell Transfer Delay and Cell Loss
Ratio) may be oered on a per-call/connection basis and negotiated between
the end-system and the network, whereas some other QoS parameters (Cell
Error Ratio, Severely Errored Cell Block Ratio and Cell Misinsertion Rate)
cannot be negotiated. For each direction of the call/connection, a specic QoS
is negotiated, based on a trac contract between the network and the user.
At call set-up time, the user declares the source trac descriptors and the
QoS class by means of signaling or subscription. The trac descriptors in the

set-up signaling message include a generic list of trac parameters, specic for
each user connection. For each connection request, the CAC function derives
the following information:
The source trac descriptors, including the trac characteristics of the

ATM source;
The Cell Delay Variation Tolerance (CDVT) value;
The requested and acceptable values of each QoS parameter, and the QoS
class.
In particular, the idea of endowing LEO satellites with on-board ATM

switching capabilities (Figure 6.4) combines the advantages of LEO systems,
like signicantly reduced propagation delay, rendering them suitable for
real-time applications, with those oered by ATM, including faster trans-
mission rate, bandwidth on demand, compatibility with existing protocols
and guaranteed QoS [12],[13]. By supporting statistical multiplexing, priority
queuing and multicasting, ATM technology can accommodate all QoS features
requested by the user and therefore, becomes a suitable solution for broad-
band multimedia communications. However, as LEO satellites coverage area
changes continuously over time, in order to maintain connectivity, end-users
must switch from beam to beam and from satellite to satellite, resulting in
frequent intra- and inter-satellite handovers.
Fig. 6.4: An on-board ATM switching/processing architecture. See reference [12].

Copyright 2003
c IEEE.
The functions of the individual modules in Figure 6.4 are as follows:

Switch Fabric: switching cells from input ports to appropriate output
ports.
Input Processor : scheduling, buer monitoring.
Output Processor : scheduling, buer monitoring and cell discarding.
Control Module: CAC, handover monitor & control, resource allocation,
routing table update, signaling protocol, etc.
The ATM switch uses dierent input/output ports for the uplink/downlink
and for the Inter-Satellite Links (ISLs). This is because of the dierent
bandwidth and signaling protocols used.
The functions of the Control Module (CM) are shown in Figure 6.5,
assuming that signaling and routing table updating are implemented [13].
For intra-satellite handover, the Handover Monitor & Control module has
to monitor and measure the handover status of all beams belonging to the
satellite.
Fig. 6.5: The anatomy of an ATM switch with CAC/handover control module. See
reference [13]. Copyright 2004
c IEEE.
It is assumed that the mobile user initiates the intra-satellite handover

process based on physical link quality measurements. Then, the mobile user
will send a handover request message to the LEO satellite, indicating the
new beam identication and the QoS requirements. The satellite CM has
to implement the handover/CAC process in order to decide whether or
not the new beam could provide the QoS requirements. If the handover is
accepted, a handover reply message is sent to the mobile user. At this point
buering processes are needed to guarantee the minimum loss of cells. After
the uplink/downlink accesses are nished, buered cells will be transmitted
through new links.
As the Internet has become a ubiquitous communication infrastructure, IP
QoS provisioning is a strategic issue in any kind of network. A view that has
been gaining considerable interest in the scientic community considers that
the IP Integrated Services approach (IntServ - service dierentiation is focused
on individual packet ows) can be used in the wireless access networks (in
our case the satellite links) in order to admit or to reject the requests of ows
according to the availability of resources and the guarantees provided to other
ows. On the other hand, the IP Dierentiated Services approach (DiServ
- scalable service dierentiation, focused on the aggregate of ows) can be
employed to avoid complexity and maintenance of per-ow state information
in the core network [14],[15]. Both IntServ [15],[16] and DiServ [17] have
been studied for satellite networks; they are considered later in conjunction
with CAC schemes.
In general, CAC schemes can be classied into those that oer Determin-
istic QoS Guarantees and those that provide Statistical QoS Guarantees [8].
Deterministic QoS guarantees: a new connection is accepted, provided that
the worst-case scenarios requirements are met (for instance, the available
capacity is greater than the peak rate of the connection). Although this
approach represents the simplest solution for trac management, it tends
to over-commit resources, thus resulting in low link utilization.
Statistical QoS guarantees: in this case, the NCC maintains a statistical
allocation instead of guaranteeing a peak rate. Losses may occur, but
high channel utilization is accomplished. This approach is based on the
assumption that having all the connections transmitting at their peak
rates at the same time is beyond the realms of possibility, allowing in this
way the statistical multiplexing of ows. However, the diculty of this
approach lies in the trac characterization problem.
An ecient integration of the aforementioned approaches can make up for
the weaknesses of each other [18]. In particular, the technique proposed in
that study combines the good characteristics of the EDF (Earliest Deadline
First) scheduler in terms of QoS provisioning with the advantages that stem
from statistical multiplexing (a description of the EDF scheduler is provided
in sub-Section 5.3.1). Moreover, the mask of the Dual Leaky Bucket used for
trac shaping is such that takes into account the statistical variability of the
peak rate and the burst size. Note that a leaky bucket is simply a nite queue
and it can be viewed as a bucket with a small hole in the bottom: no matter
at what rate water enters the bucket, the outow is at a constant rate, when
there is any water in the bucket. In other words, a leaky bucket is used to
smooth out bursts and greatly reduces the chances of congestion. Simulation
results showed that the scheme described in [18] improves channel utilization,
while providing QoS guarantees at the same time.

Another important issue closely related to CAC and QoS is represented by
fade countermeasures for rain-fading attenuation. Forward Error Correction
(FEC) techniques aim to mitigate channel impairments and diminish BER.
Notwithstanding the advantages that accrue from FEC schemes, the price
to pay is a decrease in the information bit-rate. Specically, under the
condition of xed bandwidth availability, these schemes have an impact on
the operations of at least two higher layers. These operations are:
The CAC and bandwidth allocation for guaranteed-bandwidth trac
(indirectly aecting, in turn, the residual bandwidth left to best-eort
trac);
The performance of the TCP congestion control mechanism.
The impact of fade countermeasures on TCP will be investigated in
Chapter 8. Concerning CAC, adaptive control approaches can be adopted in
the context of Constant Bit Rate (CBR) and Variable Bit Rate (VBR) connec-
tions, also taking into account the presence of best-eort trac [19],[20],[21].
In the presence of guaranteed-bandwidth trac, control actions may take the
form of CAC and bandwidth allocation, whose parameters can be determined
and, possibly, adaptively tuned on the basis of the fade countermeasures
adopted; the ensuing redundancy is applied to ongoing and incoming con-
nections. In particular, the decision on whether to accept or reject a request
for a new call is dependent on the measured level of signal attenuation.
Moreover, the trac source rate and FEC rate can be dynamically adjusted
in a co-ordinated fashion to satisfy QoS requirements.
6.3 CAC algorithms for GEO satellite systems

GEO satellite systems have been dominant in the telecommunications arena
for years and have been the subject of extensive research by virtue of their
large coverage area and their intrinsic broadcast/multicast capabilities. The
frequency band allocated for satellite services has changed many times over the
years. Several proposals for GEO satellite systems suggest the use of Ka band
(20-30 GHz). Systems operating at these high frequencies can provide a wide
spectrum of multimedia applications to users. Thus, CAC becomes an issue of
paramount importance to provide QoS guarantees to calls of dierent service
classes. The Multi Frequency-Time Division Multiple Access (MF-TDMA) air
interface solution has been adopted by most of the satellite system designers.
In the following, a description of CAC algorithms for MF-TDMA GEO
satellite systems is given.
6.3.1 CAC schemes for MF-TDMA networks

The study in [22] focuses on resource allocation and CAC in broadband GEO
satellite systems. In particular, a GEO satellite with on-board processing
capabilities is considered, thus allowing the CAC decision to be taken on board

of the satellite. The proposed algorithm, which is called Dynamic Movable
Boundary Strategy, is geared towards the specic needs of CBR and bursty
data trac and guarantees a minimum number of resources to each service
class. In brief, users access the channel in a Time Division Multiple Access
(TDMA) manner. The frame is divided into three parts: one part dedicated
to CBR calls, another one devoted to bursty data trac, and a third part
used as a common resource pool. The CAC decision for CBR trac, as well
as the resource allocation decision, is taken periodically, at the beginning
of a time interval called control period. This algorithm adapts itself to the
network loading conditions by modifying the resource allocation criterion at
the beginning of each frame.
A CAC scheme for integrated ATM-satellite systems is proposed in [10].
The proposed algorithm caters for both real-time and non-real-time variable
bit-rate trac by exploiting the statistical multiplexing of trac sources. In
particular, the supported trac is categorized into four classes. Resources
are allocated on a permanent basis for calls of the rst trac class, namely
CBR trac sources, while a semi-permanent allocation based on the statistical
multiplexing of trac sources is employed for calls of the three other trac
classes, that is, for real-time-VBR (rt-VBR), non-real-time-VBR (nrt-VBR)
and Unspecied Bit Rate (UBR) trac. The idea of the algorithm consists
in the introduction of the excess demand probability, which is the probability
that a given number of calls request in a future time more channels than
those actually available. A double check is performed before admitting a new
call into the system, ensuring that the excess demand probability is below a
predened threshold for each trac class. Specically, the rst check ensures
that the excess demand probability of all the multiplexed sources is below
1 , whereas the second check veries that the excess demand probability of
real-time trac (CBR and rt-VBR) is also lower than 2 , where 2 < 1 since
real-time trac is characterized by stringent QoS constraints.
A similar CAC technique is combined in [23] with an in-band signaling
scheme in order to combat the adverse eects of the intrinsic propagation
delay, which makes the trac prole dierent from the one declared. The
in-band technique allows requesting resources for semi-permanent connections
on a burst basis. In particular, it is adopted by VBR sources and allows
the use of a eld in the currently transmitted burst in order to notify a
new burst arrival, thus obviating the need for signaling exchange between
ground and space segments. It should be noted, that in that study only
rt-VBR trac was considered. Such study was extended in [24], where the
in-band-signaling scheme was coupled with a resource engagement prolonga-
tion technique. When the former is used in conjunction with the latter, the
time needed for resource allocation notication is reduced. In brief, if the
trac resource management scheme nds out, when processing a bandwidth
request, that resources are still occupied and used by the relevant terminal
for the transmission of previous information bursts, then it just lengthens the
time interval that those resources remain engaged, thereby diminishing the
number of bursts that are lost while waiting for the acknowledgement of the
assignment of new resources.
A CAC scheme for DVB-RCS systems is examined in [25]. The scheme
presented in that study is coupled with a capacity request scheduling technique
with the aim of meeting the QoS requirements of dierent service classes. In
particular, the CAC algorithm employs a preventive congestion control, based
on trac descriptor parameters (that is, peak bit-rate, burstiness, and service
category) and decides whether to accept or to reject a new call connection
according to the estimation of the excess demand probability. The latter sets
an upper bound on the burst loss probability.
The concept of excess demand probability is also used by the CAC scheme
presented in [26], where an integrated terrestrial-satellite system is considered.
The CAC scheme consists of two distinct phases: terrestrial admission control
and satellite admission control. The authors of that study also propose the
use of the IP IntServ architecture in the satellite network and the adoption of
a scalable IP DiServ-like architecture in the terrestrial network. Concerning
the satellite admission control, it accepts a given number of calls if the excess
demand probability is such that a target service quality can be guaranteed.
A CAC scheme geared towards multimedia GEO satellite networks with
on-board cross-connectivity, that is, connectivity between any pair of beams,
is presented in [27]. It is considered that there exists one Gateway Earth
Station associated with each beam. In addition to this, it is assumed that
any connection initiated by a user ends in the terrestrial network. Assuming
that the QoS requirements of a connection can be met in the home Gateway,
then the CAC criterion consists in opting for the destination Gateway that:
(i ) has enough bandwidth to support a connection request, and (ii ) results in
the shortest distance to the connections terrestrial destination. The amount
of resources statically allocated depends on the connection type (i.e., the
ATM-based classication of services), the trac descriptors, and the requested
QoS.
In [28], the employment of the IntServ model in a GEO satellite system
is examined. Specically, the authors of that paper study two main classes of
service, namely Guaranteed Services and Controlled Load Services. The former
is suited for real-time applications with stringent QoS requirements, whereas
the latter provides for adaptive-tolerant real-time trac (i.e., trac with loose
delay requirements). The satellite CAC supports the statistical multiplexing of
trac over the air interface. A new call is accepted if the network has sucient
bandwidth to satisfy the QoS constraints of the call without degrading the
QoS perceived by ongoing calls. Specically, the authors of that study adopt
a technique similar to the one described in [16]. Each ow is characterized by
specic parameters that are called token bucket parameters. These parameters
are the token bucket rate r, the token bucket size b, the peak data rate p and
the maximum packet size M. However, what is meant by token bucket?
Token bucket is an algorithm for trac shaping, like the leaky bucket algo-
rithm, used to regulate the average rate (and burstiness) of data transmission.
It simply counts tokens. However, in contrast to the leaky bucket algorithm,
which does not allow idle terminals to save up permissions to send large
bursts later, the token bucket algorithm does allow saving, thus permitting
some burstiness in the output stream and giving faster response to sudden
input bursts. In brief, a counter is increased by one (or a token is added in
the bucket) every 1/r seconds and decreased by one whenever a packet is
sent. When the counter hits zero, no packets can be sent. The token bucket
algorithm allows up to b tokens to be added in the bucket. All the token
bucket parameters are used by the CAC algorithm in order to estimate the
resources that are required for each new ow. Specically, the source terminal
sends a request for a new connection towards the destination. This request
serves the purpose of describing the characteristics of the ow in terms of
token bucket parameters. Each router (or, in general, each network element)
that receives this request computes how it will handle packets of this ow and
updates the request by adding this information to it. When the destination
receives the request, it can calculate the bandwidth that is required so that
the maximum end-to-end delay be below a given threshold by combining the
information that each router has added to the request.
Concerning Guaranteed Services, the destination (i.e., an edge device
located at the border between terrestrial and satellite segments) computes
for each ow the bandwidth R and the buer space B on board the satellite
that are required so that the QoS constraints be met. Then, these quantities
are sent to a designated Earth station, which decides on whether to accept
or reject this new ow. As regards Controlled Load Services, a similar CAC
procedure is applied. Nonetheless, in this case, the resources that are requested
do not guarantee that specic target values in terms of end-to-end delay and
packet loss will be met.
The performance of a CAC algorithm that is combined with a variant of
the Resource Reservation Protocol (RSVP) is assessed in [29]. In that study,
the trac carried by the satellite network is categorized into threes classes,
that is, data trac, multimedia trac, and control trac. A pool of channels
is available for all classes. However, if all these channels are reserved, the
remaining channels can be used only for the transmission of data and control
trac.
A CAC technique for DVB-S/DVB-RCS satellite systems is examined
in [18]. In particular, the CAC algorithm that is presented capitalizes on
the positive characteristics of the EDF scheduler in order to provide QoS
guarantees and attain high channel utilization. The proposed technique is
compared with two CAC schemes that are based on the Deterministic QoS
guarantees and the Statistical QoS guarantees approaches.
The authors of [30] study a CAC algorithm for DVB-RCS satellite net-
works, which is tailored for Moving Picture Experts Group (MPEG) trac
sources. MPEG represents a video compression standard for multimedia appli-
cations. In essence, MPEG subdivides the video in Group of Pictures (GOPs).
The GOP rate changes over time, therefore the CAC scheme described in that
study relies on the statistical multiplexing of this kind of trac. Specically,
the authors propose a statistical multiplexing scheme that is based on discrete
bandwidth levels of the GOP rate and compare it to another scheme that
relies on the Normal distribution of the aggregate GOP rate [31]. Concerning
the latter scheme, the MPEG trac generated by each source is modeled as
a Normal distribution of GOPs with mean rate and standard deviation .
Thereby, supposing that MPEG ows are independent of each other, according
to the central limit theorem the aggregate trac of a set of N multiplexed
connections can also be modeled as a Normal distribution.
Albeit that scheme takes some characteristics of the MPEG trac into
account, it cannot account for trac variations over time. A solution based
on a GEO satellite system equipped with on-board processing and on-board
switching is investigated in [32], where an integrated CAC and Bandwidth on
Demand (BoD) algorithm is proposed for a broadband satellite communica-
tion system of this kind, loaded with heterogeneous trac. This algorithm
is able to utilize eciently available bandwidth in order to attain high
throughput and maintain a good grade of service for all the trac types.
Last but not least, an issue of great importance for the designers of
satellite systems is the energy allocation. Power is a resource at a premium in
satellite systems, therefore a trade-o between consuming and saving energy
is always sought. At this point, it should be noted that higher levels of energy
consumption translate into higher throughput. Reference [33] derives an
optimal threshold policy for the joint problem of CAC and energy allocation,
by means of a dynamic programming approach. In particular, as usual in
dynamic programming, a value function Jk (ak , rk , dk ) is introduced which
aims to show how desirable is a satellite with available energy level ak at time
k, given that the current demand is dk and the current reward is rk . The term
rk represents the reward for consumption, namely the satellite receives rk
units of reward per unit of energy consumed. This amount of reward depends
on distances, atmospheric conditions and nancial considerations. The aim is,
then, to maximize the value function over a consumed energy ck .
6.3.2 CAC schemes for CDMA networks
Albeit MF-TDMA has been shown to be particularly eective in satellite

networks, Code Division Multiple Access (CDMA) has emerged as the main-
stream air interface solution for the 3rd Generation (3G) networks. One
scenario that holds considerable appeal involves the integration of Satellite
Universal Mobile Telecommunications System (S-UMTS) with Terrestrial
UMTS networks (T-UMTS) [34], thus resulting in a powerful integrated net-
work infrastructure. However, unlike in TDMA/FDMA networks, in CDMA
systems users share the same portion of bandwidth at the same time. This
is realized by assigning each user a pseudo-random code. A new user can be
admitted to the network as long as the Signal-to-Interference Ratio (SIR) is
adequate for processing at the receiver and the QoS requirements of ongoing
calls are met. Thereby, CDMA systems are interference-limited rather than
capacity-limited. Despite the vast literature on CAC algorithms for terrestrial
CDMA networks, only a handful of studies exists on CAC schemes for satellite
CDMA systems.
An interactive SIR-based algorithm for S-UMTS networks is delineated in
[35]. The described algorithm aims at nding out if a power equilibrium point
can be calculated so that the target SIR of all the ongoing calls and the target
SIR of the new call are met. This CAC scheme is applied to the admission of
bi-directional, high-demanding services.
The authors of [36] propose a CAC scheme that provides QoS guarantees to
integrated voice, videoconference, and data services. The essence of their goals
is to maximize the utilization of system resources. The air interface adopted
in that work is a combined CDMA/TDMA scheme. The highest priority is
given to videoconference calls.
6.4 Handover and CAC algorithms for non-GEO

satellite systems
The mind-set of satellite systems designers over the past decades has been
to keep most of the complexity on the ground segment. Notwithstanding, the
advantages that stem from this design approach, the growing exigencies for
both mobility and ubiquitous access, coupled with advances in technology,
led the designers to move satellites closer to the Earth surface in order to
enable the provision of delay-sensitive and high bit-rate services. Non-GEO
satellite systems attracted considerable attention by virtue of some of the
compelling features that are endowed with, such as the low propagation delay
and the ability to communicate with handheld terminals. The 1990s were
perhaps the public heyday of this type of satellite systems. In that decade,
several commercial satellite constellation networks were come to light, while
the end of the decade saw the launch and the start of operations of two LEO
satellite constellations, namely Iridium and Globalstar, which provide voice
service and paging. Nevertheless, the widespread usage of terrestrial cellular
systems for the provision of mobile telephony worldwide had usurped many
of the target markets, thus these non-GEO satellite networks have never
come to fruition on account of their competitive rather than complementary
role with respect to terrestrial cellular systems.
The coverage area of non-GEO satellites, referred to as footprint, is divided
into slightly overlapping cells, called spot-beams. Due to the movement of
satellites with respect to the Earths surface, end-users must switch from
spot-beam to spot-beam and from satellite to satellite in order to maintain
connectivity. Thus, as in the case of terrestrial cellular systems, the issue of
call handover arises in non-GEO satellite constellations as well. Two types of
call handover can be distinguished:
Intra-satellite handover (also referred to as cell handover or spot-beam

handover ), which refers to the handover of a call between neighboring
cells (beams) of the same satellite (Figure 6.6).
Inter-satellite handover (also referred to as satellite handover ), which
relates to the handover of a call between two contiguous satellites (Figure
6.7).
Fig. 6.6: Intra-satellite handover.
Fig. 6.7: Inter-satellite handover.

It should be pointed out that, in contrast to terrestrial cellular networks,

where the handover rate is determined by the motion of the users, in non-GEO
satellite systems, the handover rate is determined by the motion of the
satellites. In the case of LEO satellites, the ground track speed of satellites
is over 5700 m/s (note that users in fast vehicles move with a velocity of 80
m/s at most), hence a satellite is in view for up to 10 min, while the users
sojourn time in a cell can be as short as 1 min.
The handover of a call constitutes a daunting challenge in this kind of
satellite systems, since it may result in the forced termination of ongoing calls.
To overcome this problem, advanced CAC and handover control techniques
are required for improving the QoS performance. While the role of CAC
algorithms is to decide whether to accept or reject a new call, handover
techniques aim to ensure that the service of a call will not be interrupted when
the user moves from one cell (or satellite) to another. In other words, while
the aim of CAC techniques is to minimize Call Blocking Probability (CBP),
handover techniques aim to diminish Call Dropping Probability (CDP), also
referred to as Forced Termination Probability. Unfortunately, any eorts to
reduce one of these two probabilities result in an increase in the other one.
6.4.1 Intra-satellite handover and CAC schemes

Several approaches for handover prioritization proposed for terrestrial cellular
systems have been studied for the case of intra-satellite handover in non-GEO
satellite systems. The techniques that can be found in the literature are based
on either Dynamic Channel Allocation schemes (that is, any channel can be
temporarily assigned to any cell) [37],[38] or Fixed Channel Allocation schemes
(that is, a set of channels is permanently assigned to each cell) [39]-[49]. CAC
and intra-satellite handover schemes are summarized below. The works are
referenced on a time-line basis, identifying the seminal works in this eld on
the one hand, and works that primarily extended previous studies on the other
hand.
In [39], a CAC strategy is proposed, along with an intra-satellite handover
scheme. A metric, called mobility reservation status, is introduced that aims to
provide the information about the bandwidth that is required by all the active
calls in each cell as well as to predict the potential bandwidth requirements
by calls in adjacent cells. Supposing that a new call k has been accepted in a
given cell (i.e., spot-beam) m, the mobility reservation status of this cell, as
well as the mobility reservation status of the next S - 1 cells, is increased by
the following quantity:

Bk T0 i=0
Tmax ,
Cm+i (k) = (6.1)
Bk Tmax
T0 +iTmax , i = 1, ... , S 1
where Bk is the number of the trac channels required by this cell, T0
is the users dwell time in the source cell m of the call, whereas Tmax is the
maximum time interval that a user can dwell in a generic transit cell. Note
that index i from 0 to S 1 is used to denote the source cell (index m + i,
i = 0) and the next possible transit cells (index m + i, i = 1, ... , S 1). A
new call is admitted into the network only if there are at least Bk available
channels in the cell m where the user is located, and at the same time the
values of the mobility reservation status of this cell, the previous cell and the
next one are below a predened threshold, called Tnew . As far as handover
requests are concerned, a call is successfully handed over to a new cell provided
that the number of available channels in that cell is greater than Bk and its
mobility reservation status is below a predetermined threshold, called THO .
Apparently, THO is greater than Tnew in order to prioritize handover requests
over new call requests.
In [40], an adaptive dynamic channel allocation scheme is examined,
which relies on the well-known concept of guard channels, which are channels
exclusively used in each cell only to serve handover requests. In particular, the
number of guard channels is dynamically adapted based on the estimation of
future handover events. In more detail, upon the arrival of a new call request in
a cell, the algorithm, by capitalizing upon the deterministic network topology
of LEO satellite systems, computes the users dwell time in that cell. Then
it estimates the number of the potential handover requests within this time
interval as well as the expected number of channels that will be needed
to serve these requests. The request will be accepted only if the number of
available channels is greater than . As regards handover requests, a call is
successfully handed over to a new cell as long as there is at least one available
channel in that cell.
The study in [41] extends the aforementioned scheme and proposes a
geographical connection admission control algorithm that aims to guarantee
that the forced termination probability will always be below a predened
threshold. This CAC algorithm is based on the estimation of the future CDP
of both the new calls and the ongoing ones. Upon the arrival of a new call,
these two probabilities are estimated, and the call is admitted into the network
provided that these probabilities are below some predened thresholds.
The techniques presented in [37],[38],[42]-[46] rely on the queuing of
handover requests. According to this kind of handover schemes, a handover
request is queued for a specic time interval when no channel is available in
the next cell. In [37],[38],[42], the queuing time interval is dependent on the
overlapping area between contiguous cells.
In [43], a guaranteed handover service scheme was proposed. According
to that technique, a handover request can be queued up to a time interval
equal to the users sojourn time in the cell, that is, as soon as a handover
occurs, a handover request is sent to the next transit cell. As far as new calls
are concerned, a new call is admitted into the network as long as there exists
an available channel in both the current cell and the rst transit cell. That
scheme attains zero CDP at the expense, however, of a rather high CBP.
The authors of [44] propose a handover technique similar to the guaranteed
handover service scheme, which aims at increasing channel utilization and,

thus, reducing CBP. Toward this end, the channels in that scheme are reserved
only for the time intervals they are expected to be in use, hence the name
Time-based channel reservation algorithm. Therefore, channel utilization is
improved and CBP is reduced.
In [45],[46], the queuing time interval is considered to be dependent on
the value of a parameter that was called handover threshold. This parameter
should be appropriately selected in order to attain a trade-o between drop-
ping and blocking probabilities as well as to achieve high channel utilization.
In brief, a handover request is sent to the next cell at a specic time instant,
which is determined by the handover threshold parameter. When a new call
arrives, it is accepted provided that an available channel exists in the current
cell. However, if the time interval until the occurrence of the rst handover
is shorter than the one dened by the handover threshold parameter, then an
available channel should also exist in the succeeding cell in order for the call
to be accepted. It was shown that this scheme can provide dierent QoS levels
based on the value of the handover threshold parameter.
In [47],[48], CAC algorithms based on a bandwidth allocation strategy
with priority queues are examined. The handover admission policy introduced
distinguishes between real-time and non-real-time services. To each accepted
real-time connection, bandwidth is allocated in a look-ahead horizon of 2 cells
along its trajectory; while non-real-time connections reserve bandwidth only
in the forthcoming cell. According to that scheme, each cell maintains four
dierent queues, called R, S1, S2 and Q. Queue R contains those real-time
connections that have reserved at least the minimum required bandwidth in
the next two cells. Therefore, the handover to the next two cells is guaranteed
to be successful. Queue S1 contains those real-time connections that have
reserved the required bandwidth in the next cell, but not in the one after the
next cell. Regarding queue S2, it contains the real-time connections that have
not managed to reserve the required bandwidth in both these cells. Finally,
queue Q contains the non-real-time connections that have not achieved to
reserve any amount of bandwidth in the next cell. It should be noted that
non-real-time connections are successfully handed over to a new cell as long as
some residual bandwidth, even lower than the minimum required bandwidth
for this type of calls, has been reserved in that cell. The management of
the queues is such as to give priority to real-time multimedia calls over
non-real-time data calls, namely the rst priority is given to queue S2, the
second is given to queue S1, while non-real-time connections are given the
lowest priority.
The study in [49] extends the aforementioned technique and proposes a
CAC algorithm that is based on the concept of multiple sliding windows.
The rationale behind the proposed algorithm is to predict the amount of
bandwidth that will be available at the time instant of the handover occurrence
and reserve the necessary amount of bandwidth in the cells to which the call
may be handed over. The highest priority is given to handover calls that are
organized in a separate queue.

In [50], the authors examine the use of the knowledge of future capacity
changes to trade-o some additional blocking probability, in order to meet
the desired CDP. Specically, three CAC policies based on the assumption of
deterministic capacity change time instants are discussed: two for calls with
exponentially distributed holding times, and one for calls whose holding time
distributions have Increasing Failure Rate (IFR) functions. In general, the
failure rate function h(x ) (also known as the hazard rate function) is dened
as:
b(x)
h(x) = (6.2)
1 B(x)
where b(x ) is the call holding time probability density function and B (x )
is the call holding time cumulative distribution function. Note that h(x )dx
denotes the probability that the call will end in the next dx time unit given
that it has been in service for x time units. A holding time distribution is said
to be an IFR distribution if h(x ) is a non-decreasing function of x. Examples of
IFR distributions are uniform, exponential, half-Gaussian distributions, and
gamma-n with n 1. Moreover, the Admission Limit Curve for exponentially
distributed call holding times, which forms a boundary on the conditions under
which a CAC policy may accept an incoming call request, has been proved to
be able to serve as the basis for a CAC policy. The authors demonstrate how
these CAC policies and the Admission Limit Curve represent progressive steps
in developing optimal CAC policies for calls with exponentially distributed
holding times, and they extend this process to the more general case of calls
with increasing failure rate call holding times. The Admission Limit Curve
was also investigated in [51] along with the performance of a CAC policy
for increasing failure rate holding time distributions. However, in that study
stochastic capacity change time instants were assumed.
6.4.2 Inter-satellite handover and CAC schemes
Although intra-satellite handovers are more frequent than inter-satellite han-

dovers, the latter are of paramount importance to the performance of any
non-GEO satellite system with partial or full satellite diversity. By the term
satellite diversity we simply mean that a terminal has a choice of multiple
visible satellites with which it can communicate. After opting for one of them,
the terminal establishes a single duplex radio link with that satellite. This kind
of satellite diversity is also referred to as switched diversity. Towards this end,
dierent satellite selection criteria have been proposed and evaluated [52],[53],
always with a view to minimizing CBP and CDP. The satellite selection
criteria that can be found in the literature can be summarized in the following
three rules:
Maximum capacity criterion - The satellite with the maximum available

capacity is selected. This criterion aims to attain a uniform distribution
of the trac load over the satellite constellation.
Maximum serving period criterion - The satellite that oers the maximum
serving time interval is selected. The aim of this criterion is to reduce the
number of handovers per call.
Minimum distance or highest elevation angle criterion - The closest
satellite (i.e., the satellite that is seen under the highest elevation angle)
is selected. This criterion aims to mitigate channel impairments.
The aforementioned satellite selection criteria can be applied to both new

and handover calls. All the results that are presented from this point forward
refer to Scenario 3, which is detailed in sub-Section 1.4.5 of Chapter 1.
In [52], the authors assess the guaranteed handover scheme as an inter-
satellite handover technique for LEO constellations that require at least one
satellite to be visible to both the user terminal and the Gateway Earth station.
The study in [53] extends the scheme proposed in [45],[46] for the case
of inter-satellite handovers in LEO satellite diversity-based systems. The
proposed scheme is evaluated for dierent values of the queuing time interval
as well as for dierent constellations. Moreover, it is evaluated for nine
dierent combinations of the satellite selection criteria.
In [54], an inter-satellite handover technique tailored for broadband LEO
satellite diversity-based systems is proposed. The proposed technique consti-
tutes a combination of the technique that is presented in [53] with the guard
channels scheme. By using dierent parameter values for each service class,
that technique aims to minimize CDP while keeping at the same time CBP at
acceptable levels. Specically, the value of the handover threshold parameter is
dierent for each service class with the aim of satisfying its QoS requirements.
Furthermore, the notion of the guard class capacity is introduced, which stands
for the portion of the total capacity that is available only to calls of a specic
service class. The rest of the capacity is available to calls of all service classes,
and calls contend in order to reserve the capacity required for their service.
Of course, the greater the mean bit-rate of the service class, the greater the
handover threshold and the guard class capacity employed for this service
class.
CAC and inter-satellite handover schemes geared towards multimedia
LEO satellite systems are also examined in [55],[56]. In both these studies,
a mobility model that takes the Earths rotation into account was used for
the assessment of the proposed schemes. In this model, satellite footprints
are modeled as rectangles. The overlapping area between successive satellites
in the same orbital plane is not taken into account since in that case a
user should always be connected to the following satellite in order to avoid
an immediate handover. However, the overlapping area between contiguous
satellites in dierent orbital planes is taken into consideration. Moreover,
terminals are uniformly distributed over the network. In addition to this, the
velocity of users in fast vehicles is disregarded since it is negligible compared

to the satellites ground track speed and the Earths rotation. The latter is
considered to be equal to the velocity at the equatorial level.
Reference [55] relies on the queuing of handover requests in order to
achieve low CDP. The services that the system supports are classied into
two categories, namely real-time multimedia services (namely, services with
stringent QoS requirements) and non-real-time data services (that is, services
with loose QoS constraints). Handover requests of dierent service classes
are stored in dierent queues. Priority is given to the queue where handover
requests of real-time multimedia connections are stored. As soon as a call
is successfully handed over to a satellite, a handover request is sent to the
next candidate satellites for relaying the call. Thus, the queuing time interval
can be equal to the users sojourn time in a satellites footprint. Moreover,
the proposed scheme was examined for dierent combinations of the satellite
selection criteria and for two dierent queuing policies. The rst one is the
well-known FIFO policy. In this scheme, the requests are served according to
their arrival time. The second queuing policy that was examined is called Last
Useful Instant (LUI) [38]. In this technique, the requests are queued according
to the remaining time interval until the handover occurrence. Hence, a request
is placed ahead of all the other requests in the queue that have a greater
remaining queuing time.
In [55], eight dierent versions of the scheme are compared. Figure 6.8
illustrates the performance of the techniques for dierent percentages of the
overlapping area. The overlapping area is dened as the percentage of the
footprints area that is overlapped by footprints of contiguous satellites. As
far as the notations in the legend of Figure 6.8 are concerned, the rst letter
of each scheme indicates the queuing policy that was employed; namely F
stands for the FIFO policy, while L stands for the LUI policy. The second
letter denotes the satellite selection criterion that was used for new calls,
whereas the third letter indicates the criterion that was employed for handover
calls; in these two cases, C denotes the Maximum capacity criterion, whereas
the letter T denotes the Maximum serving period criterion.
The schemes have been evaluated in terms of a cost function, which takes
account of CBP, CDP, and the mean allocated capacity of all service classes.
This cost function, which is called General Grade of Service (General GoS ),
in its general form can be expressed as:

N
General GoS = ai GoSi (6.3)
i=1
where N is the number of the service classes supported by the system and
ai is a weighting factor which is equal to
Bmini i
ai = (6.4)
i
where Bmini denotes the minimum capacity that is required for calls of
the i -th service class, whereas i and i are the arrival and departure rates
of calls of this type of service, respectively. Concerning GoSi , it is a function
of CBP and CDP of the i -th service class and is dened as follows:
GoSi = W F1 CBPi + W F2 CDP . (6.5)
The terms W F1 and W F2 represent weighting factors, which are the same
for each service class. It should be emphasized that W F2 is much greater
than W F1 (almost tenfold greater) since the forced termination of a handover
call is generally considered more irksome than the blocking of a new call.
Now it is evident that ai aims at giving an added bonus to the schemes that
attain higher mean bit-rate since it reduces the eect of the corresponding
GoSi on the General GoS. Regarding the latter, the higher its value, the
poorer the performance of the scheme and the QoS provided to the users.
It becomes evident from Figure 6.8 that the FIFO policy performs similarly
to the LUI policy. Notwithstanding, the FIFO policy is more appealing on
account of its low complexity. Furthermore, the combination that employs
the Maximum capacity criterion for both new and handover calls achieves the
best performance. Moreover, we can note that the General GoS increases
commensurate with the percentage of overlapping area. Nonetheless, the
overlapping percentage can be benecial for some types of services, as it is
shown in [55].
Fig. 6.8: General GoS of dierent schemes vs. overlapping percentage.

This study was extended and another CAC and inter-satellite handover
scheme has been developed and assessed in [56]. The main mechanism behind
this second technique is based on dynamic bandwidth de-allocation. According
to the proposed mechanism, capacity reservation requests are countermanded
when the capacity that they strive to reserve is unlikely to be used. In
the handover schemes proposed in [52]-[55] the decision about the satellite
to which the call will be handed over is taken at the time instant of the
handover occurrence. This means that capacity is reserved, if possible, in all
the visible satellites, and this capacity is released if the call is handed over
to another satellite. On the contrary, in the scheme proposed in [56], when
the capacity required for a call is reserved in one of the visible satellites, the
capacity reservation requests are cast away from the queues of the other visible
satellites. Hence, that scheme does not waste the limited bandwidth of the
satellite channel. Simulations showed that this scheme can also capitalize upon
the satellite diversity that a system may provide in order to enhance network
performance. Figure 6.9 depicts General GoS versus overlapping percentage
referring to the scheme proposed in [56].
Fig. 6.9: General GoS of dierent schemes vs. overlapping percentage.
It does not make sense to use a satellite selection criterion for handover
calls in this scheme, since the decision is taken before the time instant of the
handover occurrence. Thus, the rst letter of the acronyms in the legend of
Figure 6.9 denotes the queuing policy that was employed, while the second
letter indicates the satellite selection criterion that was employed for new
calls. As shown in Figure 6.9, the FC and LC schemes exhibit the best
performance. Recall that in Figure 6.8, the best performance was achieved by
those schemes that relied upon the Maximum capacity criterion for new calls
as well. Moreover, it can be observed that there exist signicant performance
disparities among the schemes that are presented in Figure 6.9 and the ones in
Figure 6.8. It is apparent that the schemes presented in Figure 6.9 outperform
those in Figure 6.8. The mechanism behind the schemes that were presented in
[56] (i.e., those related to Figure 6.9), which allows them to attain an enhanced
performance, relies on the cancellation of capacity reservation requests when
the capacity that they strive to reserve is unlikely to be used. Moreover, it
is evident that this scheme can capitalize upon the partial or full diversity
that a LEO satellite system may provide in order to attain an improvement
in system performance.
6.5 Directions for further research
This Section lists some rather interesting proposals for future research work
in the eld of CAC:
Due to the costly nature of the satellite channel, integrated CAC and
dynamic bandwidth allocation schemes are becoming a matter of some
concern to many network operators, being these integrated schemes able
to take into account both trac pattern variations and channel conditions.
In addition to this, the performance of transport layer protocols, such
as TCP, is often exacerbated by intense variations in the received signal
power and consequent high packet error rates. Consequently, the TCP
protocol perceives an indication of congestion in the network, thus reducing
the transmit information rate. In this context, a CAC algorithm able to
interact with the transport layer is considerably appealing, since it allows
estimating the amount of capacity that is currently in use, which is smaller
than the sum of the nominal capacity of every ongoing call. In particular,
the CAC algorithm should base its decisions on the goodput of the TCP
connections instead of the nominal bit-rate of each connection.
In hybrid architectures, namely integrated terrestrial-satellite networks or
multi-layered satellite networks, the role of CAC is twofold: (i ) to decide
which network is the most appropriate to serve a new call; (ii ) to decide
whether or not the call can be admitted to the network. A study of a CAC
algorithm able to regulate dynamically the admission of new connections in
an integrated network, according to their QoS requirements, user mobility,
and available resources, is of paramount importance.
An interesting scenario involves the integration of terrestrial and satellite
UMTS networks aiming at maximizing the number of connections that can
be actually admitted to the network. The decision of the CAC procedure
should be based on the terrestrial and satellite cell layout in the area where
the connection set-up attempt occurs, the surrounding area, the mobility
and the QoS requirements of the user, and the instantaneous trac load
in the terrestrial and satellite cells. Based on the aforementioned inputs,
the CAC algorithm should decide whether to admit or reject the call, the
QoS guarantees that will be granted to the call, and the segment as well
as the cell where it is more ecient to set-up the connection.
6.6 Conclusions
CAC constitutes an issue of paramount importance for any wireless or wired
network. It is performed at the connection set-up time and determines whether
or not sucient bandwidth is available to maintain required levels of QoS. In
this respect, CAC can be viewed as a preventive congestion control procedure.
With the advent of ATM networks, signicant research eorts have been
drawn towards CAC schemes. Typically, any CAC algorithm aims at taking
a decision based on two questions:
1. Does the new call impact on the QoS of ongoing calls?

2. Can the network provide the QoS requested by the new call?
Satellite systems have acquired an important role in the telecommunica-

tions arena. Over the years they have been used for a host of dierent services,
the most important ones being television and radio broadcasts. The current
trends towards the use of higher frequency bands open new opportunities
to this type of systems. Future satellite networks will be able to support
a wide range of multimedia applications. In this context, CAC algorithms
are necessary to guarantee a fair distribution of the radio resources and to
meet the QoS requirements of each service class. CAC techniques tailored for
broadband GEO satellite systems have been the subject of considerable study
lately.
Non-GEO satellite constellations inaugurated a new era in satellite com-
munications in the past decade. This type of satellite systems can constitute
a major asset to service providers by virtue of the appealing features that are
endowed with. One of the main characteristics (and problems) of non-GEO
satellite systems is the relative movement of satellites with respect to the Earth
surface. Consequently, in parallel with CAC techniques, handover schemes
become of great importance on account of the signicant probability of service
interruption. Several CAC and handover techniques have been proposed in
the literature for the case of non-GEO satellite systems, aiming at providing
a trade-o between call blocking probability and call dropping probability.
We are in the midst of a global revolution in information technology and
satellite systems can be instrumental in the emerging network infrastructure.
Nonetheless, CAC schemes for heterogeneous networks remain an issue to be
addressed.
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7
DYNAMIC BANDWIDTH ALLOCATION
Editors: Tommaso Pecorella1 , Giada Mennuti 1
Contributors: Nedo Celandroni2 , Franco Davoli3 , Erina Ferro2 , Alberto

Gotta2 , Stylianos Karapantazis4 , Giada Mennuti1 , Antoni Morell5 , Tommaso

Pecorella1 , Gonzalo Seco Granados5 , Petia Todorova6 , Mara Angeles Vazquez
5
Castro
1
CNIT - University of Florence, Italy
2
3
4
5
6
7.1 Dynamic bandwidth allocation: problem denition

Some of the appealing advantages of satellite networks, such as the wide
coverage and the conguration exibility, make them an ideal candidate for
providing multimedia services worldwide. Satellite bandwidth is, however,
a commodity at a premium, and an inecient utilization of it may negate
some of the aforementioned advantages. To this end, an apportioning scheme
that dynamically allocates the bandwidth among the satellite terminals, while
208 Tommaso Pecorella, Giada Mennuti
fullling the QoS requirements, is of paramount importance. Moreover, the

satellite scenario adds a new dimension to the treatment of bandwidth, owing
to the presence of both variable physical channel operating conditions and
large bandwidth-delay products. Typically, control actions in telecommuni-
cation networks need to be exerted over a wide range of time scales to cope
with events that may occur with frequencies ranging from milliseconds to
minutes or hours [1]-[4]. Satellite systems not only experience variable-load
multimedia trac, but also variable channel conditions and large propagation
delays. The variability in operating conditions is due both to changes in the
trac loads and to the signal attenuation on the satellite links due to the
degradations that result from atmospheric events, which particularly aect,
for example, the transmissions in the Ka band (20-30 GHz). The variability
due to changing radio channel conditions can be counteracted by means of
Adaptive Coding and Modulation (ACM) techniques that, however, modify
the available bandwidth for higher layers, thus aecting Dynamic Bandwidth
Allocation (DBA) schemes.
Ecient bandwidth utilization and QoS provisioning are, unfortunately,
two competing goals; therefore, DBA schemes seek for a trade-o between
them. To address the vast majority of IP trac, which is inherently bursty,
a technique that implicitly evaluates the bandwidth requirement at each
satellite terminal and manages the trac ows is essential. The purpose of
this Chapter is to present a number of solutions for assigning the satellite
bandwidth to dierent users (Earth stations) and trac types.
The combined action among various protocol layers (from the physical
layer up to the application layer) is likely to be a good way to combat channel
variability. However, this procedure could be too complex to obtain in the
widest possible extent, which would imply numerous cross-layer interactions
for control purposes and the related exchange of signaling information. In or-
der to obtain optimized policies for satellite bandwidth allocation, the actions
taken in a satellite network at the physical layer (where fade countermeasure
techniques are applied) can be combined with actions at the data link layer
(where the satellite bandwidth is allocated), thus realizing a more limited
cross-layer optimization. The complexity of this procedure lies in the fast
changing measurements required at the physical layer, regarding the channel
state (signal power-to-noise ratio), which might produce an unstable allocation
at the data link layer. Hence, the feedback information has to be properly
ltered, possibly with some hysteresis to obtain a stable allocation at the
data link layer.
Regarding resource allocation, another problem is the control network
architecture, which can be centralized or distributed. A centralized allocation
is performed by a station, which plays the role of master (or Network Control
Center, NCC). The master station collects all information relevant to the other
stations (slave stations) and performs the best bandwidth allocation. This may
produce a heavy computational eort in the master station. A distributed
allocation technique solves the computational problem, but requires a robust
Chapter 7: DYNAMIC BANDWIDTH ALLOCATION 209
control channel and an ecient control protocol, which takes into account the
large communication delay. As a consequence, the available bandwidth may
be signicantly reduced by the signaling protocol.
It should be observed, however, that the bandwidth allocation problem is
somewhat dierent for dierent satellite network topologies. While the typical
focus for GEO satellites is the ecient bandwidth assignment among terres-
trial gateways, for LEO satellites handover and call prioritization procedures
become crucial aspects.
In a GEO satellite system the main limit is the time delay, in a LEO
satellite system this issue is mitigated, but the system complexity causes
several problems. In order to achieve a continuous satellite access, a large
network of LEO satellites is required with regular handovers among them.
Achieving ubiquitous coverage poses a signicant challenge, and the speed at
which the satellites ground track moves on the Earth generates rapidly chang-
ing communication channels, subject to severe Doppler spreading. Moreover,
if a constellation of LEO satellites is designed to provide global coverage,
then these satellites must be able to communicate one to another, either
by incorporating Inter-Satellite Links (ISLs) or a ground-based hub station
in each footprint. All these issues contribute in making DBA an essential
approach for providing the proper QoS but, at the same time, make its design
very dicult.
A less treated problem, moreover, could arise from satellite-based mesh
architectures. So far, the system model only considers the uplink part, relying
on the assumption that downlink is not a bottleneck. In a meshed architecture
with multiple, limited-bandwidth downlink spot-beams, the channel allocation
will have to take into account also this aspect in order to maintain the overall
QoS; this is particularly important in satellite-based switching systems [5].
7.1.1 Survey of allocation approaches
DBA schemes can be distinguished as static and adaptive.
Static algorithms
In static schemes, once a terminal is assigned a certain amount of capacity,

this capacity remains constant for the connections lifetime. The terminal can
locally handle dynamically the bandwidth, without involving the NCC. That
is, the assigned capacity can be apportioned between High-Priority (HP) and
Low-Priority (LP) trac.
Adaptive algorithms
In the case of adaptive schemes, each satellite terminal can send requests to
the NCC in order to reserve or release channel capacity, based on its dynamic
estimation of bandwidth needs.

To meet the QoS requirements of bursty and delay-sensitive trac, the
terminal can follow three approaches:
Fixed allocation proportional to the maximum source rate, to be requested
on a per-connection basis,
Fixed allocation at a given rate using DBA for peak bursts,
Full DBA techniques.
The rst approach is inecient for satellite systems, as bandwidth is
allocated in a way that does not take into account the real needs of a station;
besides, the maximum source rate is usually unknown. As regards the full
DBA techniques, these can exploit the channel capacity with good eciency,
since no capacity is reserved during inactive periods. Notwithstanding, the
capacity request signaling channel may become overloaded during transient
changes in trac, leading to higher delays and congestion. Consequently, a
mixed approach seems to be the most exible choice, where each terminal is
assigned some xed channels of moderate capacity, while a number of DBA
channels are used during peak trac periods.
As far as adaptive schemes are concerned, one of the challenging problems
that engineers are called to grapple with is the implementation of these
techniques in a GEO satellite system. The main problem stems from the high
delay between the time instant that a request is sent to the NCC and the
time instant at which the satellite terminal is informed about the bandwidth
that has been allocated to it. This latency prevents immediate changes to
the allocated capacity. Since a low latency entails better performance, a
GEO satellite system represents the worst case (approximately 500 ms when
the NCC is terrestrial-based or 250 ms when the majority of processing is
supported by the satellite as a part of its on-board capability).
Adaptive DBA schemes are generally categorized as either reactive or
proactive algorithms. Reactive schemes take into account the current queue
length, the packet loss and the average delay in order to react to trac
uctuations, without trying to anticipate them. Compared to proactive al-
gorithms, reactive algorithms are easier to implement and can utilize the
channel capacity more eciently. However, QoS requirements are not easily
met, since the requests are delayed by sending them to the NCC, and do
not therefore necessarily represent the current bandwidth needs. In [6], the
authors proposed a novel predictive bandwidth allocation and de-allocation
scheme, which frees up bandwidth allocated to connections that are unlikely
to be used. The look-ahead horizon of k cells is introduced, where k = 2. The
scheme provides the lowest Call Dropping Probability (CDP) for real-time
connections with respect to previous schemes.
Even though reactive schemes may perform well in LEO satellite networks,
they are not well suited to GEO systems owing to the high propagation delay.
proactive schemes aim at analyzing the trac and predicting the required
bandwidth. Usually, this is realized by providing a predictor with data up
to time t (e.g., with the queue lengths, the input ows and output ows);
such data are used to make a prediction at time t of the aggregated trac
in the interval [t, t + k ] (e.g., the trac within the next superframe, where
a superframe is the aggregation of k consecutive frames). Depending on the
number of simultaneous trac ows (i.e., TCP connections, application data
streams) and the QoS model in use (i.e., DiServ or IntServ), dierent trac
prediction techniques can be adopted. In a single-user per satellite terminal
scenario, an IntServ-based QoS model will be assumed, whereas for a large
aggregate of users per terminal, a DiServ model seems more appropriate.
When the number of data ows is very small, e.g., for a single-user per
satellite terminal, trac predictors may exploit the possibly known trac
patterns, like the TCP slow-start and the IntServ trac information, in order
to reserve the appropriate resources. If this is not viable, as in a DiServ
model approach, the trac predictions can resort to utilizing the statistical
properties of IP trac. Hence, the required bandwidth can be estimated.
In order to make adaptive predictions, i.e., capable of following changes in
the trac characteristics over time, the parameters of the predictor can be
regularly updated. The performance of these schemes heavily relies upon the
accurate prediction of future trac.
7.2 DBA schemes for DVB-RCS scenarios

In a DVB-RCS return link, users are multiplexed by means of a Multi Fre-
quency - Time Division Multiple Access (MF-TDMA) scheme. The DVB-RCS
standard [7],[8] permits full exibility in the way the bandwidth is divided (see
a feasible example in Figure 7.1, left upper corner [9]). The adopted solution
in this Section consists of an independent division of both time and frequency
axis, that is, bandwidth is divided into several carriers and the time duration
of the superframe is divided into timeslots. Carriers do not have necessarily
the same transmission bandwidth (dierent types of carriers are possible) and,
at the same time, the timeslot duration can be dierent from one carrier to
another.
Return Channel Satellite Terminals (RCSTs) ask for some amount of
system capacity to the NCC through capacity requests. In the DVB-RCS
standard, three types of capacity request, from highest to lowest priority,
are considered: CRA, RBDC, and VBDC. Free Capacity Assignment (FCA)
usually is not taken into consideration by DBA schemes, since it may be
granted by the NCC, but not requested [10]. Please refer to Chapter 1 for
more details on these resource allocation methods.
Note that the requests generated by all RCSTs in a beam constitute the
inputs of the bandwidth allocation problem and in principle it is not necessary
to consider how RCSTs generate requests. For each bandwidth allocation
update (which is done on a superframe basis) the NCC sends a Terminal
Burst Time Plan (TBTP) to the RCSTs. This message indicates the time
Fig. 7.1: System model. See reference [9]. Copyright 2006

c IEEE.
and the frequency that each RCST should use to transmit (see Figure 7.1
[9]).
In any case, the standard does not give strict constraints on the algorithms
to be used in the resource allocation process; hence, it is possible to develop
advanced techniques by using the standard request types. The only weakness
of the standard is related to the lack of information contained in the requests;
hence, two requests of the same type will have to be considered as equal,
even if the requesting RCSTs have to deliver dierent kinds of trac (e.g.,
volume-based requests for high priority and low priority trac).
The next improvements in DVB-RCS-based allocation strategies will be
focused on two topics, both related to a cross-layer approach. The rst one
will be to consider the eects of fading countermeasures; the second one
will be to dene a simple interface for upper layers, in order to develop a
cross-layer QoS manager, able to tune the allocation process to the actual
QoS requirements, possibly considering a pricing system, i.e., taking into
account the user willingness to pay. A possible protocol architecture to support
cross-layer interactions is proposed in sub-Section 1.6.2 referring to the BSM
standard.
7.3 Recent developments on DBA techniques
7.3.1 DVB-RCS dynamic channel allocation using

control-theoretic approaches
One of the main issues with proactive DBA is the accurate prediction of
future trac. Trac predictors are usually aected by errors due to unex-
pected network behaviors (e.g., packet loss, network congestion, etc.), TCP
behavior and, more generally, uncertainty in the user interactions. Coupling
the trac predictors with appropriate control-theoretic techniques, however,
allows maintaining the required QoS with an acceptable computational eort.
In a DVB-RCS GEO satellite system, the NCC receives the bandwidth
requests of each RCST and decides whether to satisfy or not these requests
on the basis of a fair policy of resource sharing among all the RCSTs. In order
to meet the desired QoS, both the request algorithm and the NCC allocation
strategy are of paramount importance.
In [11]-[13] the authors compared some dierent allocation strategies based
on trac prediction, assuming that each RCST is used to transmit a heavy
aggregate of trac. Figure 7.2 shows the proposed system model. It can be
observed that the bandwidth controller must take into account the trac
predictions, the actual queue sizes and the packet scheduler behavior, to satisfy
the bandwidth requests. In the gure, the NCC is depicted as a simple delay
with a disturb, due to the possibility of denying a bandwidth request.
Fig. 7.2: RCST system model.

Simulations have evaluated the bandwidth loss, measured as the number

of slots per frame allocated to an RCST but not used to transmit data, and
the QoS performance of two DBA techniques based on a Receding Horizon
Controller and a Smith Predictor Controller (RHC and SPC, respectively)
versus three xed (Fix n in Figure 7.3, where n is the number of slots/frame
assigned to a single RCST) allocation schemes with dierent bit-rate values
[13]. A self-similar trac with Hurst parameter H = 0.8 has been used to feed
each RCST. Figure 7.3 reports the bandwidth loss distribution (a) and the
delay for the Expedited Forwarding (EF), Assured Forwarding (AF), and Best
Eort (BE) DiServ [14]-[17] trac classes [(b), (c) and (d ), respectively].
It can be observed that the bandwidth loss is greatly reduced by using DBA
techniques, whereas the overall QoS of the trac classes is acceptable, in
particular by using the RHC scheme.
However, despite the great advantages of DBA techniques, both in terms
of QoS satisfaction and ecient resource utilization, some new problems arise
when they are applied. In particular:
Greedy trac ows can compromise the whole satellite systems QoS.
Compatibility between dierent control techniques and bandwidth request
methods should be validated.
Security issues on the signaling channel should be analyzed in order to
prevent denial of service attacks based on fake bandwidth reservations.
Those issues should be investigated and carefully addressed before using
DBA techniques in any actual system.
7.3.2 Dynamic bandwidth de-allocation
Several approaches for bandwidth and handover management have been

studied in the recent literature in the case of mobile satellite systems.
Publications in this area investigate only bandwidth allocation and the intra-
satellite handover management. In reference [18], an advanced bandwidth
management strategy is proposed and evaluated, allowing for bandwidth allo-
cation/deallocation and a novel inter-satellite handover management scheme,
tailored for multimedia LEO satellite networks with satellite diversity. The
main mechanism is based on bandwidth de-allocation. According to the
proposed scheme, capacity reservation requests for handover calls are removed
from the queues when the capacity that they strive to reserve is unlikely
to be used. Simulations conrmed the usefulness of bandwidth de-allocation
mechanism. Other details of this scheme have been already discussed in
sub-Section 6.4.
7.3.3 Dynamic bandwidth allocation with cross-layer issues
Some examples of cross-layer DBA schemes are here briey discussed, limiting
the description to recent works [19]-[25]. An overview of cross-layer approaches
Fig. 7.3: DBA in GEO satellite systems.

in this context can be found in [26],[27].

The rst example considered concerns the presence of mixed Guaranteed
Bandwidth (GB) real-time trac and Best Eort (BE) trac. GB is subject
to CAC, which is exerted independently by local controllers, situated at
the Earth stations (i.e., RCSTs), within an amount of bandwidth that
is assigned to these stations over a certain allocation time interval. The
bandwidth allocation is the responsibility of a master station and can be done
periodically or on-demand. Best-eort trac is represented by an inelastic
model (i.e., the congestion control mechanism of TCP is not explicitly taken
into account), used to compute the loss probability of cells (Asynchronous
Transfer Mode, ATM or DVB), stored in the Earth stations buers; this
trac utilizes the bandwidth that remains available after serving the GB class.
The cross-layer interaction stems from the fact that a fade countermeasure,
based on bit and coding rate adaptation, is used at the physical layer,
whose inuence on the bandwidth allocation is accounted for by redundancy
coecients [representing the inverse of the ratios of the Information Bit
Rate (IBR) in the specic channel condition to the one in clear sky]. Various
methods have been considered for bandwidth allocation, and the overall
structure has been evaluated in the presence of real fading traces [19],[20].
Figure 7.4 [20] represents call blocking, call dropping (due to a temporary
lack of bandwidth) and cell loss probabilities for three dierent allocation
strategies: (i ) cross-layer Optimized Centralized (OC, where the bandwidth
is allocated on demand by the master station, which solves a centralized
optimization problem); (ii ) cross-layer Optimized Proportional (OP, where
optimal allocation requests are computed locally by the Earth stations and
then passed to the master, which re-scales them and distributes the bandwidth
proportionally); (iii ) Simple Proportional (SP, based on oered load, with
no cross-layer dynamic allocation). The reported results refer to a 10,000 s
simulation, with 10 Earth stations, 5 of which experience dierent fading
conditions, whereas 5 operate in clear sky. In these graphs, the probabilities
for each point in time are computed by averaging over all stations in the
system, and over a time window of 1,000 s. The fading is dynamically variable,
according to real traces.
The advantage of the cross-layer allocations lies in maintaining blocking
probability values below a given threshold (5% in the specic case), while
minimizing the call dropping and the BE trac cell loss probabilities in the
stations buers.
The second example deals with DBA in the presence of only inelastic
packet trac with two stations, whose trac loads periodically alternate
between a lower and a higher value. Figure 7.5 [22] illustrates the convergence
properties of a gradient descent technique, based on Innitesimal Perturbation
Analysis (IPA) [21]-[23]. Station 2 is in clear sky, whereas station 1 also
experiences fading variations, besides those in trac load. The bandwidth
allocation provided by the IPA gradient estimation, based only on on-line
measurements, is capable to face both dynamic eects in order to minimize
Fig. 7.4: Call blocking and dropping (left) and cell loss (right) probabilities.
These graphs are reproduced from Adaptive Cross-layer Bandwidth Allocation in a Rain-faded
Satellite Environment, N. Celandroni, F. Davoli, E. Ferro, A. Gotta, International Journal of
Communication Systems, Vol. 19. No. 5, pp. 509530, June 2006. 2006.
c Copyright John Wiley
& Sons Limited. Reproduced with permission.
Fig. 7.5: IPA gradient descent allocation, under trac load and fade changes. See
c IEEE.
the overall loss volume.

The problem considered in [21],[22] is a pure parametric optimization. In
order to avoid transient periods in the convergence of the on-line gradient
descent technique, a dierent point of view can be adopted [23] where open-
loop feedback control strategies (i.e., stemming from a functional optimization
approach) are approximated by means of neural networks.
Finally, a DBA cross-layer optimization, aiming at achieving the best
compromise between the TCP goodput maximization and fairness, has been
treated in [24],[25], in a GEO bent-pipe satellite scenario. The numerical
details of the example shown here are the same as in [25], with a combination
of long-lived TCP NewReno connections, sharing various bottleneck links,

determined by 10 dierent fading classes (stemming from dierent source-
destination pairs), under the Hotbird 6 link budget [28] and real fading traces.
The instantaneous goodput is determined by the dynamic bandwidth and
redundancy allocation, which aims at counteracting fading eects and achiev-
ing a compromise between maximizing the total goodput and maintaining
fairness among connections.
Figure 7.6 [25] shows the behavior as a function of time of the overall
goodput (the points are the results of a moving average over a 10 s window) for
two classes operating under dierent fading conditions (note that the carrier
power-to-noise spectral density ratio, C /N0 , for the source-destination pairs
is also shown). The used strategies attempt to maximize the total goodput
and to maintain fairness in dierent ways (see [25] for a description of these
strategies):
The merge strategy is the best choice between two alternative methods
(tradeo and range, respectively) that establish a balance between
goodput and fairness;
The proportionally fair technique maximizes the sum of the logarithms
of the individual goodputs, so as to attain a Nash Bargaining Solution
(NBS);
The BER threshold strategy simply adjusts the redundancy to keep
always BER below a given limit, and assigns the bandwidths proportion-
ally to the redundancy and the number of connections of each class (no
cross-layer action).
The advantages of the cross-layer strategies, shown in detail in [25], are

not only in terms of goodput, but also in terms of fairness.
7.3.4 Joint timeslot optimization and fair dynamic bandwidth

allocation in a system employing adaptive coding
In [29], an enhanced and multi-beam DVB-RCS system is addressed, consid-

ering both Adaptive Coding (AC) and dynamic framing. AC arises when the
transmission is severely aected by channel conditions (as in the Ka band).
In order to keep the link active, framing design must be exible enough
to adapt in time and frequency, to allow for the use of dierent carriers
(this technique is also known as Dynamic Resource Allocation, DRA) and/or
dierent protection-levels of channel coding (AC).
The problem of optimal framing has been already addressed in the
literature. For example, in [30] a method is presented for optimal super-frame
pattern design for the DVB-RCS MF-TDMA return link, so that the system
data throughput is maximized. The authors formulate the design problem
as a non-linear combinatorial optimization problem. However, the developed
method considers static framing and, therefore, it is not extensible to Ka band
Fig. 7.6: Merge, Proportionally Fair and BER Threshold (thr = 106 ) strategies.
A class in fading (a); a class in clear sky (b). See reference [25]. Copyright 2006
c
IEEE.
transmissions, where adaptive physical layer is used and adaptive framing is

more appropriate. In [31], bandwidth segmentation on a super-frame basis is
also presented, but slots are of xed duration within the dierent types of
carriers.
Current systems typically make use of xed framing for its simplicity.
Instead, the DVB-RCS terminal considered in [29] is assumed to have trans-
mission capabilities suciently agile to realize multimedia communications
under adverse channel conditions. This is achieved by using the dynamic-slot
MF-TDMA feature of the standard, integrated with AC that is adapted on a
super-frame basis. This strategy segments the total bandwidth into dierent
types of carriers, according to user trac demands and weather conditions.
It is assumed that slots can be assigned to users with dierent coding rate
on the same carrier, which leads to coexisting slots of dierent size, which is
called adaptive framing, due to AC. The total bandwidth is segmented into
several carriers that can be of dierent bandwidths and the slots contained
in the frames can be of various durations, according to the chosen coding
rate; users can be granted slots of dierent durations on dierent carriers
(sequentially). Dierently from other studies, bandwidth is segmented not
only in the presence of dierent trac types, but also assuming realistic
dynamic weather conditions, to which coding rate is adapted.
Capacity is allocated giving priority to heavy rain-aected users, then
considering less aected ones, and ending with clear sky users, while there is
still bandwidth available. The major issue to keep into account concerns the
limits of capacity that can be allocated, due to adaptive framing.
In this study, the time dimension is partitioned into super-frames, a
super-frame into frames and frames into slots. The super-frame length is 26.5
ms and seven dierent coding rates (1/3, 2/5, 1/2, 2/3, 3/4, 4/5, 6/7) are
considered; the modulation is QPSK. Regarding the frequency dimension, it
is assumed that the total bandwidth can be dynamically segmented, from
super-frame to super-frame, and that up to four dierent carrier types can
be used in a super-frame: 540 kHz (carrier type I), 270 kHz (carrier type II),
135 kHz (carrier type III) and 67.5 kHz (carrier type IV). The roll-o factor
is 0.35, providing symbol rates of 400, 200, 100 and 50 kbaud, respectively.
The number and type of active carriers is adapted to the trac requests
and the needs of the users, which vary according to channel conditions. The
transmitted packet can be an ATM cell or an MPEG packet (for numerical
evaluations we will only refer to ATM cells). With AC, the length of the slots
transmitting such xed-length packet becomes variable and, therefore, the
number of slots contained by a given type of carrier becomes variable, as well.
Not all the users are necessarily always active. Active users are divided into
categories, according to both their symbol energy to noise-plus-interference
spectral density ratio, Es /No,tot and trac characteristics. Trac is assumed
to be uniform and the considered classes are: Constant Bit Rate (CBR),
Variable Bit Rate (VBR), and BE. For simplicity and without loss of gen-
erality, one user is assumed to ask only for one of these trac classes, so that
each user will have allocated slots of a given xed length on the carrier type
corresponding to its Es /No,tot [29]. Trac demands are queued according
to the type of DVB-RCS capacity request, which can be CRA, RBDC, and
VBDC. Capacity requests are prioritized: CRA has the highest priority and
VBDC the lowest. CBR trac is assigned to CRA as a whole, whereas VBR
trac is assigned to CRA and RBDC. Similarly, BE trac is also divided
between RBDC and VBDC.
The number of carriers of each type is computed at every super-frame,
given priority to the users aected by rain. Assuming a given Es /No,tot for
the user and some given requests for the current super-frame, a closed-form
estimation of the number of carriers required per carrier type is computed in
terms of an estimation of the number of slots as follows:
C C

(rCBR +rV BR )Ts
nC C,CBR
ni (s) = nni (s) + nC,V
ni
BR
(s) = NnCi (s) i L(i ) ,
(7.1)
i = 1, 2, ... , N AC

(rVRBR +rBE
R
)Ts
nR
ni (s) = nR,V
ni
BR
(s) + nR,BE
ni (s) = NnRi (s) i L(i ) ,
(7.2)
AC
i = 1, 2, ... , N
V

V +rBE Ts
nVni (s) = nVniol (s) + nV,BE
ni (s) = NnVi (s) i L(i ) ,
(7.3)
i = 1, 2, ... , N AC
where s is an index making reference to the super-frame, which consists

of 10 frames and lasts 265 ms, nX,Y ni (s) is the number of requested slots
corresponding to capacity request type X (X = C, R or V, which correspond
to CRA, RBDC or VBDC requests, respectively) of trac class type Y (Y =
CBR, VBR or BE) requiring spectral eciency i , nX ni (s) is the total number
of requested slots corresponding to capacity request type X, NnXi (s) is the
number of users requesting capacity type X, Ts is the duration of a super-frame
in seconds. N AC is the number of possible coding rates, and rYX is the bit-rate
requested by trac class Y that is mapped to request type X. V is a possibly
additional amount of bits requested as volume (instead of bit-rate), which
results in nVniol (s) slots, and L(i ) is the length in bits of the packet.
The number of carriers of each type is estimated from the total number
of slots needed according to (7.1)-(7.3). The fragmentation of the bandwidth
into carriers is performed, starting from the heavy rain-aected users down
to the clear sky ones, while there is still bandwidth available. With all these
assumptions, a key result has been obtained in [29] by applying cross-layer
design for DVB-RCS with AC. The user satisfaction strongly depends on the
distribution of users relative to the spatial distribution of channel conditions.
As a conclusion, smarter scheduling policies should be designed, taking into
account this eect in order to design fairer bandwidth allocation schemes.

In what follows, a smarter scheduling policy is proposed, based on a joint
optimization accounting for both a fair bandwidth distribution among users
and channel conditions and timeslot duration.
Proposed framework
Recall that in DVB-RCS, the TBTP (composed of several frames (F) of

duration TF ) is updated and transmitted every super-frame. If BW is the
total system bandwidth, then the scheduler is in charge of solving an allocation
problem for each BW TF block (note that it can also be applied to the whole
super-frame). BW is divided into dierent carrier types to serve dierent
users, accounting for dierent Service Level Agreements (SLAs), location and
terminal equipment. The problem to be faced consists in multiplexing N users
into C carriers of BWi bandwidth that transmit into a frame of TF seconds.
An ETSI specication [32] imposes a number of constraints to the problem,
namely:
The total transmission capacity (i.e., carriers) in the satellite beam is
divided in areas.
The symbol rate and slot timing must be the same for all carriers in one
area. Coding rates are not necessarily the same.
A given RCST belongs to one (and only one) area and can use only one
carrier at a given time.
Hence, it is possible to simplify the problem creating sub-problems, one

for each group of carriers of the same type (see Figure 7.7, on the left [9]).
It is meaningful to consider that the RCSTs in one area, while transmitting
in a common carrier type, use the same transmission rate. Note that the
DVB-RCS standard denes an adaptive-coding physical (PHY) layer with
several possible coding rates, so the mapping of users to areas is basically
dened by the quality of the link (channel conditions). As before, the minimum
transmission unit (a layer-2, MAC, packet) can be an ATM cell (53 bytes) or a
Moving Picture Experts Group (MPEG) container (188 bytes). The following
analysis is related to the case of ATM cells.
Following the previous discussion, the aim here is to obtain TBTP reduced
signaling for frame description (excessive signaling in the Frame Composition
Table, FCT, entails a reduction in bandwidth eciency). A timeslot with com-
mon duration for all areas is imposed (1 ), allowing a very simple assignment
1
Note that xing a timeslot duration common to all areas introduces some unused
bandwidth that depends on both the timeslot duration and the packet length
(ATM cell in our case). However, once a given RCST has been assigned to a certain
timeslot, it can change its transmission rate inside the timeslot without aecting
the transmission timing of the other RCSTs. This argumentation validates the
robustness of the solution proposed.
Fig. 7.7: Scheduling (bandwidth allocation) problem. See reference [9]. Copyright
2006
c IEEE.
procedure (after having known the number of timeslots per area): from left
to right and from top to bottom (according to the reading order). Regarding
signaling issues, this is translated into a simple FCT, since it indicates the
common timeslot type (which is described in the Time Composition Table,
TCT) and how many times it is repeated in the carrier. On the basis of the
area rate, one or more ATM cells can be transmitted in a single timeslot.
A possible timeslot and ATM cell assignment is shown in Figure 7.7, on
the right [9]. The problem of how to assign timeslots to areas and ATM cells to
RCSTs is discussed later, after introducing the scheduling hierarchy concept
[32].
Scheduling hierarchy
The general scheduling problem (which may involve thousands or more

RCSTs) may be complex to solve. Therefore, it seems reasonable to reduce
it to some smaller problems by imposing some known structure (that can
also facilitate signaling). This is an idea similar to that proposed in [33]
(particularly in centralized optimization algorithms). According to [32], some
minimum resources are guaranteed to the service providers. Since the relative
RCSTs for each service provider can be distributed over dierent areas, in
[32] the scheduling hierarchy presents the segment concept, i.e., a grouping of
RCSTs in a given area with a minimum predened amount of resources that

must be guaranteed to them (see Figure 7.8 [9]).
Fig. 7.8: Scheduling (bandwidth allocation) hierarchy in DVB-RCS. See reference

[9]. Copyright 2006
c IEEE.
Hence, the scheduling strategy proposed is (resources can be imagined as

ATM cells, but the solution can be applied identically to MPEG packets):
Compute per-area aggregated user requests and guaranteed minimum.
Assign the guaranteed minimum to the areas. Allocate the remaining
resources with a fair algorithm (see the next paragraph).
For each area, the problem of distributing the assigned resources among
the area terminals is solved similarly to the problem of distributing all the
satellite resources among areas:
Collect the user requests, but now per segment.
Assign the minimum amounts to the segments.
Allocate the remaining area resources among the segments, with a fair
algorithm.
Finally, the resources allocated to each segment are distributed among
the RCSTs associated to that segment taking into account the priorities
established by the capacity requests:
Allocate the resources to the users, depending only on their CRA requests
(highest priority).
Assign the remaining resources on the basis of the RBDC requests.
Assign the remaining resources on the basis of the VBDC requests.
In the next paragraph the problem of fair assignment of resources is stated

and solved.
Fair resource allocation
The following maximization problem is presented in [33], where a fair al-

location of P resources among N entities (areas, segments or terminals) is
achieved:

N
max xi
x1 , ... ,xN
i=1

N
subject to xi P (7.4)
i=1
dmini xi dmaxi
where xi is the amount of resources allocated to entity i, dmini is the part

of resources guaranteed to i, whereas dmaxi (also indicated in what follows as
di ) is the request of i.
Compared to [33], the main dierence is that now a minimum resource
allocation must be guaranteed to each entity. As consequence, the solution
is slightly changed. Note that it is simple to convert (7.4) in a convex
optimization problem [34] with the application of the logarithm function to the
objective. Moreover, the resulting problem is analytically solvable by means
of the Karush-Kuhn-Tucker (KKT) conditions [34], that force the following
solution:

1
, dmini 1 dmaxi
dmini
1
xi = dmini , (7.5)

dmaxi
1
dmaxi ,
N
where is a positive value that implies i=1 xi P .
It is possible to achieve the solution in a graphic way, by simply lling a
container (shaped accordingly with guaranteed resources and demands) with
an amount P of water (Figure 7.9 [9]).
Since (7.4) is solvable, the solution rstly assigns the minimum amounts
(namely, pale water) and then fairly distributes the rest (namely, strong
water). In this case, the solution is generally computed for a real-valued
Fig. 7.9: Fair resource distribution solution. See reference [9]. Copyright 2006
c
IEEE.
problem, but it is possible to obtain the particularization to the integer case

(i.e., with integer variables xi ), simply assuming one extra resource (round
up) to a subgroup of users, sharing the same number of resources, and round
down the remaining ones (dotted line in Figure 7.9).
Focusing now on the highest level in the scheduling hierarchy of Figure 7.8,
i.e., the timeslot distribution among areas, a possible design option consists in
setting the timeslot duration TT S to t exactly an ATM cell of the area that
transmits with the lowest rate. Higher rates allow transmitting more than one
ATM cell per timeslot (see Figure 7.7, on the right). Simply analyzing this
example, one can realize that some bandwidth remains unused. One obvious
question is if it is possible to set up TT S in order to reduce this ineciency (this
aspect will be addressed in the next paragraph). Note that TT S is forced to be
in the interval [Tmin , Tmax ], where these values are left as design parameters
(the reader can nd an example later on in the text). However, the lower limit
must full Tmin t1 , being t1 the largest possible ATM cell duration in the
system.
Area size selection and timeslot optimization
The problem in (7.4), already providing fair resource distribution, is extended

here to include the timeslot optimization. For the sake of simplicity, it is not
essential to consider now the minimum guaranteed resources. If NAT Mi is
the number of ATM cells allocated to area i, Ni is the number of timeslots
allocated to area i, andTT S the timeslot duration, the analyzed problem

N
maximizes the product i=1 NAT Mi :

N
max Ni Ki (TT S , ti )
TT S ,N1 , ... ,NN
i=1

N
subject to Ni NT OT (C, TF , TT S ) (7.6)
i=1
0 Ni Ki (TT S , ti ) di
Tmin TT S Tmax
where ti is the time duration of an ATM cell transmitted at rate ri for the
i -th area, Ki is the number of ATM cells that t in a timeslot (depending on
both the ATM cell duration, ti , and TT S ) and NT OT is the total number of
timeslots (depending on the number of carriers, the frame duration and the
timeslot duration).
The input data di must be in principle considered as MAC layer informa-
tion. However, it may be interesting to think about cross-layer mechanisms to
enable some network inuence in di (this is the case, for example, when the
RCSTs send network layer information to the NCC or the requests made at the
RCSTs take into account that information). Moreover, the proposed technique
requires PHY cross-layer information (the area rates ri ) and it inuences both
the MAC and PHY layer of the RCSTs (the latter being done through TT S
adjustment).
It is possible
to solve
the problem xing TT S and then optimizing over
the Ni s. Let Ni1opt be the solution to this problem. Fixing these values,
optimization over TT S is a one-variable optimization problem. Imagine the
solution is TT1 Sopt . Iterations of this mechanism would drive into the optimal
joint solution if the problems were jointly convex, so it is mandatory to x
both problems.
Fixing TT S , the problem:

N
N
N1 , ... ,NN
i=1 i=1

N
i=1

di
0 Ni
K (TT S , ti )
is convex, where the ceiling function () is necessary in the integer case
in order to avoid the situation of one area that requests some ATM cells, but
does not receive any timeslot. In this case, the problem in (7.7) is equivalent
to the integer version of (7.4) and, thus, the solution is known.
The following problem for the timeslot optimization (developing expres-
sions for the Ki s and NT OT ) is achieved xing the Ni s:

N
TT S
max Ni
TT S
i=1
ti
N
TF
subject to Ni C (7.8)
i=1
TT S

d
0 Ni
T
i

T S/t
i
Tmin TT S Tmax .
The oor function ( ) is obviously necessary to convert this problem into

non-convex and, hence, the joint problem too. However, the problem can be
easily solved if the integrality that the oor function introduces is exploited.
Look at the following observation:
Departing from a feasible value of TT S and increasing it, it can only
reduce the objective value unless a multiple value of some of the ti s is
reached .
It is possible to note that the only meaningful values of TT S are the
multiples of the ti values, into the interval [Tmin ,Tmax ]. The values comprised
between any of these special values do not allow to place an extra ATM cell
inside any timeslot at the expenses of a potential decrease in NT OT . In the
case under consideration, where there are few areas and the same amount of
ti s, the possible TT S values are not so many and (7.8) can be simply solved
via exhaustive (but small) search.
The optimization procedure for the joint problem consists of:
Identify the possible values of TT S .

Suppress equal values coming from multiples of dierent ti s.
Optimize the Ni s for each possible value.
Select {TT S ,Ni } with best objective value in (7.6).
In order to guarantee the optimal solution, the joint convexity is not

necessary, as there are only a few valid values of TT S and it is sucient
to explore the optimality of each of them. Next, some results showing the
importance of taking a good choice of TT S are given.
Let us consider a scenario with C = 111 carriers of 540 kHz bandwidth and
TF = 26.5 ms. Imagine a DVB-RCS situation, with the RCSTs transmitting
via 7 dierent coding rates and, hence, 7 dierent ATM cell durations are
possible (namely, one area per coding rate is dened). The relation among
areas, coding rates and ATM cells duration is presented in Table 7.1 [9]. A
Quadrature Phase Shift Keying (QPSK) modulation is assumed, transmitted
through a raised cosine pulse with roll-o factor equal to 0.35. Consider that
the timeslot duration can be adjusted between Tmin = t1 and Tmax = 3t1 .
Area identier Coding rate ATM cell duration

1 r1 = 1/3 t1 = 1.59 ms
2 r2 = 2/5 t2 = 1.325 ms
3 r3 = 1/2 t3 = 1.06 ms
4 r4 = 2/3 t4 = 0.795 ms
5 r5 = 3/4 t5 = 0.706 ms
6 r6 = 4/5 t6 = 0.6625 ms
7 r7 = 6/7 t7 = 0.6183 ms
Table 7.1: Denition of areas. See reference [9]. Copyright 2006

c IEEE.
In what follows, it is explained how to compute the aggregated demand of

RCSTs (number of requested ATM cells) per area. The Aggregated System
Demand (ASD) is dened as the mean of the sum of all demands in all
areas. Such demand is distributed among the areas according to a given
distribution p. Note that areas with higher rates accumulate more requests
since it is expected that most of the RCSTs can be found in areas with good
weather conditions. Low rate areas are designed to satisfy the transmission
requirements of RCSTs aected by rain and a small part of RCSTs can be
assumed in that situation (considering that RCSTs are uniformly distributed
in space). As an example, take into account the distribution p = [1/15, 1/15,
2/15, 3/15, 3/15, 3/15, 2/15]. After obtained ASD per area (it is a mean
value), a realization of demand in each area using a uniform probability density
function (pdf) with the given mean ASD is computed.
For convenience, let us dene a reference ASD value, which corresponds to
the capacity transported by the system when only the highest rate transmits
and TT S = t7 (i.e., the maximum possible transported capacity). In the
particular case studied, ASDref = 4662AT Mcell /frame, which corresponds
to 74.6 Mbit/s, and ASD can be greater than the ASDref value. Note that
it is possible to consider a scenario where the highest rate area asks the
reference ASD while the other areas ask their own maximum transport
capacity (depending on the area rate and, obviously, less than the reference
ASD).
In the results, computed via the Monte Carlo method, the fair allocation
algorithm that solved (7.6) is compared to an opportunistic design, analyzing
in both cases the eect of timeslot optimization. For the sake of completeness,
assume that the opportunistic design nds the optimal values of the following
problem:

N
TT S ,N1 , ... ,NN
i=1

N
i=1
0 Ni Ki (TT S , ti ) di
Tmin TT S Tmax .
In order to obtain the relative solution, it is necessary to assign all the

demand (until there are resources left) of the highest rate area and iterating
this procedure for each area (ordered by transmission rate) until the lowest
rate area is reached (if possible depending on the available resources). Note
that this design assures maximum transported capacity at the expenses of
oering poor QoS to users with degraded channel conditions, in general.
The rst analysis in Figure 7.10 [9] studies the Bandwidth Occupation
(BO), dened as:
7
i=1Ni t i
BO = . (7.10)
C TF
With the optimization of TT S , the occupation for both fair and oppor-
tunistic strategies is signicantly improved.
Fig. 7.10: Bandwidth occupation. See reference [9]. Copyright 2006

c IEEE.
The Transported Capacity (TC) is another performance index studied and

it is dened as:
7
i=1 Ni Ki
TC = . (7.11)
ASDref
In Figure 7.11 [9], the sum of the assigned ATM cells in all areas normalized
by the reference ASD value (in fact it is a maximum transport capacity
value) is shown. With the optimization over TT S , the transported capacity
is signicantly improved: over 6% more capacity in the fair case and near 8%
increase in the opportunistic design. This result shows that the increase in BO,
due to TT S optimization (Figure 7.10) eectively implies a TC increase. The
opportunistic design could reach the maximum TC value as ASD increases
(independently of the requests distribution), whereas the fair algorithm will
generally saturate at a lower value (between 0.62 and 0.69 in the studied case).
Fig. 7.11: Transported capacity. See reference [9]. Copyright 2006

c IEEE.
In what follows, the fairness issue is addressed by measuring the fairness

dierences between the solutions. This is done by using the fairness index
denition in [35]. For a given solution NAT M1 , ... , NAT M7 , new variables can
be dened as:
NAT M1 NAT M
y1 = , ... , y7 = 7 (7.12)
NAT M1 NAT M7
and the computation of the Fairness Index (FI) is as follows:
2

7
yi
i=1
FI = (7.13)
7
7 yi2
i=1

where NAT Mi is the most fair solution obtained with the fair algorithm
with optimal TT S (the fair solution is dened in this way).
FI is obtained for the following 2 solutions (see the results in Figure 7.12
[9]):
Solution 1: the fair solution with TT S = t1 .
Solution 2: the opportunistic solution with optimal TT S .
It is important to note that whereas solution 1 exhibits good fairness
performance, solution 2 reduces it signicantly.
Fig. 7.12: Fairness study. See reference [9]. Copyright 2006

c IEEE.
At the end of this sub-Section, the study of the occupation eciency for the
dierent signicant values of TT S (when only one area is requesting resources)
is addressed. Let us assume a very high demand to transmit, thus using the
maximum possible bandwidth in the proposed framework (see the results in
Table 7.2 [9]). Some TT S values achieve a better occupation eciency than
others, depending on which areas are considered as active. In particular, the
conguration TT S = 4t4 is the one that gives better results in the general case
(when all areas are active and the distribution p is totally unknown). This
conguration is the most robust choice in the max-min sense: knowing nothing
about the mapping of users to areas, the max-min robust design corresponds
to the one that gives the best (max) performance for the worst (min) possible
user distribution.
Bandwidth occupation eciency

area/TT S t1 3t7 3t6 3t5 3t4 4t7 4t6 4t5 5t7 4t4 5t6 5t5 6t7 6t6 6t5 7t7 7t6
1 0.98 0.80 0.80 0.74 0.61 0.61 0.55 0.55 0.49 0.98 0.86 0.86 0.74 0.74 0.74 0.61 0.61
2 0.82 0.66 0.66 0.61 0.51 0.51 0.92 0.92 0.82 0.82 0.72 0.72 0.61 0.92 0.92 0.77 0.77
3 0.65 0.53 0.53 0.98 0.82 0.82 0.74 0.74 0.65 0.98 0.86 0.86 0.74 0.74 0.98 0.82 0.82
4 0.98 0.80 0.80 0.74 0.92 0.92 0.83 0.83 0.74 0.98 0.86 0.86 0.74 0.92 0.92 0.77 0.77
5 0.87 0.71 0.71 0.98 0.82 0.82 0.74 0.98 0.87 0.87 0.76 0.95 0.82 0.82 0.98 0.82 0.82
6 0.82 0.66 1.00 0.92 0.77 0.77 0.92 0.92 0.82 0.82 0.89 0.89 0.77 0.92 0.92 0.77 0.89
7 0.76 0.93 0.93 0.86 0.72 0.95 0.86 0.86 0.95 0.95 0.83 0.83 0.86 0.86 0.86 0.83 0.83
mean 0.84 0.73 0.78 0.83 0.74 0.77 0.79 0.83 0.76 0.91 0.83 0.85 0.75 0.84 0.90 0.77 0.79
Table 7.2: Bandwidth occupation study. See reference [9]. Copyright 2006
c IEEE.
This Section has presented an alternative framework for bandwidth allo-

cation in the DVB-RCS scenario, which is compliant with the latest ETSI
technical specications. In the hierarchical bandwidth allocation procedure
deduced, the general fair allocation algorithm takes into account minimum
resource guaranteed. The timeslot selection has been optimized and its
implications analyzed, obtaining that the timeslot optimization is reasonably
independent of the scheduling policy (either implementing fair, opportunistic
or other strategies).
7.3.5 Dynamic bandwidth allocation for handover calls
In LEO and MEO satellite constellations, the handover problems can aect
the QoS of the connections. In [36], bandwidth for handover is dynamically
allocated, by calculating the possible handovers from neighboring beams, on
the basis of users location information. The reservation mechanism provides
a low handover blocking probability with respect to a xed guard channel
strategy. However, employing user location information seems not reasonable,
because updating locations would cause high processing load to the on-board
handover controller and increase the complexity of terminals. This method
seems only suitable for xed users.
In [37], the authors have introduced two dierent mobility models for
satellite networks. In the rst model, only the motion of satellites is taken
into account, whereas, in the second one, other motion components, like Earth
rotation and user movements, are considered. The key idea of the algorithm
is that, in order to prevent handover failure during a call, bandwidth will be
reserved in a particular number S of spot-beams that the call would handover
into.
In [38], a probabilistic resource reservation strategy for real-time services
was proposed. The sliding window concept is adopted to predict the nec-
essary amount of reserved bandwidth for a new call in its future handover
spot-beams. As for real-time services, a new call request is accepted if the
originated spot-beam has available bandwidth and resource reservation is
successful in future handover spot-beams. As for non real-time service, new
call requests are accepted if the originated spot-beam satises its maximum
required bandwidth.
In [6],[39], a selective look-ahead strategy is proposed where real-time
and non-real time service classes are dierently treated. Bandwidth allocation
only pertains to real-time connection handovers. To each accepted connection,
bandwidth allocation is performed in a look-ahead horizon of k cells along its
trajectory. This algorithm oers low call dropping probability, i.e., a reliable
management of call handovers of and acceptable call blocking probability for
new calls.
7.4 Conclusions
This Chapter has presented a set of dynamic bandwidth allocation techniques
and identied associated research topics. We can conclude this Chapter by
highlighting these two types of DBA problems and related techniques:
Handover-constrained techniques, mainly used for LEO satellites, where
the main problem is to acquire a resource among a number of dierent
satellites, since the communication lifetime is long enough to require a
number of handovers;
Bandwidth-constrained techniques, aecting mainly GEO systems, where
the main issue is to cope with the high delay-bandwidth product that
makes the reactive approaches unfeasible for delay-constrained trac
types.
The problem of multi-tier satellite systems, i.e., satellite systems using
a combination of multiple orbital systems, like GEO+LEO, has not been
considered, but it could be challenging, due to the multiple use of the
dierent techniques among the various tiers. This problem requires further
investigations as it involves also intra-tier and inter-tier routing schemes.
Most of the described DBA techniques are inherently satellite-dependent;
each satellite system should adapt or implement its own techniques in order
to maximize system eciency. A common theme is that optimizing eciency
does not always means maximizing the bandwidth occupancy, but it is
a concept more related to fullling the system goals in terms of QoS,

user satisfaction and, ultimately, system capacity to maximize the network
operators revenue. Hence, one of the possible approaches to further study
DBA techniques is to embed a cost-function into the DBA decision process,
in order to introduce an abstraction layer between the raw user bandwidth
requests and the actual bandwidth allocation decision algorithms.
Another topic that needs further investigation is represented by the fair-
ness of the proposed techniques. Most techniques that involve terminal-based
decisions (like in most DVB-RCS systems) can be heavily aected by fairness
issues in a multi-vendor and multi-algorithm environment, thus creating
serious issues in real-world deployments. At present, this problem is still an
open point and should be addressed either by allowing the centralized decision
process to take into account the dierent behaviors, or by dening some
fairness threshold that every user equipment implementation must comply
with. We must observe that the rst option is not viable in the long-term, as
it requires extra-work in the bandwidth allocation decision unit, along with
the knowledge of every implementation, and this is not always possible. The
second option requires the denition of precise fairness metrics and test suites
to certify the user terminal fairness.
The DBA implementation is therefore a key element for the ecient oper-
ation of many satellite systems. Design choices in DBA techniques can greatly
impact the overall system performance, and the evolution of appropriate
techniques and analysis methods will remain important research topics for
future generations of systems.
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Part III
Cross-Layer Techniques for

Satellite-Independent Layers
8
RESOURCE MANAGEMENT AND
NETWORK LAYER
Editors: Ulla Birnbacher1 , Wei Koong Chai2
Contributors: Paolo Barsocchi3 , Ulla Birnbacher1 , Wei Koong Chai2 ,

Antonio Cuevas4 , Franco Davoli5 , Alberto Gotta3 , Vincenzo Mancuso6 , Mario
Marchese5 , Maurizio Mongelli5 , Jose Ignacio Moreno Novella4 , Francesco
Potort`3 , Orestis Tsigkas7
1
TUG - Graz University of Technology, Austria
2
UniS - Centre for Communication Systems Research, University of Surrey,
UK
3
4
5
6
UToV - University of Rome Tor Vergata, Italy
7
8.1 Introduction
The Internet protocols have become the worldwide standard for network
and transport protocols and are increasingly used in satellite communication
244 Ulla Birnbacher, Wei Koong Chai
networks. Also traditional telecommunication and broadcast applications like

VoIP and video streaming are transported over the Internet, although it does
not support natively applications with tight QoS requirements. In satellite
communication networks, further challenges arise, as bandwidth resources
are limited and physical transmission time adds some more pressure on
delay constraints. Since resources are limited, the ecient assignment of
bandwidth to dierent data streams has always been an issue for satellite
communications. However, supporting QoS for IP-based applications results
in additional requirements for resource allocation. In order to provide QoS for
applications, several layers of the protocol stack of a satellite communication
system will need to be adapted or have to interact with each other in some
way. This Chapter will concentrate on dierent resource management schemes
at the MAC layer (layer 2) for supporting IP QoS (layer 3).
This Chapter begins with an overview of the current IP QoS frameworks
in Section 8.2. In Section 8.3, the discussion is focused on the interaction
of layer 2 and layer 3 in satellite environments for the support of IP QoS.
This Section ends with an example of implementation for a variant of one
of the most popular IP QoS frameworks. The following Section 8.4 provides
an in-depth work on achieving QoS requirements by a cross-layer approach
over SI-SAP. Section 8.5 looks into another aspect of resource management:
the QoS provisioning for terminals supporting dual network access (WiFi
and satellite). Implicit cross-layer design methodology is used in Section 8.6
for switched Ethernet over LEO satellite networks. Finally, this Chapter is
concluded in Section 8.7. In the studies carried out in this Chapter, Scenario
2 (i.e., GEO-based DVB-S/-RCS systems; see Chapter 1, Section 1.4) has been
adopted, except for the considerations made in Section 8.6, where Scenario 3
(i.e., LEO satellite) has been considered.
8.2 Overview IP QoS framework

In order to support the emerging Internet QoS, some QoS frameworks have
been proposed. These service models and mechanisms evolve the IP architec-
ture to support new service denitions that allow preferential or dierentiated
treatment to be provided to certain trac types. Integrated Services and
Dierentiated Services have already been introduced in Section 3.3, but are
discussed below in more detail with satellite networks in mind, including
Multiprotocol Label Switching (MPLS).
8.2.1 Integrated services
The Integrated Services (IntServ) model [1] requires resources, such as band-
width and buers, to be reserved a priori for a given trac ow to ensure
that the QoS requested by this trac ow is fullled. The IntServ model
includes additional components beyond those used in the best-eort model
Chapter 8: RESOURCE MANAGEMENT AND NETWORK LAYER 245
such as packet classiers, packet schedulers, admission control and signaling.

A packet classier is used to identify ows that have to receive a certain level
of service. A packet scheduler manages the service provided to dierent packet
ows to ensure that QoS commitments are met. Admission control is used to
determine whether a router has the necessary resources to accept a new ow.
Fig. 8.1: Implementation reference model for routers with IntServ [2].
A notable feature of the IntServ model is that it requires explicit signaling

of QoS requirements from end-systems to routers. The Resource Reserva-
tion Protocol (RSVP) [3] performs this signaling function and is a critical
component of IntServ. RSVP is a soft state signaling protocol. It supports
receiver-initiated establishment of resource reservations for both multicast
and unicast ows. Recently, RSVP has been modied and extended in several
ways to reserve resources for aggregation of ows, to set up MPLS explicit
label switched paths with QoS requirements, and to perform other signaling
functions within the Internet.
Two services have been dened under the IntServ model: guaranteed
service [4] and controlled-load service [5]. The guaranteed service provides
a rm quantitative bound on the end-to-end packet delay for a ow. This
is accomplished by controlling the queuing delay on network elements along
the data ow path. The guaranteed service model does not, however, pro-
vide bounds on jitter (inter-arrival times between consecutive packets). The
controlled-load service can be used for adaptive applications that can tolerate
some delay, but are sensitive to trac overload conditions. This type of
application typically operates satisfactorily when the network is lightly loaded,
but its performance degrades signicantly when the network is heavily loaded.
Controlled-load service, therefore, has been designed to provide approximately
the same service as best-eort service in a lightly loaded network regardless of

actual network conditions. Controlled-load service is described qualitatively
in that no target values of delay or loss are specied.
The IntServ architecture represents a fundamental change to the current
Internet architecture, which is based on the concept that all ow-related
state information should be in the end-systems. The main problem of the
IntServ model is scalability, especially in large public IP networks, which may
potentially have millions of active micro-ows concurrently in transit, since
the amount of state information maintained by network elements tends to
increase linearly with the number of micro-ows.
8.2.2 Dierentiated services
One of the primary motivations for Dierentiated Services (DiServ) [6] was
to devise alternative mechanisms for service dierentiation in the Internet that
mitigate the scalability issues encountered with the IntServ model. Scalable
mechanisms are deployed within the DiServ framework for the categorization
of trac ows into behavior aggregates, allowing each behavior aggregate to
be treated dierently, especially when there is shortage of resources such as
link bandwidth and buer space.
A DiServ eld in the IPv4 header has been dened. Such eld consists
of six bits of the part of the IP header, formerly known as TOS octet, and
it is used to indicate the forwarding treatment that a packet should receive
at a node. Within the DiServ framework, a number of Per-Hop Behavior
(PHB) groups have been also standardized. Using the PHBs, several classes
of services can be dened using dierent classication, policing, shaping, and
scheduling rules.
Conceptually, a DiServ domain consists of two types of routers, namely
core router and edge router. Core router resides within the domain and is
generally in charge of forwarding packets based on their respective DiServ
Code Point (DSCP). The edge router is located at the boundary of the network
domain which will either further connect to another domain (inter-domain)
or to end-users. It can be further categorized as ingress router which operates
on trac owing into the domain and egress router which operates on trac
exiting the domain.
In order for an end-user to receive DiServ from its Internet Service
Provider (ISP), it may be necessary for the user to have a Service Level
Agreement (SLA) with the ISP. An SLA may explicitly or implicitly specify a
Trac Conditioning Agreement (TCA), which denes classier rules, as well
as metering, marking, discarding, and shaping rules. Packets are classied,
and possibly policed and shaped at the ingress routers of a DiServ network
according to SLAs.
When a packet traverses the boundary between dierent DiServ domains,
the DiServ eld of the packet may be re-marked according to existing
agreements between the domains. DiServ allows only a nite number of
Fig. 8.2: DiServ network; (top) DiServ domain illustration; (bottom) logical view
of DiServ packet classier and trac conditioner.
service classes to be indicated by the DiServ eld.

The main advantage of the DiServ approach relative to the IntServ model
is scalability. Resources are allocated on a per-class basis and the amount of
state information in the routers is proportional to the number of classes rather
than to the number of application ows. A second advantage of the DiServ
approach is that sophisticated classication, marking, policing, and shaping
operations are only needed at the boundary of the network.
The DiServ control model essentially deals with trac management issues
on a per-hop basis and consists of a collection of micro-control mechanisms.
Other trac engineering capabilities, such as capacity management (including
routing control), are also required in order to deliver acceptable QoS in
DiServ networks.
At the current stage, the DiServ approach is still evolving. Two directions
of its development can be categorized: namely, absolute DiServ and relative
DiServ [7]. The absolute DiServ approach is the more traditional approach
detailed above. The newer and simpler approach is the relative DiServ,
whereby QoS assurances are provided relative to the ordering between several
trac or service classes rather than specifying the actual service level or
quality of each class. This approach is lightweight in nature, since it minimizes
computational cost as it does not require sophisticated mechanisms such as
admission control and resource reservation. As such, recently, it has gain in
popularity not only in terrestrial networks, but also in wireless [8],[9] and
satellite systems [10].
8.2.3 Multiprotocol Label Switching (MPLS)
MPLS is an advanced forwarding scheme, which extends the Internet routing

model and enhances packet forwarding and path control [11]. MPLS stands
for Multiprotocol Label Switching; note that the word multiprotocol is used,
because these techniques are applicable to any network layer protocol.

In conventional IP forwarding, when a packet of a connectionless network
layer protocol travels from one router to the next, each router makes an inde-
pendent forwarding decision for that packet. That is, each router re-examines
the packets header and independently chooses a next hop for the packet,
based on the results of the routing algorithm. Choosing the next hop can be
thought of as the composition of two functions. The rst function partitions
the entire set of possible packets into a set of Forwarding Equivalence Classes;
the second function maps each class to a next hop.
In the MPLS forwarding paradigm, the assignment of a particular packet
to a given class is done just once, at the ingress to an MPLS domain, by Label
Switching Routers (LSRs). As the forwarding decision is concerned, dierent
packets that get mapped into the same class are indistinguishable and will
follow the same path. The class to which the packet is assigned is encoded as
a short xed-length value known as a label. When a packet is forwarded to its
next hop, the label is sent along with it; that is, packets are labeled before
they are forwarded. At subsequent hops, there is no further analysis of the
packet network layer header. Rather, the label is used as an index into a table,
which species the next hop, and a new label. The old label is replaced by
the new label, and the packet is forwarded to its next hop. Most commonly, a
packet is assigned to a class based (completely or partially) on its destination
IP address. However, the label never is an encoding of that address.
A Label Switched Path (LSP) is the path between an ingress LSR and an
egress LSR through which a labeled packet traverses. The path of an explicit
LSP is dened at the originating (ingress) node of the LSP. MPLS can use a
signaling protocol such as RSVP or Label Distribution Protocol (LDP) to set
up LSPs. MPLS is a very powerful technology for Internet trac engineering
because it supports explicit LSPs, which allow constraint-based routing to be
implemented eciently in IP networks.
8.3 Resource management for IP QoS
Resource management schemes at MAC layer (layer 2) are essential in

supporting IP QoS (layer 3). The current IP QoS frameworks (i.e., IntServ
and DiServ) dene several service classes to cater for users with dierent
QoS requirements. The resource management scheme must be able to allocate
dynamically the available resources in an IP-based satellite network to achieve
the requirements of the dened service classes. This includes a mapping
scheme between layer 3 and layer 2, dynamic bandwidth allocation and
scheduling mechanisms.
In this Section, the specic scenario under consideration is a DiServ
satellite domain with DVB-RCS architecture for multimedia xed unicast
users. The choice of DiServ is mainly in view of the problems of the IntServ
framework, such as scalability and deployment. For DiServ, in general, there
are three types of PHBs being used: namely Expedited Forwarding (EF),
Assured Forwarding (AF) and Best Eort (BE). EF PHB caters for low loss,
low delay and low jitter services. The AF PHB consists of four AF classes,
where each class is allocated with dierent amounts of buer and bandwidth.
Hence, each subscriber with a specic Subscribed Information Rate will receive
assured performance for trac within such rate while excess trac may be
lost depending on the current load of the AF class. Finally, the BE PHB is
the same as the original best eort IP paradigm.
For the DVB-RCS architecture, there are four transmission capacity
allocation schemes; namely Continuous Rate Assignment (CRA), Rate Based
Dynamic Capacity (RBDC), Volume Based Dynamic Capacity (VBDC) and
Free Capacity Assignment (FCA). For the description of these dierent
resource allocation schemes, please refer to Chapter 1, sub-Section 1.4.3.
Before mapping the DiServ PHBs to DVB-RCS resource allocation
schemes, it is vital to note that the entire DiServ domain is assumed to
be properly dimensioned. This is because there is no one mapping scheme
that can achieve high eciency in all types of trac mixture. A particular
scheme, which performs well in one scenario, may perform poorly in another.
The network management and dimensioning problem is not within the scope
of this study.
Usually, EF PHB is used to transport non-delay tolerant application
tracs such as VoIP and video conferencing. To achieve the stringent QoS
requirements of this class of applications, the use of CRA in the MAC layer
is a must. However, considering system eciency, a minimal use of RBDC
combined with CRA is plausible. The entire DiServ domain has to be
properly dimensioned as noted above. For example, if a very high trac
percentage is of the EF type, then the satellite bandwidth will be quickly
consumed with all the slots being reserved with CRA, thus causing high
blocking and drop rate. As for AF PHB, the combined use of RBDC and
VBDC is proposed with RBDC as the main resource provider. Under low load,
packets belonging to each class of AF will receive similar treatment. However,
to dierentiate between the AF classes, a dierent maximum RBDC value
(i.e., maximum bit-rate that can be allocated with RBDC) can be dened so
that the higher AF class will receive better treatment. If the request is higher
than the maximum RBDC, the users can still request for VBDC resources.
For BE trac, the use of VBDC and FCA is proposed.
8.3.1 Relative DiServ by MAC Scheduling
An alternative scenario on resource management schemes at MAC layer

(layer 2) to support IP QoS (layer 3) is the work on attempting to realize
relative service dierentiation in a Bandwidth on Demand (BoD) satellite IP
network. The Proportional Dierentiated Service (PDS) [7] model is one of
the most recent developments of DiServ in the direction of relative service
dierentiation. It strives to strike a balance between the strict QoS guarantee
of IntServ and the scalability of DiServ. Similarly to DiServ, PDS segregates

tracs into a nite number of service classes. However, it does not provide
them with absolute QoS guarantees. Instead, it controls the performance gap
between each pair of service classes, i.e., quantitative relative dierentiation
amongst the supported classes.
Formally, the PDS model requires
i ri
= ; i, j {1 . . . N } . (8.1)
j rj
where each class is associated with a Dierentiation Parameter (DP), ri ,

and i is the performance metric of interest for class i, e.g., throughput, packet
loss or queuing delay. N is the total number of supported service classes in
the network.
In this Section, classes are numbered in decreasing priority order, i.e.,
the lower the class index, the better the service provided to it. All DPs are
normalized with reference to the highest priority class (= 1):
0 < rN < rN 1 < ... < r2 < r1 = 1
This Section demonstrates how such a model can be realized in an IP-based

broadband multimedia BoD GEO satellite network with resource allocation
mechanisms analogous to the DVB-RCS system standard [12].
Figure 8.3 [10] illustrates the main nodes of the network architecture:
Fig. 8.3: Reference satellite system, resembling the DVB-RCS architecture. See
c IEEE.
Satellite(s): The satellite is assumed to be equipped with On-Board Proces-

sor (OBP) and the scheduler is located on-board.
Trac Gateway (GW): In line with the DVB-RCS denition, GWs are
included to provide interactive services to networks (e.g., Internet) and
service providers (e.g., databases, interactive games, etc.).
Satellite Terminal (ST): STs represent the users. They may represent one
(residential) or more users (collective).
Time Division Multiple Access (TDMA) is used for the forward path
whereas on the return path, Multi Frequency - TDMA (MF-TDMA) is
assumed. In an MF-TDMA frame, the basic unit of the link capacity is the
Time Slot (TS) with multiple TSs grouped in TDMA frames along several
frequency carriers. In this Section, xed MF-TDMA frame is considered
whereby the bandwidth and duration of successive TSs is static. For more
details on MF-TDMA characteristics, please refer to Chapter 1.
The BoD scheme used in this Section is derived from [13]. It is a cyclic
procedure between two stages: the resource request estimation stage and the
resource allocation stage. It involves the BoD entity located at the ST and
BoD scheduler located onboard the satellite. The BoD entity handles all
packets of the same class which are stored in the same queue, i.e., there
will be x BoD entities in an ST if this ST supports x classes. In the resource
request estimation stage, the BoD entities (i.e., STs) periodically compute and
send Slot Requests (SRs) to the BoD scheduler, when there are new packet
arrivals at their queues. In the resource allocation stage, upon reception of
SRs, the BoD scheduler allocates TSs to each requesting BoD entity based on
a certain scheduling discipline and policies dened by the network operator. It
then constructs and broadcasts the Terminal Burst Time Plan (TBTP) that
contains all the resource allocation information to the BoD entities. Figure
8.4 [10] gives the BoD timing diagram, which also describes the basic tasks
involved.
Due to the unique characteristics of satellite networks, the realization of
such framework is very dierent from those solutions provided for terrestrial
and wireless systems. For terrestrial wired networks, the scheduler only needs
to schedule the departure of each contending packet locally within a router.
In wireless and satellite domain, the access to the transmission medium is
often controlled in a distributed manner by a MAC protocol. Hence, packets
from one node may contend with packets from other nodes. This leads to
the consideration of using layer 2 scheduling to realize the model instead of
purely depending on layer 3. Based on the layer 3 QoS classes, the MAC layer
scheduler will decide how best to schedule the packets in order to achieve the
QoS required.
Moreover, there are several fundamental architectural and environmental
dierences between terrestrial wireless networks and satellite networks sup-
porting dynamic bandwidth allocation mechanisms. Firstly, for a BoD-based
satellite architecture, resource has to be requested by the STs before they can
Fig. 8.4: BoD timing diagram. See reference [10]. Copyright 2005
c IEEE.
make use of it, so that the scheduler ends up scheduling requests for resource
rather than packets. Secondly, there is a non-negligible propagation delay
between the STs and the scheduler that may, depending on the access control
algorithm, inate the waiting time of a packet in the ST queue. The impact
of this semi-constant delay has to be taken into account by the scheduler in
providing relative service dierentiation.
The Satellite Waiting Time Priority (SWTP) scheduler is a satellite
adaptation of the Waiting Time Priority scheduler [7], proposed by Kleinrock
in [14]. SWTP schedules SRs from BoD entities rather than individual packets.
SWTP has been shown to be able to provide proportional queuing delay to
several classes of MAC frames in the context of BoD environment. Its main
elements are as follow.
1. Resource request. Formally, if Qm i is the set of newly arrived packets at
the i -th queue of BoD entity m, i.e., packets that came within the last
resource allocation period, q the set cardinality, and j the arrival time
of packet j, 1j q, indexed in increasing order of arrival times, then the
BoD entity m computes at time t the SR timestamp tsm i , according to
the arrival time of the last packet that arrived in the queue during the
last resource allocation period, namely: tsmi = t q .
2. Resource allocation: the BoD scheduler computes the priority of each SR.
The priority Pim (k ), assigned to S Rim in the k -th resource allocation
period is
" #
Pim (k) = ri wiSR (k) + (8.2)
where accounts for the propagation delay of TBTP and the processing
delay of BoD entities, while wiSR (k) = t tsm m
i and tsi is the timestamp
information encoded in each SR. Finally, ri denotes here the Delay
Dierentiation Parameter (DDP): each one of the N MAC classes is

attached with a specic ri , 1i N. At each allocation period, the SWTP
allocates TSs by considering requests in decreasing priority order: requests
are fully satised as long as they do not exceed the available capacity.
All unsatised requests will be buered for the next allocation period.
At the next allocation period, the priorities of the buered SRs will be
recalculated to account for the additional waiting time of SRs at the
scheduler.
The setup of the simulations is as follow. The capacities for all links are
congured to be 2048 kbit/s. Unless explicitly stated otherwise, the network is
set to have DDPs: 1, 1/2, 1/4, 1/8. By this setting where the dierentiation is
exactly half of the next adjacent class, the ideal performance ratio according
to the PDS model will be 0.5. The IP packet size used is 500 bytes, while MAC
frames are of 48 bytes with additional 5 bytes due to header (ATM case).
Figure 8.5 shows the queuing delay for each service class, while Figure 8.6
presents the corresponding delay ratios under constant bit-rate trac [10]. The
ideal value for the ratios is 0.5 for all cases. From the plotted results, it is clear
that the SWTP scheduler can indeed emulate closely the PDS model. Since
the PDS model requires that the spacing between any two service classes
strictly adheres to the ratio of the DDPs for the specied service classes, the
scheduler should not be dependent on the trac distribution between service
classes. Figure 8.7 [10] shows the result of this test at a utilization of 95%:
the achieved ratios are very near to the ideal value of 0.5.
Fig. 8.5: Queuing delay of dierent service classes following the specied spacing
of the model. See reference [10]. Copyright 2005
c IEEE.
Fig. 8.6: Delay ratios achieved that are close to the ideal delay ratios. See reference
[10]. Copyright 2005
c IEEE.
Fig. 8.7: SWTP emulating the PDS in dierent load distributions with all values
achieved close to the ideal value. See reference [10]. Copyright 2005
c IEEE.
To illustrate the capability of SWTP in accurately controlling the spacing

between dierent service classes, three sets of DDPs have been dened below.
Set A: [1, 1/2, 1/4, 1/8]

Set B: [1, 1/2, 1/3, 1/4]
Set C: [1, 1/4, 1/5, 1/6].
Simulations with utilization of 95% have been conducted based on these

DDP sets and the results given in Figure 8.8 [10] show the normalized ratios
of all the three cases, where the normalized ratios are dened as the achieved
performance ratios divided by the respective ideal ratios. With the ideal value
as 1.0, it can be concluded that SWTP is indeed able to control the class
spacing. However, due to the long propagation delay, the spacing between the
highest and lowest DDP should not be too large to ensure reasonable delay
for the lowest class.
Fig. 8.8: SWTP with 3 sets of DDPs: all normalized delay ratios are close to the
ideal value. See reference [10]. Copyright 2005
c IEEE.
The behavior of SWTP in short timescale is investigated to ensure that

the predictability requirement of the PDS model is satised. Figure 8.9 [10]
shows the individual packet delays upon departure in a four-class scenario for
a period of 100 ms. The graph shows that SWTP can consistently provide the
appropriate spacing for the service classes.
Fig. 8.9: Short time scale behavior of SWTP showing its predictability property.
See reference [10]. Copyright 2005
c IEEE.
8.4 QoS mapping over satellite-independent service

access point
In what follows, we are specically concerned with the cross-layer interac-

tion between the network and the MAC layer, in order to preserve QoS
requirements, or, in more precise terms, to operate a mapping between
the QoS mechanisms operating at the two layers. Within a more general
view, with reference to the ETSI Broadband Satellite Multimedia (BSM)
protocol architecture [15],[16], we might refer to the inter-working between
the Satellite-Independent (SI) and the Satellite-Dependent (SD) architectural
components at the SI-SAP (Satellite-Independent - Service Access Point), by
taking into account both the change in encapsulation format and the trac
aggregation in the passage from SI to SD queues. Note that the ETSI BSM
architecture has been described in Chapter 1, Section 1.5.
Cross-layer RRM problems, involving network and MAC layers, have been
extensively considered in [17]-[19]. Reference [20] also provides guidelines and
architectural details. In particular, in [17]-[19] Dynamic Bandwidth Allocation
(DBA) is applied by computing bandwidth requests for each Earth stations
DiServ queue, which are passed to a centralized scheduler, typically residing
in a Master Control Station (MCS). The latter assigns the bandwidth pro-
portionally to the requests received; the remaining capacity is assigned on a
free basis. Such scheme has been called Combined Free/Demand Assignment
Multiple Access (CF/DAMA).
In a similar context, the problem of QoS mapping between adjacent layers
has been recently treated in [21]-[23]. Rather than considering specically the
network and the MAC layers, the problem is posed in the more general ETSI
BSM scenario mentioned above. In the presence of IP DiServ queues at the
higher layer, the problem consists in dynamically assigning the bandwidth
(service rate) to each SD queue, so that the performance required at the
IP layer is guaranteed. By considering a uid model and the loss volume as
the performance indicator of interest, the Innitesimal Perturbation Analysis
(IPA) technique of Cassandras et al. [24] (already mentioned in Chapter 7 in
a dierent scenario) is applied in order to maintain on-line the equalization
between the loss volumes at the two dierent layers (by assuming that the
resource allocation at the SI layer is capable of satisfying the requirements).
In doing so, both trac and fading variations are taken into account. Further
details on the application of the IPA technique are provided in sub-Section
8.4.2.
8.4.1 Model-based techniques for QoS mapping and support
Earth stations use reservation mechanisms (bandwidth requests) to transmit

their trac ows (voice or MPEG video, bandwidth reserved for DiServ
aggregates, MPLS pipes, etc.), which may be carried with priority at the
satellite link level within some specic DVB service classes. The control
process works upon requests for bandwidth allocation, which can be satised
within a Round Trip Time (RTT) for the request to reach the scheduler
and the response to be received (referred to as DBA cycle time in [17]).
Hence, whenever trac ows are characterized by a relatively low burstiness
(e.g., the peak-to-average ratio of their rates is close to 1), simple DAMA
schemes (e.g., VBDC) can be employed to manage the trac of Earth stations
[19]. The bandwidth allocation can be controlled in this case by means of
CAC functions. When burstiness is higher, DBA is applied by computing
bandwidth requests (on the basis of a model) for each Earth stations DiServ
queue, which are passed to a centralized scheduler that assigns the bandwidth
proportionally to the requests received; the remaining capacity is assigned on
a free basis, according to CF/DAMA. Various trac models have been used to
represent the burst-level behavior of real-time Variable Bit Rate (VBR) trac;
among them, we can consider voice with silence detection and VBR-encoded
MPEG video. In this case, two control functionalities at dierent time scales
should be employed, namely, CAC at the call level and DBA at the burst
level, to guarantee at the same time both a specied degree of QoS and an
ecient bandwidth utilization.
In [17], models capturing both Short Range Dependent (SRD) and Long
Range Dependent (LRD) behaviors have been used to represent the arrival
processes of trac aggregates to the User Terminal (UT) IP queues in a
DiServ scenario. They are based on Markov-Modulated Poisson Processes
(MMPP) and Pareto-Modulated Poisson Processes (PMPP), giving rise to
MMPP/G/1 and PMPP/G/1 queuing systems, respectively. The adopted
service-dependent QoS metric is the probability that the length of each
service queue exceeds a given threshold; we consider the constraint that this
probability must be kept below a specied value, beyond which the station is
considered in outage. The scheduling of the MAC queues must be such that
this constraint is fullled for the IP-level queues (i.e., those corresponding to
EF, AF and BE services within a given Earth station). No fading variations
are taken into account, but, as noted in [17], the eect of fade countermeasures
might be included as a reduction in the available uplink bandwidth. Note that
if the state of the sources can be assumed to change more slowly than the
DBA cycle time, within which the allocated bandwidth remains constant, the
queuing behavior in these intervals can be approximated by a much simpler
M/D/1 system.
8.4.2 A measurement-based approach for QoS mapping

and support
The work done in [21]-[23] takes a dierent look at the QoS mapping and
support problem, by disregarding the use of models, but rather relying on
measurement-based optimization techniques. This framework is that of ETSI-
BSM [15],[16] (let us consider for example the RBDC scheme). In such a
context, two basic facts are taken into account: the change of information
unit (e.g., from IP to IP-over-DVB) and the heterogeneous trac aggregation,
since, for hardware implementation constraints, the number of available SD
queues can be lower than that of SI queues (see also Chapter 1, sub-Section
1.4.3). Figure 8.10, taken from [21], reports and example.
The problem is then how much bandwidth must be assigned to each
SD queue, so that the SI IP-based SLA (i.e., the performance expected)
is guaranteed. In doing this, the eect of fading on the satellite channel is
also taken into account. As in other works (see, e.g., [25]), when the fade
countermeasure in use is modulation and coding rate adaptation, the eect
of fading is modeled as a reduction in the bandwidth (i.e., the service rate)
eectively seen by a layer 2 trac buer.
IP Packet Loss Probability (PLP) is one of the SLA performance metrics
considered in [23] (the other being IP Packet Average Delay). However, we
concentrate here on PLP. The mathematical framework is based on Stochastic
Fluid Models (SFM) of the SI-SAP trac buers [24],[26]. N SI queues and,
without loss of generality, one single SD queue are considered for the analytical
formulation (Figure 8.11).
Let iSI (t) be the input process entering the i -th trac buer at the SI
layer at time t, i = 1, ... , N. After entering one single buer [with service
rate iSI (t)] at the SI layer, each iSI (t) process is conveyed to a single SD
buer$ [whose service% rate is SD (t)] at the SD layer after a format change.
i SI
LV iSI (t) , iSI (t) denotes the loss volume of the i -th IP buer according
to the bandwidth allocation iSI (t).
Let SD (t) be the input process of the buer at the SD layer at time
t. The SD (t) process derives from the output processes of the SI buers.
Fig. 8.10: Queuing at the SI-SAP interface: satellite-independent (DiServ) over

satellite-dependent layer (ATM). See reference [21]. Copyright 2005
c IEEE.
Fig. 8.11: Stochastic processes and buer set for the envisaged SI-SAP queuing
model.
The loss
$ volume of the i -th %trac class within the SD buer is indicated by
LV SD (t) , SD (t) (t) . It is a function of the following elements: the SD
i SD
input process SD (t), the fading process (t) and the SD bandwidth allocation
SD (t). It is remarkable that i LSD
V () cannot be obtained in closed-from.
The problem reveals to be the equalization of the QoS measured at the
dierent layers of the protocol stack (i.e., SI and SD):
QoS Mapping Optimization (QoSMO) Problem: nd the optimal

bandwidth allocation, Opt SD (t), so that the cost function J(, SD (t)) is
minimized:
Opt SD
(t) = arg min J(, SD (t)); J(, SD (t)) = E LV (, SD (t))
SD (t)
(8.3)

N
$ %2
LV (, SD (t)) = V (i (t), i (t)) LV ( (t), SD (t) (t)) .
i
LSI SI SI i SD SD
i=1
In (8.3), denotes a sample path of the system, i.e., a realization of the

stochastic processes involved in the problem (coming from quantities (t),
iSI (t), i = 1, ... , N, SD (t)). Note that the cost function [see the second
row in (8.3)] weighs the sum of the quadratic deviations of the loss volumes
at the two layers, over all trac classes associated with SI queues.
This QoSMO problem is very complex to be solved. Two approaches are
considered below; one is based on the equivalent bandwidth concept and the
other is based on IPA.
Traditionally, equivalent bandwidth techniques are based on the statistical
characterization of the trac generated by users applications. The only
simply applicable statistics, useful for the SD rate provision, are the mean
(m) and the standard deviation () of the SD process. Hence, a popular
equivalent bandwidth technique, actually applicable in this context, is ruled
by (8.4) below [27]. Let us consider the following notations: k = 1, 2, ... the
time instants of the SD rate reallocations, mSD (k) and SD (k) the mean
and the standard deviation, respectively, of the SD input process measured
over the time interval [k, k +1]. Therefore, the bandwidth provision SD (k +1)
at the SD layer, assigned for the time interval [k +1, k +2], may be computed
as:
SD (k + 1) = mSD (k) + a SD (k) (8.4)

&
where a = 2 ln() ln(2) and represents the upper bound on the
allowed PLP. Such allocation method is called Equivalent Bandwidth approach
(EqB) in what follows.
In [23], another measurement-based equivalent bandwidth algorithm is
proposed that can face:
Heterogeneity of the QoS requests in the aggregated trunk;

Change of encapsulation format;
Fading counteraction;
No knowledge of SD input processs statistical properties;
No knowledge of SD buer size.
To match these requirements, the derivative of the cost function LV ()
is used:
LV (, SD ) N
i LSD SD $ %
V ( ) i SD SD
SD
= 2 SD
LV ( ) i LSI SI
V (i ) . (8.5)
i=1

i LSD ( SD )
Using IPA (see, e.g., [24],[26] and references therein), each V
SD
component can be obtained in real-time only on the basis of some trac
samples acquired during the system evolution. Let [k, k +1] be the time interval
between two consecutive SD bandwidth reallocations. The interval of time in
which the buer is not empty are dened as busy periods. The derivative
estimation is computed at the end of the decision epoch [k, k +1] as follows:
" SD # '' Nki " SD # ''

i LSD
V ' i LSD
k, '
' = (k) ' (8.6)
SD ' SD SD ' SD
(k) =1 (k)
'
i SD SD '
Lk, ( ) ' i k SD i k SD
' = (k) (k) (8.7)
SD ' SD
(k)
where i LSD
k, (
SD
) is the -th contribution to the SD loss volume of the
i -th trac class for each busy period Bk within the decision interval [k, k +1];
k is the starting point of Bk ; k is the instant of time when the last loss
occurs during Bk ; Nki is the number of busy periods within the interval [k,
k +1] for service class i. It must be noted that SD (k ) represents the SD
bandwidth reduction due to fading within the time interval [k, k +1] (i.e.,
SD (k) = SD (k) (k), where (k ) represents the bandwidth reduction seen
at the SD layer, due to redundancy applied at the physical layer to counteract
channel degradation).
The proposed optimization algorithm is based on the gradient method,
whose descent step is ruled by (8.8):
" SD # ''
LV , '
SD (k + 1) = SD (k) k ' ; k = 1, 2, ... (8.8)
SD ' SD
(k)
In (8.8), k denotes the gradient step size and k the reallocation time in-
stant. This method is called Reference Chaser Bandwidth Controller (RCBC).
8.4.3 Performance evaluation and discussion
These rate control mechanisms (i.e., RCBC and EqB) have been investigated
through simulations [21],[23]. An ad-hoc C++ simulator has been developed
for the SI-SAP environment described above, considering a general satellite
system. In what follows, for the sake of simplicity, only the trac aggregation
problem is faced by assuming no channel degradation over the satellite
channel.
The case considered is that of two SI trac buers. The rst one con-
veys the trac of 30 VoIP sources. Each VoIP source is modeled as an
exponentially-modulated on-o process, with mean on and o times equal
to 1.008 s and 1.587 s, respectively. All VoIP connections have peak rate of
64 kbit/s. The IP packet size is 80 bytes. The SI service rate for VoIP assures
an SLA target PLP below 102 (SI VoIP buer size is 30 IP packets). The
second buer is dedicated to a video service. Jurassic Park I video trace,
taken from the Web site referenced in [28], is used. The SI rate allocation for
video (also measured through simulations), is 350 kbit/s. It assures a PLP
= 103 , which is the target SLA for video (the SI video buer size is 10,500
bytes). Both outputs of the SI buers are conveyed towards a single queue at
the SD layer. DVB encapsulation (header 4 bytes, payload 184 bytes) of the
IP packets through the LLC/SNAP (overhead 8 bytes) is implemented in this
case. The SD buer size is 300 DVB cells.
In Figure 8.12 (rstly presented in [21]), the SD bandwidth provision
produced by RCBC is compared with EqB. The loss probability bound for
EqB is set to 103 , being the most stringent PLP constraint imposed at the SI
level. The time interval between two consecutive SD bandwidth reallocations
is denoted by TRCBC and TEqB , for RCBC and EqB respectively. Note that
in the following graphs, for the sake of simplicity, T denotes TRCBC (TEqB )
in the RCBC (EqB) case.
TRCBC is xed to 7 minutes, while TEqB is set to the following values:
{TRCBC 1/3, TRCBC 1/2, TRCBC , TRCBC 2, TRCBC 4}
in dierent tests in order to highlight the possible inaccuracy introduced

by the real-time computation of the EqB statistics using dierent time scales.
According to Figure 8.12, RCBC captures the bandwidth needs of the SD
layer in a single reallocation step. Whereas, EqB produces strong oscillations
in the SD rate assignment. It is also clear from Figure 8.12 that the IPA-based
estimation (8.5) is more robust than the on-line estimation of mSD and SD .
The IPA sensitivity estimation drives RCBC toward the optimal solution of
the QoSMO problem.
The SD buers video PLP, averaged over the entire simulation horizon,
is shown in Figure 8.13 (taken from [21]). The performance of RCBC,
referenced to as SD layer RCBC is very satisfying: actually, the RCBC
video PLP is 7.56104 . A result below threshold has been measured for
Fig. 8.12: Aggregation of VoIP and Video. SD allocations. RCBC versus EqB. See
c IEEE.
Fig. 8.13: Aggregation of VoIP and Video. Video PLP. See reference [21]. Copyright
2005
c IEEE.
EqB only for frequent reallocations (TEqB = TRCBC 1/3 = 2.33 minutes). The
corresponding bandwidth allocations, averaged over the simulation duration,
are shown in Figure 8.14 (taken from [21]). RCBC not only allows saving
bandwidth compared to the SD layer EqB T = 2.33 min strategy, but oers
a performance comparable to the other EqB cases, whose oered PLP is far
from the SI threshold. In brief, RCBC nds the optimal operation point of
the system, namely, the minimum SD bandwidth provision needed to track
the SI QoS thresholds.
Fig. 8.14: Aggregation of VoIP and Video. Average SD bandwidth provision. See
c IEEE.
8.5 QoS provisioning for terminals supporting dual

network access - satellite and terrestrial
When terminals support dual network access -satellite and terrestrial (WLAN,
UMTS, etc.) links- it is quite critical to select the appropriate network for
each application, depending on both the resources available and the kind
of application involved. In some instances (such as real-time tele-operation),
it is not only a matter of user satisfaction, but also of satisfying critical
service goals. For example, the QoS provision may be related to the deadline
fulllment: violating a deadline may cause a sea farm hitting the sea bottom
or a remote probe bump into a rock.
This Section provides an analysis on relevant technologies in this context
and focuses on QoS frameworks to support terminal mobility between satellite,
wireless, and terrestrial networks. In particular, we analyze the problem of the
multiple access to dierent networks (which includes satellite, wireless, and
terrestrial networks) in order to support more than one access network at the
same time. In such a context, the focus is on network selection based on QoS
parameters. We work on QoS parameter identication at layer 2 for selected
applications as well as IP-oriented solutions for network mobility and network
selection. Let us consider two specic topics:
Redundant codes in hybrid networks and

Mechanisms for error recovery in WiFi access points.
Redundant codes in hybrid networks

Hybrid networks consisting of satellite links and mobile ad hoc networks
present a series of challenges due to dierent packet-loss patterns, delay, and,
usually, scarce available bandwidth. In this scenario, redundant encoding, in
the form of Forward Erasure Correction (FZC) codes [29],[30], can provide
an eective protection against losses in multicast videoconferencing and video
streaming applications. The use of ecient encoding techniques can provide
further reduction on bandwidth requirements.
A real test-bed based on a remote video streaming server interconnected
via a GEO-satellite pipe to a local WLAN (both 11 Mbit/s and 5 Mbit/s
cases have been considered, according to IEEE 802.11b) is presented in [31],
by adopting the multicast network protocol. The satellite pipe is based on
the commercial Skyplex network [32] that operates in the Ka band with
the Hotbird 6 transponder. The developed platform, described in [33], is
shown in Figure 8.15. The purpose is to provide users with a low-cost,
high-availability platform for performing experiments with IP packets over
the Skyplex platform. Such devices have been also used to experiment the
FZC encoding.
Fig. 8.15: Test-bed platform architecture.
The obtained experimental measurements show the performance of FZC

codes based on Vandermonde matrix [34], for multicast video streaming
applications. Basically, k blocks of source data are encoded to produce n
blocks of encoded data (with n > k ), such that any subset of k -encoded
blocks suces to reconstruct the k -block source data. Considering the real
implementation, the encoder works at the transport layer by fetching k

information packets from the video stream and then transmitting k +l UDP
packets (k of information + l of redundancy) towards the receiving host. The
encoder adds a preamble of 4 bytes for the sequence number, which is then
cancelled by the decoder. The decoder fetches k of the k +l packets per block
and recovers the information, provided that no more than l packets are lost in
a single block of packets. The receiver then feeds the MPEG-4 decoder with
the received stream. Dierent MPEG-4 movies have been broadcast towards
the terrestrial WLAN through the satellite channel by using a standard video
decoder and home-made FZC encoding/decoding software. The authors in
[31] used this software to work between the application layer and the UDP
transport layer; however, this software can also be used between layer 2
and 3.
The performance evaluation has been based both on subjective perception
of QoS and on objective parameters of QoS. The used subjective assessment
has been the perceptual quality, called the Mean Opinion Score (MOS), which
requires several observers and many tests in order to provide a reasonable
statistical spread of results (1 ). The reference measure used for an objective
video quality assessment has been the Peak Signal to Noise Ratio (PSNR),
calculated on the dierence signal between the original error-free sequence and
the received decoded sequence. Other general objective parameters of QoS,
such as packet delivery delay and packet loss, have been considered in [31]. In
what follows the numerical results are presented.
Packet Loss - The experiment in un-coded mode (no FZC) shows that
packet loss can be reduced from 13% to about 6% by changing the transmission
rate from 11 to 5.5 Mbit/s (see Table 8.1). In this case, the channel occupancy
increases by 56%. Interestingly, we can note that with FZC and 200/100
coding ratio the packet loss is almost zero.
Coding ratio Packet Loss Residual Bandwidth [Mbit/s]

Uncoded 11 Mbit/s 12.98% 4.77
110/100 8.17% 4.67
Uncoded 5.5 Mbit/s 5.39% 4.24
120/100 3.25% 4.58
130/100 0.83% 4.48
200/100 0% 3.83
Table 8.1: Packet loss and residual bandwidth after FZC encoding.
1
According to ITU-R Recommendation 500-5, MOS values are: imperceptible
(5), perceptible but not annoying (4), slightly annoying (3), annoying (2), very
annoying (1).
Packet delivery delay - The maximum packet delivery delay is evaluated

as the time necessary to recover all the information when a number of packets
equal to the redundancy packets are lost. Table 8.2 shows the packet delivery
delay versus the coding ratio.
Coding ratio Max [ms] Mean [ms] Var.[ms2 ]

110/100 105.6 52.149 0.769
120/100 115.2 54.99 0.792
130/100 124.8 52.637 0.805
200/100 192 53.805 1.675
Table 8.2: Maximum, mean and variance of delivery delay. See reference [31].
Copyright 2005
c IEEE.
Mean Opinion Score (MOS) - Thirty persons have answered to three

quality questions (overall, video, and audio quality) for each considered coding
ratio; MOSs have been calculated. The percentages of people, who have
considered the video acceptable, are shown in Table 8.3.
Uncoded 110/100 Uncoded 120/100 130/100 200/100

11 Mbit/s 11 Mbit/s 5.5 Mbit/s 11 Mbit/s 11 Mbit/s 11 Mbit/s
7.7% 15.4% 26.9% 34.6% 100% 100%
Table 8.3: Acceptability of received video. See reference [31]. Copyright 2005
c
IEEE.
PSNR - The PSNR-based video quality metric (henceforth denoted as

VQMP - Peak Video Quality Measurement) uses a form of the logistics
function that is recommended in references [35],[36] and evaluates the mean
of how much transmitted frames dier from the original ones. Sixty seconds of
the transmitted video have been compared with the received video to evaluate
the VQMP parameter. Results are presented in Figure 8.16, where VQMP is
compared for dierent coding ratio values.
Error recovery in WiFi access points
The second topic of our study on the interconnection of WLAN and satellite
networks deal with some mechanisms for error recovery when a Fast HandOver
(FHO) occurs between dierent IEEE 802.11b Access Points (APs). Fast
handover techniques using paradigms like make-before-break or bi-casting
reduce the L3 handover time to extremely short delays that are acceptable
for all the applications [37]. However, in layer 2 (L2), the handover time is
Fig. 8.16: Peak Video Quality Measurement.
very high for some technologies. So, when doing an intra-technology handover
(e.g., the interface changes its AP and/or channel), an unacceptable disruption
may occur. For instance, in IEEE 802.11b (WiFi) an FHO procedure can take
from about 500 ms in a standard mode, to less than 10 ms in an optimized
mode, where the number of frequencies, scanned in order to establish the new
communication, is sensibly reduced [38]. During such a time, all transmitted
information can be lost. In such a context, the adoption of robust FZC codes
can permit to recover totally the lost information. This procedure may cost
in terms of bandwidth utilization and computational complexity, due to the
generation of the redundant information. To minimize these facts we propose
to use FZC in the last hop, i.e., between the Mobile Node (MN) and its Access
Router (AR). In what follows, AP and AR terms are used interchangeably.
The Core Network (CN) will then not need extra computational power and
use more bandwidth in its access link.
During the L2 disruption, both the ARs and the MN can stop sending pack-
ets and buer them and send them when the connectivity is re-established.
Indeed, during an FHO, the MNs have means to predict that there is going
to be an L2 disruption. For instance, they may know the instant in which
the FHO takes place or nishes or its duration, but, perhaps, not with
accuracy or some parameters may be unknown. Buering by its own is not
a perfect solution; hence, we propose to complement it with FEC techniques
of the FZC type. Our solution consists in adding (or increasing) the FEC
used between MN and ARs during the predicted FHO duration. Table 8.4
permits to understand the advantage of FEC techniques with respect to
pure buering. This table depicts an FHO (beginning and end instants of
the L2 disruption are indicated) and also shows when the disruption actually
happens. If we use buering, the communication is cut between the disruption
indications and then the buered packets are sent. But using FEC, MN and
ARs continue sending packets and the communication is cut only during the
actual disruption. When this disruption ends, lost packets may be recovered
by exploiting FEC capabilities. When L2 disruption end is indicated, FEC
can be stopped. Of course the advantage of FEC versus buering depends on
how big is the shift between disruption indication and actual disruption.
TIME
Communication Normal L2 L2 L2 L2 Normal
status (mobile Tx disruption disruption disruption disruption Tx
node) indication end
indication
Mobile node Buering Buering Buering Buering
behavior
Correspondent pkt Rx pkt Rx
node behavior
Mobile node FEC FEC FEC FEC
behavior using
FEC techniques
Correspondent pkt Rx pkt Rx pkt Rx pkt Rx
node behavior
when the MN
uses FEC during
handover
Table 8.4: Buering versus FEC techniques while the transmitter node undergoes
an FHO.
A rst problem to solve is how the MN will indicate its own software and
the AR that it is going to perform an FHO. That may depend on the actual L2
technology and even, in some technologies, the AR will be the one telling the
MN that it has to move. In WiFi, when the MN detects that the signal level
of its current AR decreases bellow a threshold, it initiates the FHO procedure
(scanning new channels in new ARs and then doing the FHO to the selected
channels). This issue can trigger two actions in the MN: it starts doing FEC
and tells the AR to do so as well.
The second issue to solve is to determine the ideal amount of redundancy
in the FEC technique. Hence, we must calculate the maximum number of
packets lost during L2 disruption (we must estimate the total disruption time
and the packet rate). This number of packets is the redundancy that must
be included in the FEC. In Table 8.5 this disruption time (shadowed parts)
corresponds to 3 packets and thus 3-packet redundancy is added (packets 3, 4
and 5). Note that information packets are labeled with a letter and redundancy
packets with a number.
Also the buering needed at the receiver (both MN side and the AR one,
depending on the direction of transmission) must be determined. For doing so,

we suppose the worst case scenario, when the disruption occurs while sending
the information packets and not when sending redundancy packets. In such a
case, to recover the information, the whole block composed of information and
redundancy packets (for instance block formed by packets D to 5), must be
received before being able to extract the information packets. This determines
the minimum buering time of 8 packets in Table 8.5.
Time 5 6 7 8 9 0 1 2 34 5 6 7 8 9 0 1 2 3 4 56 7 8 9 0 12 3
Informat. A B C D E F G H I J
frames
Tx frames AB C12 D E F G 3 4 5 H I J 67
Rx frames AB C12 F G 3 4 5 H I J 67
Buer at Rx A B C D E F
Table 8.5: FEC technique example. Shadowed cells correspond to disruption.
Note that these aspects can also apply if the FEC technique is employed
between the CN and the MN, being the ARs transparent to that. However,
there are many advantages of employing the FEC technique in the MN-AR
link and at link level. First, in our proposed FHO scheme, the AR already must
have special functionality like bicasting, thus adding this FEC functionality
will not complicate too much the AR. Second, redundancy is only present
in the last hop, besides in this last hop, FEC techniques may already be
employed because the link (e.g., air link) may be prone to errors, so, perhaps
our solution can be implemented just modifying the parameters of the existing
FEC. Finally, in a multicast scenario, the source will send the data and the
ARs would be the ones to add the appropriate FEC redundancy.
8.6 Switched Ethernet over LEO satellite: implicit

cross-layer design exploiting VLANs
The aim of this Section is to introduce the possibility of exploiting Switched

Ethernet facilities in a LEO network using Inter-Satellite Links (ISLs). We
refer here to Scenario 3 (see Chapter 1, sub-Section 1.4.5) for the study carried
out in this part of Chapter 8. The interest is to support QoS-aware resource
management procedures by tacking into consideration the mapping between
logical and physical network topologies (that are both well known), like in the
case of terrestrial switched-networks. In fact, the optimization of connection
paths can be performed, in a large network scenario, by means of routing
algorithms or switching mechanisms. The former solution is particularly suited
in case of networks whose topology is not completely known or predictable,
while the switching approach has been adopted in the scope of single domain
networks with known topology, or single-provider-managed area networks
(LAN, WAN, MAN). LEO satellite can be classied as an extension of
LAN/MAN, in which the network topology changes over time in a regular
and predictable way, thus the switching approach can be a natural candidate
for network management issues. Furthermore, an Ethernet-like switching
solution (i.e., the one proposed by IEEE 802.1) addresses the problem of
the interoperation between network layer and the satellite-specic MAC. The
mechanism is known as Logical Link Control (LLC), and provides a useful
framework to deploy an IP-MAC mapping with QoS control.
IEEE 802.1 standards oer a set of facilities meant to build, operate and
manage a network comprising one or more transmission technologies and one
or more communication methods (i.e., physical and MAC layers). Bridging of
dierent technologies is obtained by using special devices (i.e., bridges) that
are able to translate, when needed, frames from a given MAC format to a
dierent one. In particular, IEEE 802 standards propose a special LLC layer
[39], located just above the MAC layer [40], that is common to all network
segments and whose frames can be transported in any kind of MAC frames.
LLC is the glue that allows the system to interconnect dierent physical and
MAC solutions of the IEEE 802 family.
Switches can be used in order to reduce the number of competitors to the
same shared medium in a network, by segmenting the overall network and
bounding the frame transmission in a limited range. In a switched-network,
users can receive and transmit at the same time by means of two separate
channels, the so-called Full Duplex Switched Ethernet facilities that can be
suitably used in Gigabit LANs.
Now, it is worth noting that:
LLC functionalities are analogous to protocol adaptation functions that
are used for the transportation of IP trac over satellite devices;
LEO payload with bridging/switching modules is envisioned for future
satellite networking;
Full-duplex techniques are consistent with satellite communication sys-
tems, where dierent channels are commonly adopted for uplink and
downlink.
Thus, we rstly questioned how much existing Ethernet-like solutions
could be reused to obtain a protocol harmonization for satellite and terres-
trial devices; secondly, we investigated how existing mechanisms should be
enhanced to mach with satellite-specic issues, and in particular with LEO
mobility problems.
Our research turns in the exploitation of a cross-layer design of layers
2 and 3. In fact, layer 2 switching is performed on the basis of end-to-end
connections to be established (known from the IP demand, which does not
involve all the LEO network at once, but only a set of sub-networks, possibly
separated), and by taking into account the knowledge of logical path changes
due to the turnover of LEO satellite positions. In other words, we deploy

virtual sub-networks (Virtual Local Area Networks, VLANs) that span that
portion of the LEO network needed to accommodate the IP demand, while
continuously adjusting the logical VLAN topology on the basis of the predicted
satellites motion. As a result, it is possible to choose data-link connections in
order to balance the network trac, to optimize the usage of MAC resources
and to dierentiate users and services (i.e., groups to be handled in VLANs).
In turn, an enhanced degree of connectivity, robustness and QoS is provided
to the IP level.
8.6.1 Protocol harmonization and implicit cross-layer design via

IEEE VLAN
Protocol harmonization is meant for QoS-aware internetworking of commu-

nication segments, avoiding intricacies due to satellite gateways needed to
access a satellite sub-network, and also due to routing and path discovery to
be managed for time-varying topologies, typical of non-GEO satellite eets. As
a consequence, a LEO satellite constellation could be merged at layer 2 with
legacy terrestrial networks, and end-to-end communications could seamlessly
use satellite and terrestrial links.
Cross-layer design of layers 2 and 3 is meant to avoid the negative impact
of topological changes, deep signal shadowing, data loss, and reconguration
of layer 2 links needed to span the entire network and to avoid loops. However,
note that we deal here with implicit cross-layer design, since we aim at
optimizing the resource management in the MAC layer to support IP QoS
and robustness. The optimization of resource management is performed by
joining the eectiveness of spanning tree algorithms and the possibility to
congure remotely and proactively LEO switching devices on-the-y. To this
aim, a centralized path/VLAN manager is required that selects ISLs to be
activated at any time instant, provides path redundancy by means of multiple
disjointed VLANs, and avoids service discontinuity by switching the IP trac
from a VLAN to another.
The rationale of our proposal is based on the consideration that most of
topology changes in the network can be foreseen from the knowledge of the
satellite motion and from a statistical analysis of signal strength at receivers,
so that the switched-network can be proactively managed.
In order to manage eciently a switched-network, it is necessary to
maintain only a sub-set of inter-node links to form a connected loop-free graph
(i.e., a tree-like logical topology). This permits to conne broadcast trac,
eliminates looping frames, and, mostly important, makes easier the routing of
data frames by exploiting a tree that connects all nodes without redundancy.
However, the extraction of a tree from the original network graph has to be
performed in accordance with trac management criteria. Indeed, multiple
trees could be adopted to segment the network trac, to create virtual
sub-networks, to perform load balancing, or also to provide some redundancy
if a link abruptly fails. These advanced features are oered by IEEE standards
for VLANs [41] that include LLC for switching and bridging, Spanning Tree
Protocol (STP) and its variants Rapid STP (RSTP) [42] and Multiple STP
(MSTP) [43], and VLAN tagging methods. IEEE VLAN and MSTP can be
suitably adopted in order to simplify the management of a huge number of
connections, even though adopting satellite switching implies the constitution
of very large WANs or MANs where IP routing is unnecessary. Important
advantages can be obtained, such as the possibility to exploit a particularly
broad connectivity, or the possibility to eliminate IP route discovery latency,
frequent inconsistencies in IP routing tables due to LEO topology changes,
and path elaboration delays.
Although spanning tree protocols are able to rearrange their conguration
after a link or a node fails, the satellite VLAN approach only works if
spanning trees and VLANs are proactively adjusted when the LEO logical
topology changes. Note that legacy reconguration procedures could take
several seconds (due to the huge network diameter) during which the network
graph results unconnected. However, the adoption of proactive mechanisms is
reasonable since: (i ) the LEO satellite eet is known a priori ; (ii ) the LEO
eet can be designed so that at lest one satellite is always visible in the target
coverage area, and at least two satellites are visible during a handover event.
Thus, the proactive management simply consists of setting up a new VLAN
with a new spanning tree before the handover is performed, thus forcing a
VLAN handover before the physical handover. Note that, as for the VLAN
handover procedure, it simply requires to change the tagging of frames at the
edge of the satellite path from the old VLAN tag to the new one.
Considering the service oered to end-users, the adoption of VLANs with
proactively managed multiple spanning trees allows avoiding: (i ) IP data-ow
discontinuities due to physical topology changes, (ii ) waste of large time
intervals in spanning tree recongurations, triggered by the failure of a link
or a node, and (iii ) waste of bandwidth due to possible ooding eects after
recongurations.
8.6.2 Performance evaluation
Multiple VLANs allow the network provider to hide network topology changes.
In fact, during a topology transition, a new path will be available before the
old path goes down. We include these dierent paths within dierent VLANs
(i.e., addressed by dierent VLAN tags in the frame header) and switch to
the new path during the topology transition. The VLAN manager knows the
topology transition and enforces a VLAN tag change (i.e., a VLAN handover)
at the edge of the network, so that each frame will follow a path in the VLAN
identied by its new tag. In practice, we use multiple VLANs as redundant
sub-networks.
Table 8.6 [44] reports what happens to frames generated in a message
exchange between two terrestrial users at the edge of a Teledesic-like LEO
network, when UDP is used, comparing the case in which a VLAN handover is
adopted to a scenario without such a feature. We consider the case where only
two users are in the network and only one connection is active (2 ). Summariz-
ing, we can say that the absence of VLANs implies service discontinuities and
long ooding phases after recongurations (due to the unidirectional nature
of UDP exchange considered). Using VLANs, discontinuities of the service are
avoided: packets are not ltered and end-to-end connectivity is not broken.
The ooding after reconguration is bounded to the VLAN used after the
handover; whereas, an STP-based approach would ood the entire network
(STP uses a single tree for the entire network). Similar considerations can
be made when TCP is used, with the exception of the occurrence of short
ooding phases, due to frequent TCP ACKs.
UDP
Event Action (No VLAN) Action (two VLANs)
Service request Flooding Flooding in VLAN#1
from Client Switches learn Client VLAN#1 switches learn
address Client address
Service response Switching/no ooding Switching/no ooding
from Server Switches learn Server VLAN#1 switches learn
address Server address
Service data sent Switched/no ooding Switching/no ooding
Topology change Re-compute Spanning Tree Handover to VLAN#2
LAN temporarily Re-compute CIST and
disconnected VLAN#1 Spanning Tree
All frames discarded VLAN#1 temporarily
disconnected
VLAN#2 ooded, ltering
databases are empty
Downstream Network ooded by new VLAN#2 ooded by all
(Upstream) frames switches until an up- switches until an up- (down-)
after (down-) stream frame is stream frame is sent by Client
reconguration sent by Client (Server) (Server)
Table 8.6: How topology changes aect UDP connections. Note that each network
region that is managed by MSTP needs a Common Internal Spanning Tree (CIST)
to interconnect all nodes in that region. See reference [44]. Copyright 2005
c IEEE.
Table 8.7 [44] collects a set of actions performed by network entities at the
occurrence of specic TCP-related events. The advantage of using proactive
VLANs is clear from the comparison with the legacy behavior of switches,
2
This is the worst-case scenario, since switching devices need bidirectional ows
in order to learn the route towards a remote user, otherwise frames are ooded
in the VLAN.
as described in the central column of the table, and the behavior of the
VLAN with support for o-line reconguration, as described in the rightmost
column: VLANs reduce ooding eects and o-line reconguration eliminates
service discontinuities. However, a ooding phase is still needed in order to ll
the ltering database (a layer-2 routing table built by bridges by monitoring
incoming frames - source address and incoming port - to discover the outgoing
port to be used to forward frames without the need of ooding) of the new
VLAN.
TCP
Event Action (No VLAN) Action (two VLANs)
Request from Flooding Flooding in VLAN#1
Client (TCP SYN) Switches learn Client VLAN#1 switches learn
address Client address
Response from Switching/no ooding Switching/no ooding
Server (TCP ACK) Switches learn Server VLAN#1 switches learn
address Server address
TCP SYN-ACK Switched/no ooding Switched/no ooding
Service data ow Switched/no ooding Switched/no ooding
Clients ACK Switched/no ooding Switched/no ooding
Topology changes Re-compute Spanning Tree Switch to VLAN#2
LAN temporarily Re-compute CIST and
disconnected VLAN#1 Spanning Tree
All frames discarded VLAN#1 temporarily
disconnected
Downstream Flooded by new switches Flooded in VLAN#2
(Upstream) frame until an up- (down-) switches until an up- (down-)
after stream frame is sent by stream frame is sent by Client
reconguration Client (Server) (Server)
Table 8.7: How topology changes aect TCP connections. See reference [44].
Copyright 2005
c IEEE.
In what follows, we show some results obtained in a scenario similar to the

one depicted in Figure 8.17 [44], by means of the OPNET [45] simulator with
modied bridging/switching devices. Satellite orbits are designed following
the design principles of the Teledesic system, where LEO orbits have a 1375
km altitude and the satellite capacity is set to 32 Mbit/s for both uplink and
downlink with terrestrial users, even though channels can be allotted to users
as multiple of the 16 kbit/s basic channel. Two terrestrial bridges/switches are
considered (T1 and T2 in the gure). Users are located close to the terrestrial
bridges; the longest path between two users in the simulation has a delay
bounded to 200 ms. We tested both UDP and TCP-based applications with
and without VLAN supports.
Let us describe the generation of both UDP trac and the TCP one. In
Fig. 8.17: Possible network topology with two VLANs available. See reference [44].
Copyright 2005
c IEEE.
particular, UDP trac is obtained via simulation of unidirectional constant-

rate video connections; also packet distribution is constant. Three UDP groups
of users have been considered that dier in terms of both bit-rate (i.e., 256,
128 and 64 kbit/s, respectively for Class I, II and III users), and the average
request inter-arrival time (which is exponentially distributed, so that the
arrival process of connection requests is Poisson). Mean inter-arrival times
are, respectively: 45 s for Class I, 22.5 s for Class II, and 11.25 s for Class III.
Each connection has a duration exponentially distributed with mean value of
180 s. The number of users in a group is selected so that each UDP group
oers 1 Mbit/s trac in average, directed from nodes located in T1 to T2.
As for TCP-based trac, the following results have been obtained by
considering the separate contribution of three groups of FTP users. Every user,
located in T1, requests les of B bytes, where B is exponentially distributed
with a mean of 5,000,000 bytes, while the le request inter-arrival time is
exponentially distributed with a mean of 5 s. User groups are dierentiated
based on the available resources allotted in the access link: Class I (High
Rate) has an aggregate guaranteed rate of 512 kbit/s for the downstream
and 128 kbit/s for the upstream; Class II (Medium Rate) has an aggregate
guaranteed rate of 128 kbit/s for the downstream and 32 kbit/s for the
upstream; eventually, Class III (Low Rate) has an aggregate guaranteed rate
of 64 kbit/s for the downstream and 16 kbit/s for the upstream. Each group
saturates its link capacity due the high le request rate (5 les per second
are requested, i.e., about 25 MByte per second, which requires at least 120
Mbit/s plus the protocol overhead: the system is overloaded and the number
of FTP requests overwhelms the number of FTP sessions that reach the end of
transmission). As a matter of fact, simulations conrm the behavior described
in Table 8.6 for UDP and the considerations about TCP in Table 8.7. Details
are provided below.
Fig. 8.18: UDP throughput with STP, no VLANs.
Fig. 8.19: UDP throughput with RTSP, no VLANs.
Figures 8.18 to 8.20 show the throughput of a unidirectional UDP connec-

tion between two remote hosts. In the simulations, a physical topology change
occurred at t = 950 s, and one can notice that a traditional STP approach
requires up to 45 s to recover the path; using RSTP this time is shortened,
but several seconds, about 10 s, are still needed to recongure the large
switched-network. On the contrary, precongured VLANs allow a seamless
handover, without service discontinuities. In Figure 8.20, a VLAN handover
is enforced at t = 940 s, just a few seconds before the physical topology change.
Similar considerations could be made by considering bidirectional UDP ows,
where the trac is generated in each direction as in the unidirectional case.
Fig. 8.20: UDP throughput with VLAN handover.
Figures 8.21 and 8.22 depict the throughput of a TCP connection for
hosts requesting FTP les from a network server. In this case, trac ows
are bidirectional, due to the presence of ACK packets in the return channel,
even though the connection is strongly asymmetric. In these simulations,
a topology change occurred at t = 800 s. By using STP (Figure 8.21) or
RSTP, we can notice a service interruption with a duration similar to that
experienced in UDP simulations, but the eect is partially masked by the
build up of long queues at the last satellite-to-ground station link, especially
for the TCP Class III, which is allotted the minimum resources. It is worth
noting that after the network reconguration, each trac group aggregate
suers from high uctuation due to the synchronization of TCP ows after
the outage period. In particular, Class I experiences a very drastic uctuation,
while lower rate trac classes grow very slowly. Eventually, if we consider the
adoption of VLAN (Figure 8.22), with a handover operated at t = 790 s, no
signicant variation can be noted in the trac aggregate of each class. Again,
o-line congured VLANs allow ground stations to switch seamlessly between
VLANs, and avoid service discontinuities.
As for the ooding eects due to topology changes, rst we consider
unidirectional UDP ows in the network, from site T1 to site T2. Figures
Fig. 8.21: TCP throughput with STP, no VLANs.
Fig. 8.22: TCP throughput with VLAN handover.
8.23 and 8.24 represent data ooded by switches when no appropriate entries
are found in the ltering database. Each ooded data frame is accounted for
only once, no matter if multiple switches will ow again the same frame. In
practice, a ooding phase starts after an automatic route change, performed
by RSTP (or STP, not showed here). This is the reason why Figure 8.23
shows ooded packets for multiple sources after the rst disrupted path is
recovered, which is not mandatory for the data path we are interested to.
Fig. 8.23: Unidirectional UDP connections: normalized aggregated ooding

(RSTP).
Fig. 8.24: Unidirectional UDP connections: normalized aggregated ooding

(VLAN).
Thus, the ooding phase ends only after the network is fully recongured and
a new frame is sent in the reverse path for each user (i.e., after a new request
is sent per each UDP trac class, which is represented, in these simulations,
by a single user). Figure 8.24 shows that by adopting VLAN-based network
management, a simple VLAN handover is required a few seconds before the
original path goes down. However, VLAN handover requires a brief ooding
phase just after the handover, since the ltering database learning phase has
to be performed as well.
As for the ooding eects in case of bidirectional UDP connections, in the

same network and trac conditions as before, it can be found a very limited
ooding due to the fact that an intense trac is used in both directions, so
that database learning phases are very short.
Figure 8.25 shows the burden of ooding data for TCP connections when
RSTP is adopted. Data are related to frames carrying TCP segments (data
and ACKs) that experience at least one duplication in a generic network node.
Due to the asymmetric nature of TCP upstream and downstream (i.e., the
dierent size and bandwidth occupied by data and ACKs), it is appropriate
to distinguish between the amount of ooded bits and the number of ooded
frames: we represent the ooding in terms of ooded packets, which give
a normalized estimate of how much dangerous the ooding can be. Note
that ooding occurs while the network is reconguring itself. However, RSTP
operation allows data to be ooded immediately after the link failure. In fact,
when using spanning trees, each link is represented as an arch of an oriented
graph. The orientation of each arch is from the root to the leaves of the
tree. Thus, a link connects an up-node to a down-node. Using RSTP, the
down-node is in charge of sensing the link failure and starting the recovery
phase; the down-node is also allowed to use alternative links to reach the
root of the tree. The most important cause of ooding in RSTP is given by
frames that reach a down-node of a broken link. Eventually, ooding is almost
completely avoided by using VLANs, as stated by Figure 8.26.
Fig. 8.25: TCP connections: normalized aggregated ooding (RSTP).

Fig. 8.26: TCP connections: normalized aggregated ooding (VLAN).
8.7 Conclusions
Over the past decades, with the emergence of many multimedia Internet
applications and services, the research community has devoted a big eort
in an attempt to satisfy their stringent and varied QoS requirements. A clear
example of this eort is the initiative by IETF in proposing two IP QoS
frameworks. These frameworks are mainly designed with terrestrial networks
in mind. However, the problems of achieving QoS in networks with wireless
medium such as satellite networks are much more complicated since the link
is dependent on channel conditions. Hence, the resource management block is
vital in realizing the IP QoS frameworks.
Standard mechanisms which operate solely in the network layer most
often cannot guarantee the QoS when the end-to-end path involves satellite
or wireless networks as they disregard the variability of channel conditions.
This leads to the investigation of utilizing MAC layer resource management
schemes or protocols to improve this situation. More recently, the idea of using
cross-layer techniques further open up the potential of what can be achieved
in terms of QoS provision.
Being in adjacent layers in the protocol stack, resource management (layer
2) in satellite networks is always tightly coupled with the IP QoS frameworks
(layer 3). This Chapter has been dedicated to the cross-layer interactions and
issues between these two layers. A review of the current state of the IP QoS
frameworks in relation with the satellite network shows that DiServ is being
increasingly accepted and an example implementation of relative DiServ is
given as an illustration on how MAC layer scheduling can support the QoS
provisioning. The problem of mapping between the QoS mechanisms operating
at the two layers has been formulated and a measurement-based approach has
been presented. The problem is also discussed in two other scenarios; namely
the dual network access and Switched Ethernet over LEO satellites.
From the discussions and results presented in this Chapter, it is clear
that achieving IP QoS in a satellite environment can certainly benet from
cross-layer mechanisms from layer 2. Nevertheless, caution must be observed
when designing such cross-layer schemes. Uncontrolled implementation of
cross-layer mechanisms may cause other problems that may not be apparent
in a short period of time. Cross-layer design aimed at improving a specic
performance metric may not have the entire system performance considered
while cross-layer design involving multiple layers may lead to spaghetti design
with high number of convoluted interactions. All these aspects will increase
system complexity and hence will pose problems for future innovations. Worse,
system update may require complete redesign. Another example of a negative
impact of uncontrolled cross-layer design is on network security issues: the
increased interactions among layers may increase the channels for security
attacks. In conclusion, designers must have the long-term eects in mind.
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[45] The ocial OPNET Internet site with URL: http://www.opnet.com.
9
RESOURCE MANAGEMENT
AND TRANSPORT LAYER
Editors: Gorry Fairhurst1 , Michele Luglio2 , Cesare Roseti2
Contributors: Nedo Celandroni3 , Paolo Chini4 , Gorry Fairhurst1 , Giovanni

Giambene4 , Michele Luglio2 , Francesco Potort`3 , Cesare Roseti2
1
2
UToV - University of Rome Tor Vergata, Italy
3
4
9.1 Introduction
The main challenges to be faced by transport protocols using a satellite link
are the variability of the channel, due to weather conditions and the large
propagation delay. Adaptive network management and control algorithms are
therefore desirable to guarantee the Quality of Service (QoS) to data ows over
Additive White Gaussian Noise (AWGN) channels (a baseline assumption in
the following analysis).
Many popular Internet applications including email, le transfer, remote
access, and Web browsing require a reliable data delivery service. End-to-end
reliability for Internet trac is guaranteed by the Transmission Control
Protocol (TCP) at the transport layer. TCP specication covers a wide family
of implementations, some of them having traditionally very poor performance
over satellite links [1],[2]. Furthermore, TCP performance actually depends on
the adopted Radio Resource Management (RRM) techniques. The DVB-RCS
290 Gorry Fairhurst, Michele Luglio, Cesare Roseti
standard [3],[4] (see Scenario 2 for GEO-based communications, detailed in

Chapter 1, Section 1.4) denes a set of Demand Assignment Multiple Access
(DAMA) schemes, based on control loops with time constants similar to those
used by TCP. Specic design choices and circumstances can lead to unwanted
interactions between the link layer, implementing DAMA, and TCP, with a
signicant degradation of the overall performance. In particular, the so-called
access delay (DAMA loop) becomes an important factor that aects TCP
performance (see the following Section 9.4 for further details).
The variability of the satellite channel conditions also impacts Internet
ows that do not utilize TCP (e.g., streaming multimedia utilizing User
Datagram Protocol, UDP). This trac typically is less demanding in terms
of reliability, but more demanding in terms of jitter. Loss and/or corruption
of multimedia packet payloads can be eciently handled by the multimedia
codec at the receiver and does not imply the need for retransmissions (as in
TCP). The eect of propagation delay is also much less important, except
for interactive services requiring low round trip delay (e.g., Internet gaming,
which is not suited for satellite use).
Adaptive RRM procedures can be optimized for multimedia trac. In
particular, cross-layer optimization approaches are becoming widespread for
wireless networks in general, and their application to satellite links needs
to be studied both from a performance and an architectural viewpoint.
This Chapter addresses the design and the implementation of cross-layer
interactions between RRM (layer 2) and transport layer (layer 4) protocols
in satellite environments. In particular, Section 9.2 oers an overview of TCP
over GEO satellite links; Section 9.3 proposes an interaction between TCP and
lower layers aiming at maximizing the TCP connection throughput; Section
9.4 describes the design of a cross-layer interaction between TCP and Medium
Access Control (MAC) layer; Section 9.5 focuses on UDP and the performance
of multimedia applications over satellite links, and, nally, Section 9.6 provides
conclusions.
9.2 Overview of TCP over satellite
TCP is the primary transport protocol in the TCP/IP suite, designed to

provide good performance over congested packet-switched networks [5],[6].
Similarly to other window-based protocols, TCP aims at guaranteeing a
reliable and fair data exchange, despite congestion events (i.e., segment losses).
Satellite links present characteristics that signicantly dier from those of
wired links; moreover, TCP protocol mechanisms can be impacted by the
signicant delay and the presence of transmission errors [1]. In the GEO
scenario, satellites are located at an altitude of approximately 36,000 km
above the Earth at the equator latitude. In such a scenario, the Round Trip
propagation Delay (RTD) between two ground stations at the Equator is on
the order of 500 ms; this latency will be increased by other factors in the
Chapter 9: RESOURCE MANAGEMENT AND TRANSPORT LAYER 291
network (e.g., the transmission and propagation time of other links in the
terrestrial path, and queuing delays in gateways and routers). Hence, the
high RTD value entails a slow TCP congestion window (cwnd ) increase that
signicantly aects the end-to-end transfer rate [7].
Much research has been directed to improve the TCP mechanism eciency
over satellite links. This Section provides a complete survey of dierent
solutions proposed.
9.2.1 TCP standard mechanisms
TCP is implemented on top of the Internet Protocol (IP) and provides a

reliable, byte-streaming and bi-directional connection, allowing applications
to receive all segments in the correct order (a sequence number is associated
to each TCP segment). Two concepts can be identied as the basis of the
protocol: the acknowledgement (ACK) and the sliding window. ACKs are
short packets generated by the receiver, when a TCP segment is received. They
report to the sender the sequence number of the next segment the receiver
expects. The sliding window denes the amount of unacknowledged data that
can be sent at any given time [6].
TCP implements two basic mechanisms: ow control and congestion
control. The ow control scheme allows an adequate exchange of data between
two TCP nodes using a sliding window protocol, while the congestion control
scheme is based on two algorithms called Slow Start (SS) and Congestion
Avoidance (CA) [8]. These algorithms typically use two variables: cwnd and
slow start threshold (ssthresh). In particular, each sending end-system probes
the network congestion by gradually increasing the window of transmitted
segments (outstanding in the network) until the network becomes congested
and drops segments (or marks them as having experienced congestion).
Initially, the cwnd increase is exponential during the SS phase. The SS
algorithm is quite simple and based on the amount of data sent in each
Round Trip Time (RTT). Note that RTT is meant as the delay perceived
at the transport layer, that is the RTD plus extra delays due to queuing,
transmission time, processing, etc. At the beginning, the source sends one (or
a small number, e.g., 2) TCP segment and waits for the relative ACK. Then,
for each received ACK, it sends two segments. Therefore, each time the sender
receives the ACKs for the window of data it has sent (cwnd ), it doubles the
amount of packets sent in the next RTT. When the cwnd size reaches the
sstresh value, the increase becomes linear (i.e., one extra segment for each
RTT period), thus allowing for a gentler probing of the available capacity;
this is the CA phase.
TCP loss recovery
The default loss recovery mechanism of TCP is the Retransmission TimeOut

(RTO) [6]. TCP performs a retransmission if RTO expires before the related
segment is acknowledged. When RTO is over, TCP re-enters the SS phase

by resetting cwnd and by re-transmitting the rst unacknowledged segment.
Basically, TCP uses the ACKs reception time to estimate the RTT and a
smoothed average of RTT is used to set the RTO timer.
In modern implementations, TCP RTO is the nal fall-back method and
the timer rarely expires, hence, other methods are used for loss recovery. In
particular the Fast Retransmit algorithm [8],[9] allows a sender to re-transmit
a lost segment before RTO expires by exploiting the reception of duplicate
ACKs (dupACKs) generated when segments are received out of order. In
fact, TCP interprets the reception of a small number of dupACKs (usually
3) as an indication that a segment has been lost and retransmits it without
waiting for the RTO expiration. Furthermore, TCP considers the loss as a
congestion signal and reduces its transmission rate. To this purpose, the Fast
Recovery algorithm halves cwnd as described in [9] and TCP enters the CA
phase. Most TCP implementations, used within the Internet, add a method
called Selective ACK (SACK) [10], to further optimize loss recovery when
multiple losses occur in the same window of data. This option allows a sender
to recover quickly from multiple lost segments. SACK also permits to achieve
better performance with respect to multiple fast retransmissions. When SACK
is used, a sender is generally able to determine which segments need to be
retransmitted in the rst RTT following loss detection. This avoids a slow start
period (especially costly for high-delay links) following multiple segment losses
and permits the sender to continue to transmit segments (retransmissions and
new segments, if appropriate) at a suitable rate.
9.2.2 Criticalities of TCP on satellite links
TCP standard mechanisms, optimized to work correctly in wired (congested)

networks, suer from a certain number of factors when used over satellite links
[1]. In particular, the high latency, the large Bandwidth-Delay Product (BDP),
link asymmetry, and channel errors can negatively impact TCP performance.
TCP problems are experienced both while increasing the sliding window
and in the steady-state. The former is due to the dependence of the TCP
congestion control algorithms on the experienced RTT that in a satellite link
is one or two order of magnitude greater than that in terrestrial networks.
In fact, both SS and CA algorithms increase cwnd by using RTT as time
parameter. The formulas below describe the time spent in SS and CA phases
at the beginning of a TCP connection:

slow start time = RT T log2 (ssthresh)
(9.1)
congestion avoidance time = RT T (W ssthresh)
where W denotes a suitable cwnd value reached in the CA phase (the ideal
one corresponding to BDP).
Hence, the time needed to reach a given W is proportional to RTT.

Furthermore, packet losses are interpreted by TCP as possible indication
of congestion, leading to a cwnd reduction, thus slowing down the growth
process.
In the latter type of problems, the maximum achievable throughput and
then TCP steady-state performance are limited by the following formula:
T CP receiver window
M ax throughput = (9.2)
RT T
where TCP receiver window indicates the maximum amount of data the re-
ceiver can store in its buer at every time. In many systems, this value is adver-
tised using a 16-bit eld in the TCP header (maximum TCP receiver window
= 65535 bytes).
Despite widespread support for much larger TCP receiver windows within
the current deployed protocol stacks, these are rarely enabled by default;
therefore, in GEO satellite links (with RTT about equal to 540 ms), the upper
bound for the throughput is around 1 Mbit/s.
9.2.3 Survey of proposed solutions
Many solutions can be adopted to improve the TCP eciency over satellite
links; some of them are specically proposed for the satellite environment
while others for more general cases. Such solutions can be classied as follows:
Enhancements of the standard
Loss recovery enhancements (i.e., SACK option [10]);

Large initial window;
Delayed acknowledgements to reduce the ACK ow;
Byte counting.
Modied algorithms
Experimental implementations of modied ow control mechanisms and

options (i.e., TCP Vegas [11], TCP Peach [12], TCP Hybla [13], TCP
Westwood [14]).
Modied architecture
Performance Enhancing Proxies (PEPs) [15] are often employed to im-

prove the TCP performance. Many deployed satellite systems use PEPs
to improve the performance of the TCP protocol to compensate for eects
such as: delay, appreciable packet loss, variable bandwidth, asymmetry,
mobility or other eects. A range of PEP techniques can be and are
used; there is no standard PEP that satises all needs, and the most
appropriate method will depend upon the service requirements (whether
IPv6, mobility, IPsec, etc. are used), the link characteristics and the degree
of complexity that users can tolerate in a middlebox. One common PEP
method is to modify the end-to-end architecture at the transport protocol
level (i.e., splitting the path, terminating connections, acknowledging
packet receptions) by using one of the following approaches:
Enhanced TCP version;

Optimized protocols (i.e., XTP [16], SCPS-TP [17], etc.)
9.3 Cross-layer interaction between TCP and physical

layer
It is possible to optimize the TCP performance over satellite links without
changing the TCP behavior by operating on transmission parameters. Let us
focus here on how TCP performance can be improved by trading packet loss
rate for bottleneck link bandwidth. This procedure does not interfere in any
way with the normal behavior of the TCP stack, but requires the capability
of tuning link parameters at the physical level.
For a given available radio spectrum, antenna size and maximum trans-
mission power, there are many choices for the selection of modulation scheme,
symbol rate and Forward Error Correction (FEC) type. Commonly used
wireless systems make such design choices in a static mode, permitting the
user to change manually some of them, or in some cases to switch dynamically
among a limited number of preset parameters. For each possible set of
parameters, we dene Information Bit Rate (IBR) as the link speed seen
by the network layer, that is, the product of the symbol rate, the number
of bits per symbol and the FEC rate, as detailed in what follows. Even in
those cases where a wide range of IBR values is available, the criterion to
switch among them is only dependent on the perceived channel quality, i.e.,
on the performance measured at the physical layer [18]. For terrestrial wireless
environments, an example of physical layer with multiple choices is provided
by the IEEE 802.11 standard, where it is possible to change dynamically
modulation and coding schemes [19].
The hopping among dierent sets of physical layer parameters is due to the
highly variable physical characteristics of most wireless links; in particular,
all satellite links are subject to variable atmospheric attenuation of the
signal; Low and Medium Earth Orbit (LEO and MEO) satellite constellations
additionally suer from variable signal attenuation (due to changing slant
path), blocking (due to obstacles in the Line of Sight, LoS), multipath fading,
and changing satellite distance.
Internet has been conceived assuming that Packet Error Rate (PER)
should be as low as possible for a good TCP performance [2]: a common rule
of thumb is to engineer the link so that PER due to link errors is negligible
with respect to the loss rate caused by congestion. However, this could not
be the appropriate choice; in [20] in fact, it is shown that a good rule of
thumb is to set the ratio between PER and congestion loss to a value equal
to the number of TCP connections. This nding opens the possibility for
dening adaptive algorithms that dynamically choose the optimal channel
parameters (e.g., modulation and coding) with the aim of maximizing the TCP
performance. Implementing such algorithms requires a cross-layer approach,
because physical layer parameters such as modulation and coding need to be
tuned depending on information available at the transport layer.
The following analysis of the TCP performance highlights the cross-layer
issues and interactions with the physical layer by relating PER with the TCP
throughput. In 1997, a simple and elegant formula, relating the steady-state
performance of TCP to its segment loss rate, was discovered [21]. This formula
connects the maximum throughput on an unlimited bandwidth channel with
the Maximum Segment Size (MSS), RTT, and the packet (segment) loss rate,
PER:
M SS
throughput = K f or P ER < 1% (9.3)
RT T P ER
where K is a constant equal to 1.31 in the case of random segment losses
without delayed ACKs.
Subsequently, (9.3) was modied to take into account the TCP behavior
in the presence of timeouts [22], thus allowing for a greater accuracy at higher
PER, resulting in:
M SS
throughput = ( ( (9.4)
2bq
RT T 3 + RT O min 1, 3 3bq
8 q (1 + 32q 2 )
where b is the number of segments acknowledged by each ACK and q is

the PER.
Finally, in [23], a method for computing the TCP throughput in band-
limited channels was proposed.
The above described relationship between PER and TCP throughput was
used in [24] to optimize single-connection TCP performance, and in [20] for the
case of multiple connections. In fact, for a given wireless transmission system,
it is realistic to assume that some of the parameters are dynamically tunable,
so that the channel PER can be traded for IBR in order to nd the optimal
conguration that maximizes the TCP throughput computed at the end-user
(i.e., goodput). For example, it is possible to change the modulation scheme,
thus reducing the channel bit-rate, in order to obtain higher bit energy to
one-sided noise spectral density ratio, Eb /N0 . The modulation scheme can be
changed together with the FEC characteristics to have a wider range of choices
and to exploit as much as possible the available radio spectrum. Most modern
transmission systems provide for variable bit-rates by changing the used FEC
redundancy, some of them for each individual packet. It is even possible to
seamlessly change the FEC, while maintaining bit time synchronization of the
data stream, by using rate-compatible punctured convolutional codes [25].
Concerning the modulation scheme, let us consider a satellite carrier modu-
lated at a rate of S symbols/s. We envisage widely diused M -ary modulations
such as Amplitude and Phase Shift Keying (APSK) or Quadrature Amplitude
Modulation (QAM) types. In these schemes, M is the number of points, in the
phase-amplitude space, relative to the constellation of the modulated symbols.
Typical values of M are 2, 4, 8, 16, 32, and 64; Binary Phase Shift Keying
(BPSK) and Quadrature Phase Shift Keying (QPSK) schemes correspond to
M values of 2 and 4, respectively.
FEC types come in a multitude of modes, often concatenated between
them. We dene the coding rate r as the inverse of the coding redundancy.
For example, in the case of a (255, 223) Reed-Solomon code concatenated
with a 1/2 convolutional code, the resulting r is 223/(255 2) = 0.437.
The IBR (i.e., the TCP bottleneck rate) is given by
IBR = Sr log2 M. (9.5)

For a given channel condition C /N0 , a modulation scheme with M points
and a coding rate r, the operating Eb /N0 expressed in dB is given by
Eb /N0 = C/N0 10 log10 IBR [dB]. (9.6)

Diagrams similar to the one shown in Figure 9.1 [20] are used to estimate
PER, given the channel parameters.
For each C /N0 , PER is a function of M and r. Since the TCP goodput
depends on IBR and PER, it is possible to nd M and r that maximize
the goodput for each given C /N0 . The way the goodput depends on IBR and
PER can be evaluated using either experimental measurements, or simulation,
or an analytical expression, such as (9.3) or (9.4). As an example, Figure 9.2
shows the goodput achievable by considering two modulation schemes, namely
BPSK (M = 2) and QPSK (M = 4) and four coding schemes, all based on the
standard NASA convolutional coding with constraint length 7, with code rates
r = 1/2 (base code), r = 3/4 (punctured base code), r = 7/8 (punctured base
code), r = 1 (no coding) [26]. Four curves are depicted, each for a dierent
C /N0 . Each curve represents how the TCP goodput changes as a function of
IBR (i.e., the bottleneck rate). For each channel condition (i.e., C /N0 value),
an optimum combination of parameters exists which gives the maximum TCP
goodput. Notice that the curves may exhibit notches due to the discontinuous
parameter space. In Figure 9.2, for example, this happens for the case of C /N0
= 67 dB, where BPSK at 7/8 coding rate has both a smaller IBR and a worse
PER performance than QPSK at 1/2 coding rate.
Fig. 9.1: PER as a function of Eb /N0 for dierent combinations of concatenated

codes. The int in the legend means ideal interleaving. See reference [20]. Copyright
2006
c IEEE.
Fig. 9.2: Goodput of a single TCP NewReno connection versus the available IBR
for dierent C/N0 values. Packet size = 1 kB, S = 1.024 Msymbol/s, bottleneck
buer size = BDP of a GEO link. Labels indicate modulation schemes and FEC
rates.
This gure is reproduced from Transport Layer Protocols and Architectures for Satellite
Networks, C. Caini, R. Firrincieli, M. Marchese, T. de Cola, M. Luglio, C. Roseti, N.
Celandroni, F. Potort`, International Journal of Satellite Communications and Networking,
Vol. 25, No. 1, pp. 126, January/February 2007. Published Online October 10, 2006
http://www3.interscience.wiley.com/cgi-bin/jissue/104548349. 2006.
c Copyright John Wiley &
Sons Limited. Reproduced with permission.
The above discussion leads to the conclusion that satellite systems could
benet from adaptive algorithms for choosing the transmission parameters by
means of cross-layer interactions between transport and physical layers. An
additional possibility is that MAC and physical layers interact by inserting
a link-layer erasure code [20],[27] just above MAC layer, which could be an
all-software solution, independent of the underlying hardware characteristics.
The recent DVB-S2 standard [28] considers very powerful error-correcting
codes. For ideal AWGN channel conditions, an optimization based on channel
coding would be useless because the curves that give PER versus Eb /N0 are
very steep [29], causing a sort of on-o behavior of the physical channel: either
PER is negligible, or it is so high that it collapses TCP performance. However,
optimizing channel parameters makes sense in non-ideal channel conditions,
and, in general, on the satellite return channel [3].
9.4 Cross-layer interaction between TCP and MAC

The interaction between TCP and MAC protocols in a shared network can
greatly improve the eciency of satellite systems. MAC protocols play a
fundamental role to guarantee good performance to higher-level protocols by
managing the arbitration of uplink access. Two cases must be distinguished:
(i ) when TCP operates end-to-end (as the general Internet standard, or when
an end-to-end IPsec protection scheme is used); (ii ) a PEP scheme violates the
end-to-end semantics. Without loss of generality, hereafter we will consider the
latter case, where, referring to a DVB-RCS network (see Chapter 1, Section
1.4), the Gateway acts as a PEP (i.e., it is a local TCP receiver -from remote
RCSTs-, located in the Earth station).
Satellite networks employing a DAMA scheme introduce an additional con-
tribution to the end-to-end delay, called the access delay that can signicantly
impact the end-to-end performance of TCP ows. In a DVB-RCS-like network,
the Network Control Center (NCC) assigns return link capacity in response to
explicit requests received from RCSTs [3]. This capacity negotiation requires
a signaling exchange that regulates the data ow. Therefore, when TCP is
used as transport protocol, two nested control loops exist with the same time
constants (i.e., RTT):
At MAC layer: resource request - resource assignment loop;
At TCP layer: TCP segment - acknowledgement loop.
The consequence of this interaction is an increase in the latency perceived
by the end-systems. To mitigate this eect it is possible to reduce the access
delay with a preventive allocation scheme driven by a cross-layer interaction
between MAC and TCP layers. The idea is to use the TCP parameters, such
as cwnd and ssthresh, to estimate in advance the resources needed by a given
TCP ow [30],[31]. In fact, from the comparison of these two quantities, it
is possible to determine the TCP congestion control status (i.e., SS or CA).
Consequently, the MAC layer can know the law according to which cwnd is
enlarged on an RTT basis and can predict with a very good accuracy the
necessary resource allocation needed by each TCP ow. In this way, it is
expected to reduce signicantly queuing delay, while also achieving an ecient
utilization of the satellite shared capacity. More details on this approach are
provided in the following sub-Sections.
9.4.1 A novel TCP-driven dynamic resource allocation scheme
The implementation of a dynamic access scheme allows optimizing the re-

source sharing. The DVB-RCS standard denes the following set of capacity
request methods (see for more details Chapter 1, sub-Section 1.4.3): CRA,
RBDC, VBDC, AVBDC, and FCA.
In particular, VBDC performs a capacity request, as long as new data
arrive in the RCST queue. The amount of capacity per frame, a generic
RCST requests at the k -th super-frame, can be expressed by using the formula
dened in [32]:
) L1 *
q (k) ns a (k) ns j=1 r (k L + j) ns w (k)
r (k) = (9.7)
ns
where:
. denotes rounding to the upper positive integer;
q(k ) = amount of queued data;
ns = number of frames per super-frame;
ns a(k ) = capacity assigned in the k -th super-frame;
L = system response time expressed in super-frames (also indicated as
allocation period ); it represents the time elapsed from a capacity request
transmission
L1 to the actual assignment of the requested capacity;
ns j=1 r (k L + j) = resources requested in the previous super-frames,
but not yet assigned;
ns w (k ) = resources requested in the previous allocation periods and not
yet assigned.
Unfortunately, the VBDC allocation method leads to a huge increase in
the end-to-end delay perceived by the systems where TCP applications are
running. In fact, the above mentioned access delay involves in this case the
following contributions:
Reservation delay: since requests are sent at a xed rate in dedicated slots,
a time interval occurs between the arrival of data in the MAC buer and
the transmission of the corresponding capacity request;
RTD contribution: sum of the time to propagate the capacity request from
the RCST to the NCC and the time to deliver the Terminal Burst Time
Plan (TBTP) in the opposite direction;
Processing (and synchronization) delay: time spent by the DAMA con-

troller (in the NCC) to transmit the TBTP message with the capacity
assignment;
Forwarding delay: time between the reception of the TBTP by the RCST
and the actual transmission of data.
On the basis of the above delay contributions, the RTT values correspond-
ing to the VBDC case can be of the order of 1.6 s (1 ) in a standard GEO
bent-pipe system [33].
The DVB-RCS standard also supports an RBDC capacity request method.
In this case, resources are allocated on the basis of the rate at which an RCST
wishes to transmit (usually based on monitoring the arrival rate at its layer
2 queue). This method reduces the access delay.
Most RCS systems provide a wide range of Bandwidth on Demand (BoD)
schemes based on a combination of both methods (VBDC and RBDC). As
already anticipated in Section 9.4, our interest here is in reducing the access
delay, keeping optimal network eciency, by using TCP status information to
predict the amount of data that will feed the RCST queue in the future. In
order to exchange cross-layer signaling between layer 2 and layer 4, dedicated
local messages [31] are generated each time that TCP parameters (e.g., cwnd )
go beyond a certain threshold; this is according to an explicit cross-layer
method.
Let us assume a system response time greater than the physical RTD
(2 ), in computing the r (k ) request. Such assumption allows to the proposed
algorithm predicting the further data that will be present in the RCST queue
when the resources will be allocated, according to both the amount of data
transmitted in the k -th super-frame and the TCP phase (SS or CA):

2 ns " a (k) # Slow Start
Q (k) = . (9.8)
ns a (k) 1 + cwnd 1
Congestion Avoidance
Therefore, in our TCP-driven RRM a new term is added to (9.7) and,

therefore, the amount of resources per frame requested at the k -th super-
frame, r (k ) is:
r (k) = (9.9)
) L1 *
q (k) ns a (k) ns j=1 r (k L + j) ns w (k) Q (k)
= + .
ns ns
1
The value of RTT 3 RTD is due to the use, for the simulations, of an architecture
where NCC is separated from the Gateway.
2
This assumption is appropriate to current DVB-RCS systems when the TCP
ow is not encrypted, especially when PEP mechanisms are used at the satellite
Gateway to end TCP connections within the satellite segment.
Finally, in addition to r (k ), also the TCP phases will be communicated by

the RCST to the NCC in the capacity request message by setting the following
ag (TCP phase ag):
1 Slow Start;
0 Congestion Avoidance.
On the other side, the NCC serves all incoming requests by considering
two priority levels: a High priority level associated to requests with the TCP
phase ag set to 1, and a Low priority level associated to requests with the
TCP phase ag set to 0. Our aim is to prioritize connections in the SS phase
with respect to those operating in the CA one to favor both short transfers
and just started connections. In each queue (i.e., the queue for requests in the
SS phase and the queue for requests in the CA phase), requests are satised
according to Maximum Legal Increment (MLI) algorithm [34] to guarantee a
fair allocation among the dierent competing ows.
If the amount of needed resources exceeds those available in a super-frame,
the NCC creates a waiting list to assign the resources in the next super-
frames and stops the cwnd growth of all the connections coming from the
RCSTs that have not obtained the requested resources. In particular, the
proposed allocation scheme at the NCC performs the following two tasks:
Assure that resources are fairly shared among all the active TCP connec-
tions;
Provide a further cross-layer action that sets a new variable, named cwnd*,
in order to modify the current cwnd value used by the TCP source in the
RCST as follows: cwnd cwnd*. Note that the NCC (acting like a
PEP) sends back the cwnd* value by using a eld for TCP options (layer
4 ACKs) in the headers. The rationale of this modication on the TCP
protocols is to avoid internal congestion on the RCST side and, then, the
possibility of layer 2 buer overows.
The main expected eects of the proposed cross-layer-based access scheme

are:
Reduction of the access delay: since the request algorithm predicts also the
amount of data that will feed the RCST queue due to the TCP congestion
control mechanism, the access delay will be reduced of an RTD;
Avoidance of internal congestions at the RCSTs: the cross-layer interaction
between RRM and TCP layers permits to prevent layer 2 buer overows
due to satellite network congestion;
Ecient and dynamic resource allocation: resources are dynamically as-
signed on a super-frame basis according to explicit requests, thus allowing
a better utilization of the available capacity.
Analysis of the allocation process
A simulator has been implemented using ns-2 (release 2.27) [35], in order to
evaluate the performance of the cross-layer allocation process and the resulting
performance. In particular, the ns-2 extensions that reproduce a traditional
GEO satellite network have been modied to simulate a centralized Multi
Frequency - Time Division Multiple Access (MF-TDMA) scheme and the NCC
functionalities.
The interaction between the TCP cwnd trend and the corresponding
allocation process has been analyzed by means of the average resources
assigned (in slots) as a function of time; such parameter has been monitored
for one or more TCP connections sharing the return link of a communication
network compliant to Scenario 2 described in Chapter 1, sub-Section 1.4.5.
The main simulation parameters are detailed in Table 9.1.
Physical parameters
Physical RTT (RTD) 515 ms
Return link bandwidth 2048 kbit/s
Maximum number of RCSTs 32
Frame parameters
Super-frame duration 96 ms
Number of slots per frame 32
Protocols
Transport Protocol TCP NewReno
Application Protocol FTP
TCP parameters
TCP packet size 1500 bytes
PER Variable, from 0 to 0.0001
Table 9.1: Main simulation parameter values.
In particular, by considering a le transfer (where the application layer

is achieved by means of the File Transfer Protocol, FTP) from an RCST to
the NCC, Figure 9.3 highlights how the allocated resources (continuous line)
are strictly correlated to the cwnd trend (dotted line) with our scheme. In
particular, three dierent phases can be recognized in the allocation process
according to the following sequence:
1. An initial exponential growth corresponding to the TCP SS phase;

2. A clear reduction of the allocated resources (approximately one half) when
the Fast Recovery mechanism is invoked as reaction to the detection of a
loss;
3. A linear growth corresponding to the TCP CA phase.
Fig. 9.3: Comparison between allocated resources and cwnd trend versus time (1
TCP connection, PER = 104 ).
Referring to our TCP-driven RRM scheme, Figure 9.4 focuses on the fair
resource sharing between two TCP connections, when losses occur. At the
beginning, the capacity is saturated (i.e., the NCC stops the cwnd growth of
both the connections in order to prevent congestion and losses): the overall
capacity is perfectly divided between the two connections. When a connection
is aected by a transmission error (loss), with consequent cwnd reduction, the
NCC re-assigns temporarily the unused capacity to the other connection in
order to optimize the utilization of resources.
Performance evaluation
The TCP performance strictly depends on the perceived latency at the

end-systems, as shown by (9.1) and (9.2). Therefore, RTT can represent a
valid parameter to evaluate the TCP performance. Hence, we have compared
our TCP-driven RRM scheme with the classical CRA and VBDC capacity
allocation techniques [3]. The main simulation parameters, compliant to
Scenario 2, are those provided in the previous Table 9.1.
Figure 9.5 shows the average perceived RTT for the three considered access
schemes. In particular, the obtained results allow the following considerations:
VBDC presents the higher delay equivalent to about three times the
physical RTD (see Chapter 1 for RTD characteristics) [33]: 1 RTD for
the capacity request (on the basis of new data in the layer 2 queue, RCST
side) and notication exchange; 1 RTD for the TCP segment and ACK
Fig. 9.4: Comparison among allocated resources in the RTT versus time (2 TCP
connections, PER = 104 ).
Fig. 9.5: Comparison among average RTT values obtained with the following
techniques: VBDC, CRA and cross-layer scheme.
exchange; 1 RTD for the capacity allocation for the availability of the
channel for ACK transmissions (Gateway side).
In the CRA case, RTT is only aected by the physical delay RTD, since
the capacity is not negotiated, but permanently assigned in the set-up
phase of a connection;
The proposed TCP-driven RRM scheme (also simply called cross-layer
scheme in what follows) reduces the overall VBDC delay by almost 1
RTD, trying to predict the amount of data that will feed the RCST queue.
Then, by evaluating only the end-to-end performance in terms of RTT for

a single TCP connection, the proposed cross-layer technique represents a good
trade-o solution between VBDC and CRA.
The principle of assigning capacity on the basis of the real needs of
data sources leads to signicant improvements in terms of both end-to-end
performance and network utilization when multiple TCP connections compete
for the overall capacity.
The following simulations have been performed considering: 20 FTP
transfers coming from dierent RCSTs and with start time instants spaced
of 5 s; 10 Mbytes les have to be uploaded to a remote system through the
satellite Gateway. As a reference, a xed allocation scheme (i.e., CRA) is
considered where the capacity is equally divided at the beginning among the
RCSTs in a static manner. The average le transfer time has been measured
for dierent PER values and then compared with the mean transfer time of the
proposed cross-layer scheme. The results, shown in Figure 9.6, highlight that
the TCP-driven RRM scheme with cross-layer information allows a reduction
of the mean transfer time ranging from 12.3% (PER = 0) to 26.5% (PER =
0.01).
Finally, Figure 9.7 highlights the benets derived from the use of the
proposed cross-layer scheme with respect to CRA in terms of channel uti-
lization. In fact, the continuous line indicates the percentage of the average
utilization increase, for the cross-layer scheme with respect to CRA, when 5
FTP transfers (10 Mbytes) are running at instants spaced of 5 seconds with
PER = 103 . This gure also shows the curve representing the instantaneous
channel utilization when the cross-layer scheme is used (dashed line), in order
to show the optimal values constantly achieved.
9.5 Overview of UDP-based multimedia over satellite
This Section focuses on multimedia transport in satellite networks, with a

specic reference to Scenario 2 described in Chapter 1, sub-Section 1.4.5.
Cross-layer methods oer new opportunities for satellite systems to adapt
RRM to the needs of multimedia trac. The challenge is the design of
cross-layer mechanisms that can optimize the overall end-to-end multimedia
application performance over satellite links, while minimizing the utilized
Fig. 9.6: Average le transfer time versus PER (20 FTP transfers starting at
instants spaced of 5 s).
Fig. 9.7: Cross-layer access scheme: utilization and percentage of the average
utilization increase with respect to the CRA scheme (5 FTP transfers starting at
instants spaced of 5 s, PER = 103 ).
radio resource. This topic requires a combination of expertise in propagation

analysis, channel modeling, coding and modulation, jointly with consideration
of link framing design and transport protocol design/evaluation. Analysis can
be performed by combining physical simulation (based on propagation models)
with packet-level protocol simulation (including application modeling).
9.5.1 Cross-layer methods for UDP
Examples of multimedia cross-layer methods include adapting transport pro-

tocols and application mechanisms to make them more robust to changes in
the link quality conditions [36].
A rst type of cross-layer method uses RRM and QoS techniques to tailor
lower layer parameters to the characteristics of particular multimedia ows
(as proposed for TCP in earlier Section 9.4). The requirements for multimedia
trac can dier from application to application. This kind of cross-layer
communication also implies some form of signaling exchange between dierent
protocol layers.
Recognizing the emergence of error-tolerant codecs, IETF has recently
standardized a new multimedia transport protocol, named UDP-Lite [37],
allowing an application to specify the required level of payload protection,
while maintaining end-to-end delivery checks. In order to benet from using
UDP-Lite, the changes at the transport layer must be reected in the design
of satellite link and physical layers. Hence, it is important to tune the
characteristics of lower layers in terms of modulation and coding (trading
BER for IBR).
Cross-layer signaling may also be valuable to indicate the prevailing system
performance to transport entities (in PEPs or end-hosts); this could also
permit multimedia applications to adjust their choice of media codec in
response to increased delay or reduced capacity. Hence, the use of cross-layer
methods can provide increased information to the transport layer and ap-
plications concerning the quality and characteristic of the channel they are
using. This new exibility gives opportunity to higher-layer protocols to react
in appropriate ways.
The success of multimedia cross-layer approaches relies not only on the
development of suitable techniques, but also on the selection of appropriate
signaling methods, and on the adoption of design methodologies that will
permit cross-layer systems to inter-work and to evolve.
9.6 Conclusions
This Chapter provides an overview of the key issues that concern transport
protocol performance over paths that include a GEO satellite segment. In
particular, it gives a detailed survey of several approaches that permit a better
interaction of transport layer protocols with RRM and physical layers.
Adaptive resource management can both guarantee ecient network uti-

lization and satisfy QoS requirements, using satellite links that are aected
by variable weather conditions and large propagation delays. An approach
tuning the satellite link parameters at the physical or link layer and trading
bottleneck bandwidth with segment error rate can permit to improve TCP
performance. Moreover, where it is possible to evaluate the channel conditions
in real-time, further sophisticated cross-layer interactions could be exploited
for an adaptive selection of the physical layer parameters.
DAMA schemes may be used to achieve an ecient resource allocation,
but they degrade the TCP performance by adding an access delay that
increases the whole end-to-end delay. Then, explicit cross-layer approaches
can also mitigate the interactions between TCP and DAMA (MAC layer).
The rationale is to use TCP information to estimate in advance the amount
of resources needed for a given TCP source. This should permit MAC layer
to perform capacity requests based on both the volume of queued data and
the predicted TCP trac behavior. Simulations show that this TCP-driven
RRM scheme represents a good trade-o solution with respect to both xed
access schemes, optimizing TCP performance, and classical dynamic access
schemes, optimizing network eciency.
Transport protocols not based on TCP can also benet from cross-layer
methods. Multimedia trac ows using UDP and UDP-Lite are also expected
to benet from improved communication between protocol layers; in fact,
application performance can be tuned to link and physical layer conditions,
in order to achieve a system optimization.
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10
CROSS-LAYER METHODS
AND STANDARDIZATION ISSUES

Editors: Gorry Fairhurst1 , Mara Angeles Vazquez Castro2 ,
3
Giovanni Giambene
Contributors: Gorry Fairhurst1 , Giovanni Giambene3 ,

Gonzalo Seco Granados2 , Alessandro Vanelli-Coralli4 ,

Mara Angeles Vazquez Castro2 , Fausto Vieira2
1
2
3
4
UoB - University of Bologna, Italy
10.1 Introduction
This Chapter describes a number of dierent techniques, approaches and
architectures for cross-layer design. It also seeks to position the work presented
throughout this book with respect to current and anticipated standards,
indicating opportunities for future standardization.
The challenge to be faced is the design of cross-layer mechanisms that
can optimize the overall end-to-end application performance over satellite
links, while minimizing the utilized radio resources. This optimization can also
require additional signaling between the protocol layers. This new area of work
is consistent with the end-to-end argument [1], provided that system-level
implications are understood [2]. Suitable methods are expected to improve
314 G. Fairhurst, M. A. V
azquez Castro, G. Giambene
signicantly the performance of applications in the next generation of satellite

systems, but will require changes to the design of protocols and systems, with
implications on the related standards.
The discussion in this Chapter utilizes some basic ideas introduced in the
previous Chapters 1 and 4.
10.2 Cross-layer design and Internet protocol stack

The current Internet protocol stack in Chapter 4 is used as the reference
architecture for discussion of cross-layer design throughout this Chapter. De-
sign principles categorize and dene the placement and operation of functions
within a given system. These design principles impose a structure on the
design area, rather than solving a particular design diculty. This structure
provides a basis for discussion and analysis of trade-os, and suggests a strong
rationale to justify design choices.
The various standardization bodies dene protocols that may be used by a
system to exchange information typically specifying a protocol at a single layer
of the system architecture. A cross-layer design goes beyond this structure
in two ways: by increasing the awareness between layers or by implicitly
conveying information between layers.
The rst case usually entails an exchange of information between protocols

to enable them to work jointly towards a specic goal.
The second case requires a redesign of the system architecture. This
redesign allows layers to exchange implicitly information by, for example,
mapping the functionality of one layer into a queue of an adjacent layer,
without the need for cross-layer signaling. There is no actual exchange of
information between layers: the trac passing through a queue provides
sucient in-band information for the cross-layer method.
There are many mechanisms that display these properties and which have
already been standardized, although these were not considered cross-layer
approaches, since the term was not then dened. One possible example
of cross-layer design is Random Early Detection (RED) that was initially
proposed in 1993 [3]. The on-going standardization of cross-layer design will
allow a better understanding of current schemes and cleaner approaches for
future systems.
10.3 Cross-layer methodologies for satellite systems
The following sub-Sections provide a classication of cross-layer methodolo-

gies, based on a review of current literature and the work that has been
presented in the previous Chapters.
Chapter 10: CROSS-LAYER METHODS AND STANDARDIZATION 315
10.3.1 Implicit and explicit cross-layer design methodologies
An important aspect for dierentiating cross-layer methods, that was high-

lighted in Chapter 1, was the presence or absence of signaling of the internal
protocol state between protocol layers. This may be used as a basis to
dierentiate between implicit and explicit cross-layer design/techniques, as
summarized below.
In an implicit cross-layer design, cross-layer interactions are taken into
consideration during the design phase, but there is no exchange of control
information among dierent layers during operation. Layers are designed
to complement each other and unnecessary duplicated functions can be
eliminated. For example, the objective may be to prevent MAC-level collisions
in the case of network ooding, or to apply a retransmission policy at the
link-layer that is aware of the requirements and behavior of the transport
layer protocols.
An explicit cross-layer design requires the exchange of additional control
information between dierent layers during operation. This method can be
used to tailor dynamically the operation and/or performance of the various
protocol entities, for example to signal periods of outages to higher layers, or
expected link capacity requirements to the lower layers.
10.3.2 Cross-layer techniques categorized in terms of the direction

of information ow
Another cross-layer classication method considers the direction of the cross-

layer information ow [4]. This approach is appropriate to an explicit cross-
layer design and primarily focuses on optimizing the information ow. Such
an approach should allow ecient ad hoc optimizations for each layer and/or
protocol, compatible with future versions of current protocols. Moreover, it
could provide an optimized cross-layer mechanism that could be used for
dierent kinds of optimization, rather than dening isolated cases that are
optimized for a particular communication system.
Developing an integrated cross-layer framework may be important to
the satellite community, since it not only leads to improved multimedia
performance over existing networks, but could also provide valuable insights
into the design of next-generation algorithms and protocols for satellite
multimedia systems.
This approach does not follow a traditional layered design. History has
shown that devoting time to build a solid framework (like the OSI refer-
ence model) failed, when the more integrated TCP/IP protocol stack has
succeeded. However, if the Internet continues its current gradual evolution,
this may be too slow to be able to satisfy the immediate needs for cross-layer
satellite optimizations.
One criterion for the evaluation of cross-layer methods is the eciency,
i.e., a ow of information would be considered more ecient if a maximum
of information is available to other layers when passing a minimum set of

parameters or signaling. Another criterion is the evaluation of the chosen ad
hoc performance parameter that benets from the information ow.
The impact on system design is a key constraint when designing to achieve
ecient information ow. A cross-layer approach does not necessarily require
a re-design of existing protocols, and can be performed by selecting and jointly
optimizing the upper layers and the strategies available at the lower layers,
such as admission control, resource management, scheduling, error protection,
power control, etc.
Bottom-up approach
A bottom-up method seeks to design an ecient information ow among layers

from the lowest layers up to the application. This approach can be appropriate
to a satellite system, implementing Adaptive Coding and Modulation (ACM),
since it would enable upper layers to be informed of the dynamics and
adaptation that is taking place at the physical layer. This cross-layer solution
may be less optimal for multimedia transmissions over terrestrial wireless
systems, due to the delays incurred with respect to the fast variations of the
radio channel conditions. However, in broadband satellite communications,
slow channel variations can allow a bottom-up method to signal upper layers
in time for them to react.
This approach requires dening general per-layer parameters that could be
useful to the upper layers. Moreover, protocols operating at each layer should
be reviewed assuming that all cross-layer parameters owing up from lower
layers are (instantaneously) available.
One serious issue is that of nding appropriate parameters that application
developers will wish to utilize in the design of their applications. The wide
variety of dierent environments in which modern Internet applications op-
erate makes it unattractive to tune applications to specic types of networks
(WiFi, WiMAX, satellite, ber-channel, etc.), although one could envisage
the communication of common transmission path characteristics (e.g., an
indication of path change, of QoS change, etc.) in the same way that network
stacks currently respond to indications of congestion or network-reachability
information.
Top-down approach
A top-down approach designs an ecient information ow among layers

from the application layer down to the physical layer. This can be seen as
an application-centric approach: applications indicate their expectations of
required network behavior, and lower-layers can then use this information to
optimize lower layer parameters.
There are drawbacks with this approach. One problem is that applications
are frequently unaware of the network paths over which they operate. They
are therefore unable to express their requirements in a way that maps easily
to the capabilities of specic lower-layers. Moreover, applications typically
operate over longer time-scales with coarser data granularities (multimedia
ows or blocks of data) than those used at lower layers (operating on bits
or frames). It is therefore non-trivial to perform adaptive source-channel
coding tradeos, given the time-varying channel conditions and the fact that
multimedia applications cannot be expected to adapt instantaneously their
behavior to achieve an optimal performance.
While lower layers can benet from notications of requirements (capacity
estimates, delay bounds, FEC/ARQ needs, priority, etc.) this does not provide
a complete solution. For example, it has limited benet for a satellite system
implementing ACM, since the upper layers may not be able to inuence
usefully the behavior of lower layers, rather, the channel dynamics require
upper layers to adapt themselves.
Hybrid approach
There are cases in which system level constraints are rened in a top-down
fashion, while the target architecture performance is abstracted in a bottom-
up fashion and a meet in the middle approach decides the nal optimization.
In this case, strategies are determined by exhaustively trying/combining all
the possible techniques of both the top-down type and the bottom-up one; the
aim is to achieve the best performance. This presents the highest exibility in
design choices.
However, this hybrid approach can have draw-backs. Constraints on the
design will often prevent an exhaustive analysis of all the possible strategies
(and their parameters) to choose an optimized composite strategy that
would lead to the best possible performance. When designing a cross-layer
methodology, general software architecture principles, such as information
hiding, modularization, and separation of concerns should be considered. A
hybrid approach also poses challenges to design.
10.4 Potential cross-layer optimizations for satellite

systems
This Section provides a summary of the set of cross-layer optimizations for
the satellite systems presented throughout the previous Chapters of this book.
The summary is presented ordered by scenario.
10.4.1 Optimizations aiming at QoS harmonization across layers
This sub-Section addresses aspects of QoS harmonization across layers for

multimedia trac in IP networks that contain a GEO satellite node. Two
approaches have been investigated, as summarized below:
MAC resource utilization optimization to support IP QoS (see

Section 8.3). Current IP QoS frameworks are considered (i.e., IntServ and
DiServ) to investigate how to manage the available resources (layer 2) in
an IP-based satellite network. The aim is to be as compliant as possible
with a predened QoS specication. The envisaged system is based on
Scenario 2 dened in Chapter 1, Section 1.4.
Optimization of resource sharing mechanisms at transport layer
(see sub-Sections 3.4.1 and 3.4.2). In this case, the optimization is per-
formed at the transport layer referring to the delayed real-time service
(streaming services), an interesting application, employing buers to add
an articial delay at the beginning of the play-out, so that a recovery
procedure can be started when data is lost. The aim is to enhance
the legacy satellite broadcast service with a specic multicast recovery
algorithm. The envisaged scenario employs GEO satellites (Scenario 2 in
Chapter 1, Section 1.4).
10.4.2 Optimization of the Radio Resource Management
Radio Resource Management (RRM) optimization normally involves the

physical and MAC layers. However, the selection of RRM techniques (and
the consequent optimization techniques) strongly depend on the envisaged
scenario. The following techniques were presented in previous Chapters for
a GEO scenario with xed users (i.e., Scenario 2 in Chapter 1, Section 1.4).
Results include:
Parametric optimization of bandwidth allocation strategies for

TCP connections (see Section 9.3). This study addresses bandwidth
allocation to TCP connections sharing a satellite bottleneck based on the
cross-layer adaptation of bit-rate and coding rate. A cross-layer method is
used to coordinate the actions at the satellite link physical layer (where
the fade countermeasure technique is applied) as well as at the MAC layer
(where the satellite bandwidth is allocated) to optimize the TCP goodput;
long-lived TCP connections are considered.
Bandwidth allocation strategies and dynamic bandwidth seg-
mentation algorithms to maximize fairness and the net satellite
return capacity (see sub-Section 7.3.4). This study assumes that the
total available resource is dened as a region in the time-frequency plane,
i.e., a MF-TDMA frame. ACM and/or Dynamic Rate Adaptation are
assumed. It seeks to design jointly the bandwidth segmentation, the time
slots duration and the bandwidth allocation to users in a way that provides
maximum fairness and eciency in the use of the return link (DVB-RCS
system).
The following technique was presented in a previous Chapter with results
for a GEO mobile scenario (i.e., Scenario 1 in Chapter 1, Section 1.4):
Scheduling strategies for satellite HSDPA transmissions (see sub-

Section 5.3.2). The aim of this study is to investigate the applicability
of the HSDPA air interface in the GEO satellite case, characterized by
high propagation delays, but also by slow channel variations. Suitable
scheduling techniques are investigated to guarantee the QoS support for
multimedia trac ows sent to mobile users.
The following technique has been proposed in Chapter 6 along with results
for a LEO mobile scenario (i.e., Scenario 3 in Chapter 1, Section 1.4):
Handover and Call Admission Control (see sub-Section 6.4). An
inter-satellite scenario with satellite diversity in a multimedia LEO satellite
system has been investigated. The interest is in identifying handover
techniques that are able to fulll the requirements in terms of both call
blocking probability (new calls) and call dropping probability (calls in
progress).
10.4.3 Optimizations combining higher and lower layers

A set of approaches summarized below have been investigated, that yield
cross-layer methodologies and optimization involving MAC and IP and/or
transport layers.
Bandwidth allocation taking into account TCP behavior (see
Section 9.4). TCP results in a bandwidth request that is time-dependent,
following the slow start and congestion avoidance mechanisms. A xed
RRM allocation strategy can lead to either wastage of resources, if dimen-
sioned to the maximum, or inability to satisfy transient requirements. This
study has proposed a TCP-driven RRM scheme that allocates resources by
taking into account the behavior of the TCP congestion window (internal
state of the TCP protocol) for each ow. In this case, Scenario 2 has been
considered. The results obtained highlight that the proposed cross-layer
RRM scheme can improve the performance at the TCP level.
Protocol integration between Ethernet-like layer-2 and satellite-
specic MAC (see Section 8.6). This optimization focuses on the en-
capsulation of MAC frames dened in the IEEE 802 project into satellite
MAC frames for a LEO-based satellite system. Additionally, the reuse of
bridging concepts as dened in IEEE 802.1 and the use of an extended
LLC sub-layer (derived from IEEE 802.2) is considered to harmonize the
satellite-dependent levels and Ethernet technologies.
Optimization of the bandwidth provision at the SD layer consid-
ering QoS management issues arising at the SI-SAP interface (see
Section 8.4). This optimization refers to a general satellite network with
multiple trac classes. This study has provided a technique that denes a
resource allocation at layer 2 on the basis of the requirements of IP-based
trac (layer 3 queue are also considered) in terms of IP packet loss rate
and IP packet average delay.
Optimization of higher layer coding for an ecient delivery of

multimedia contents across hybrid wireless networks (see Section
8.5). This work is related to the optimization of the packet loss perfor-
mance across multiple wireless access networks (i.e., satellite, and wireless
networks) that are characterized by dierent radio channel conditions
and, hence, by dierent packet loss patterns. Erasure codes (FZC) are
proposed for implementation between the transport and network layers.
The introduction of FZC allows a packet loss rate reduction without
introducing additional delays.
A summary of the above cross-layer optimization proposals is provided in
Table 10.1 below.
Scenario Users Above Layers Main Requirements

Section Involved Optimization
2 FIXED 10.4.1 2,3 Eciency DiServ, IntServ
2 FIXED 10.4.1 3,4 QoS DiServ, IntServ
2 FIXED 10.4.2 1,2,4 TCP goodput Parametric
optimization ACM
2 FIXED 10.4.2 1,2 Throughput, Parametric
bandwidth optimization ACM
segmentation
1 MOB 10.4.2 1,2,3 Transport format, ACM of
layer 3 QoS HSDPA type
3 MOB 10.4.2 2 Call blocking LEO multimedia
and call dropping sat. system
1 FIXED 10.4.3 2,4 TCP goodput DVB-RCS
3 FIXED 10.4.3 2 service disruption IEEE 802, LLC layer
due to LEO network extended to a
topology change satellite scenario
General FIXED 10.4.3 2,3 Resource allocation SI-SAP
sat. syst. optimization
1-like MOB 10.4.3 1,2,4 & Packet loss FZC codes
above
Table 10.1: Summary of cross-layer approaches for satellite systems addressed in

this book, in the order of appearance in the previous sub-Sections 10.4.1 - 10.4.3.
10.5 Cross-layer signaling for satellite systems

A number of signaling methods have been proposed that may carry cross-layer
information between layers. According to [5], four dierent techniques can be
considered (see Figure 10.1), as described below.
Fig. 10.1: Signaling (a) based on packet headers, (b) based on ICMP, (c) based on
a network service, (d ) based on local proles.
Method based on packet headers
A packet header method uses IP data packets as in-band message carriers

with no need to use a dedicated internal signaling protocol [5]. An IP packet
normally can only be processed layer-by-layer, and it is not easy for higher
layers to access the IP-level header. This method can be visualized as a
signaling pipe (see Figure 10.1a).
Method based on Internet Control Message Protocol (ICMP)
ICMP is a widely-deployed signaling protocol in IP-based networks. In com-

parison with the signaling pipe previously described, this method tries
to punch holes in the protocol stack and propagates information across
layers by using ICMP messages (see Figure 10.1b). In this system, desired
information is abstracted to parameters, measured by corresponding layers.
A new ICMP message is generated only when a parameter exceeds a suitable
threshold. Cross-layer communications are provided through selected holes
and not through a general pipe. For this reason, this method seems
more exible and ecient; moreover, it is more mature since it has been
already implemented in the Linux operating system with suitably-developed
Application Programming Interfaces (APIs). However, an ICMP message is
always encapsulated in an IP packet and, hence, such message has to pass
by the network layer even if the signaling is only desired between link layer
and application. Utilizing ICMP messages generated within a network also
requires extreme care, since this creates a vulnerability to denial-of-service
attacks, and can also lead to confusing indications in cases where not all
packets follow the same path through a network.
Method based on a network service
In this scheme, channel and link states from the physical layer and link
layer are collected, abstracted and managed by third parties, i.e., distributed
servers (e.g., Wireless Channel Information, WCI, server, see Figure 10.1c).
Interested applications access the servers for their required parameters from
the lowest two layers. Even if there is not a cross-layer signaling scheme
within a terminal, this is a complementary solution to the two above pre-
sented schemes. Nevertheless, intensive use of this method could introduce
considerable signaling overhead and delay over a radio access network.
Method based on local proles
In this approach, local proles are used on end-hosts to store periodically up-
dating information: cross-layer information is abstracted from each necessary
layer and stored in separate proles. Other interested layer(s) can select the
prole(s) to obtain the desired information as shown in Figure 10.1d.
10.6 Standardization issues

Some of the mechanisms summarized in Section 10.5 may be ready for
standardization in the short-term (a few years). Initial discussions are already
proceeding in many organizations, but a generalized framework and higher-
layer interactions will require more substantial research before standardization
may start.
A wide range of organizations participating in standardization of com-
ponents of the satellite systems has been identied. In particular, we can
refer here to the Internet Engineering Task Force (IETF) [6] at the transport
layer (e.g., TCP, UDP, RTP, QoS), to organizations such as the European
Telecommunications Standards Institute (ETSI) [7] and the International
Telecommunication Union - Telecommunication sector (ITU-T) [8] that deal
with the network layer and QoS, to satellite/broadcast/mobile fora and
organizations that focus on lower layer functions (e.g., DVB, SatLabs, 3GPP).
Below the network layer, mobile and broadband systems have traditionally
been standardized by dierent organizations (e.g., 3GPP, DVB-Forum) or by
dierent areas within the same organization (e.g., ETSI Broadband Satellite
Multimedia -BSM-, ETSI Satellite-UMTS, S-UMTS). This is likely to continue
for cross-layer methods. It is therefore important to disseminate information
about key issues, available options and requirements to these groups in
preparation for future standardization work. Cross-fertilization of ideas and
results may be of benet to both mobile and broadband communities, allowing
them to converge (perhaps using the common trac classes established for
QoS interworking).
Above the network layer, there has been little attention paid to the
demands and benets of cross-layer optimization. Recent work within the

Internet Architecture Board (IAB) suggests that after much exploration of
the issues, IETF is starting to understand the architectural issues, and
standardization within appropriate IETF working groups could follow in the
longer-term.
The multi-disciplinary nature of cross-layer approaches, not only compli-
cates the analysis of the system, but is expected to pose also practical problems
to standardization. To be successful, standards for cross-layer mechanisms
and interfaces will require close liaisons and information ow between these
organizations on both technical and architectural issues. Such close liaisons
are complicated by diering terminology and by dierent standardization
processes employed, and are not the current norm. Standardization of cross-
layer methods will therefore pose its own challenges. A rst example of
standardization of cross-layer methods can be represented by encapsulation
allowing adaptive coding in DVB-S2, a work recently started within DVB-
Return Channel via Satellite (DVB-RCS), DVB-Global Broadcast Service
(DVB-GBS) and IETF working groups.
The following Sections introduce the key standardization bodies and
groups relevant to the eventual standardization of cross-layer methods for
satellite systems [9].
10.6.1 Standardization bodies and groups
The groups of interest in connection with standardization activities that could

be related to cross-layer issues are listed in Table 10.2.
10.6.2 European Conference of Postal and Telecommunications

Administrations
The European Conference of Postal and Telecommunications Administrations

(CEPT), through its permanent European Radiocommunication Oce (ERO),
is a body of policy-makers and regulators with 44 country members covering
almost the entire geographical area of Europe. Within CEPT, the Electronic
Communications Committee (ECC) is responsible amongst others for devel-
oping policies on electronic communication activities in a European context
and for harmonizing within Europe the ecient use of the radio spectrum.
10.6.3 ETSI
The objective of ETSI [7] is to produce and perform the maintenance of

the technical standards and other deliverables which are necessary to achieve
a large, unied European market for telecommunications and related areas.
ETSI is an independent, non-prot organization, based in Sophia Antipolis
(France). The principal role of ETSI is technical pre-standardization and
standardization at the European level in the following elds:
Organization Working group Sub-group Layers

IETF - - L2-L7
3GPP - - L1-L2
CEPT - - L1
CEN/CENELEC - - L1-L2
ETSI TC-SES S-UMTS L1-L3
ETSI TC-SES SDR L1-L2
ETSI TC-SES BSM L2-L3
DVB DVB-CM - L1-L2
DVB DVB-TM - L1-L2
DVB DVB-RCS - L1-L2
DVB DVB-S2 - L1
ITU-R SG4 WP 4 A L1
ITU-R SG4 WP 4 B L1
ITU-R SG6 - L1
ITU-R SG8 WP 8 D L1
ITU-R SG8 WP 8 F L1
WorldDAB - - L1-L3
GBSI-ITSO Standards and - L1-L3
Regulatory Groups
Table 10.2: Standardization fora of interest for cross-layer issues.
Telecommunications.
Information and communication technology in co-ordination with the
European Committee for Standardization (CEN) and the European Com-
mittee for Electro-technical Standardization (CENELEC).
Areas common to telecommunications and broadcasting (especially audio-
visual and multi-media matters) in co-ordination with CEN, CENELEC
and the European Broadcasting Union (EBU).
The ETSI Technical Committee for Satellite Earth Stations and Systems
(TC-SES) is responsible for all types of satellite communication services
(including mobile and broadcasting) and for all types of Earth station equip-
ment (especially the radio frequency interfaces and network and/or user
interfaces). It maintains an internal liaison with the ETSI EMC and Radio
spectrum Matters (ERM) working group (for electromagnetic compatibility
issues and radio spectrum matters), with the ETSI Special Mobile Group,
SMG (for GSM and S-UMTS), and with the working group TM4 of the
ETSI Technical Committee Transmission & Multiplexing, TM (for xed radio
links). TC-SES also maintains external liaisons with other bodies, including:
ITU-R (SG4 on Fixed Satellite Services, JWP10-11S on satellite broadcasting,
WP 8 D on Mobile Satellite Services, TG8/1 on IMT-2000), CEPT-ERO and
the European Co-operation on Space Standardization (ECSS). Many of the
standards produced by the TC-SES are relevant to mobile satellite systems,
broadcasting satellite systems and hybrid networks, comprising satellite and

terrestrial infrastructures.
The ETSI TC-SES S-UMTS working group oversees the Satellite compo-
nent of the UMTS as part of the International Mobile Telecommunications
(IMT-2000) standard. It is the ETSI focal point for liaising with the other
bodies for the development of standards on S-UMTS/IMT2000. S-UMTS
systems will complement Terrestrial UMTS (T-UMTS) and interwork with
other IMT-2000 family members through the UMTS core network. S-UMTS
mobile satellite services will be delivered utilizing either Low or Medium Earth
Orbit (LEO, MEO), or Geostationary (GEO) satellite(s). One of the main
objectives of the S-UMTS working group is to enforce a signicant level of
compatibility with T-UMTS in order to minimize user terminal modications
required to receive S-UMTS mobile satellite services. Three main directions
are currently explored:
The adaptation of the 3GPP W-CDMA specications to satellite;
The adaptation of the 3GPP Multimedia Broadcast Multicast Service
(MBMS) specications to satellite;
The analysis of the feasibility of an OFDM air interface for mobile satellite
networks.
The S-UMTS Family G specication set aims at achieving the satellite
air interface fully compatible with W-CDMA-based systems. However, due
to the dierences between terrestrial and satellite channel characteristics, not
all the T-UMTS specications are directly applicable, but some of them need
modications. Family G has been released as a multipart standard consisting
of the following six documents (1 ) specic to the satellite air interface:
Part 1, Physical channels and mapping of transport channels into physical
channels (S-UMTS-A 25.211), denes transport channels and physical
channels [11];
Part 2, Multiplexing and channel coding (S-UMTS-A 25.212), describes
multiplexing and channel coding [12];
Part 3, Spreading and modulation (S-UMTS-A 25.213), species spread-
ing and modulation [13];
Part 4, Physical layer procedures (S-UMTS-A 25.214), describes physical
layer procedures [14];
Part 5, UE Radio Transmission and Reception, establishes the minimum
RF characteristics for the user equipment [15];
Part 6, Ground stations and space segment radio transmission and
reception, describes the space segment RF characteristics [16].
The TC-SES S-UMTS technical activity related to Satellite MBMS (S-
MBMS) is based on the design of a Satellite Digital Multimedia Broadcasting
1
Part 1 through part 4 are based on their counterparts developed within 3GPP
for terrestrial UMTS in frequency division duplexing mode.
(S-DMB) system [17]. This oers a unidirectional point-to-multipoint bearer

service from a single source entity (satellite) to multiple recipients using broad-
cast or multicast mode. S-MBMS is dened by six specications currently
under approval within TC-SES S-UMTS.
Finally, TC-SES S-UMTS is studying OFDM as a possible satellite air
interface. OFDM techniques are being used by several digital broadcast
terrestrial systems and are characterized by high spectral eciency. These
techniques have been considered for 3G air interfaces, but entail technical
challenges due to both the rather high peak-to-average power ratio and the
non-linear distortion induced by the on-board High Power Amplier (HPA).
The TC-SES S-UMTS studies show that with ad hoc pre-distortion techniques
and turbo coding the eect of the HPA non-linear distortion drastically
reduces, thus allowing for the adoption of OFDM on satellite air interfaces
[18].
Although current specications and study items do not explicitly deal with
cross-layer aspects, the need for TC-SES S-UMTS to maintain compatibility
with the T-UMTS system evolution will indeed require opening new work
items related to the development of capacity-improving techniques, such as
interference mitigation, multi-user detection, and macro- and micro-diversity
algorithms. TC-SES S-UMTS is a crucial working group regarding interests
for cross-layer activities on S-UMTS. Its liaisons with organizations outside
ETSI include: 3GPP, ITU-R SG8 WP 8 D, and ITU-R SG8 WP 8 F (for
IMT-2000 and systems beyond).
10.6.4 DVB
The DVB Project was initiated in 1992 [19] and has subsequently imple-
mented an approach of pre-competitive co-operation in the development
of open digital TV standards that can be freely adopted worldwide. The
motivation was to promote a common, standard, European platform for
digital TV broadcasting, and the idea was supported by all players (i.e.,
broadcasters, operators, standardization bodies, media groups and industry).
Today, DVB has 220 members from more than 30 countries worldwide. By
incorporating both commercial and technical bodies within the organization,
DVB has succeeded in delivering transmission standards for television systems
operating over a range of media, including DVB-S, DVB-C and DVB-T
standards. The advent of interactive networks stimulated the standardization
of Return Channels for Cable (i.e., DVB-RCC), Satellite (i.e., DVB-RCS),
Local Multipoint Distribution System, LMDS (i.e., DVB-Return Channel for
LMDS, DVB-RCL), and Terrestrial (i.e., DVB-RCT) systems.
The work in the DVB technical area is organized in ad hoc groups. Each of
them works on commercial requirement documents provided by the Commer-
cial Module. This is a set of user requirements that outline market parameters,
such as user functions, timescales and price range. A DVB specication is
developed in the Technical Module and its working groups, where technological
implications of user requirements are examined and available technologies

are explored. Once the Technical Module reaches consensus on the resulting
specication, and the Commercial Modules support for it has been ensured,
the Steering Board is solicited to give the nal approval. It is then oered for
standardization to ETSI or CENELEC through the EBU/ETSI/CENELEC
Joint Technical Committee as well as sometimes to ITU-T or ITU-R.
The main DVB standards that are relevant to satellite communications
are considered below.
DVB-RCS
The Digital Video Broadcasting-Technical Module (DVB-TM) created an ad

hoc group early 1999, called DVB-RCS, which lead to specication ETSI EN
301 790 [20]. This document species a satellite terminal known as a Satellite
interactive Terminal (ST) or Return Channel Satellite Terminal (RCST) that
supports a two-way DVB satellite system.
The return link in DVB-RCS uses an MF-TDMA air interface where STs
have allocated capacity in slots within a certain time-frequency structure.
The entire system is controlled by a Network Control Center (NCC) (e.g., at
the Gateway side of the satellite) controlling the ST behavior. The NCC is
responsible for synchronization of the system, via the Network Clock Reference
(NCR), and sends out a number of specic system tables in order to give the
STs all the information needed for receiving and transmitting in the system.
This includes, in addition to the tables already existing in the DVB-S system,
tables informing on frame composition, capacity allocation, regulation of ST
timing and frequency osets, etc.
The DVB-RCS standard adopts the DVB-S standard for the forward link,
that uses a Time Division Multiplexing (TDM) carrier, usually with practical
data rates on present Ku band transponders ranging up to tens of Mbit/s.
The second generation of DVB-S, DVB-S2 (see later) is also compatible with
the DVB-RCS standard.
The DVB-RCS standard does not currently include any specic features
for mobility management, and this issue is a current research topic and a
standardization target. To achieve this, it is important to devise a robust
variation of the DVB-RCS return link to support high-speed mobility as
maintaining synchronization after acquisition. More details are provided in
a further sub-Section.
Finally, the interoperability issues between DVB-RCS terminals and net-
works are addressed by the SatLabs Group, an international, non-prot
association whose members are interested in promoting two-way satellite
networks based on the DVB-RCS open standard. The SatLab Web site
[21] contains a wide collection of documents (some of them have a public
access) dealing with specications, recommendations and technical issues.
The SatLabs qualication programme was dened to achieve DVB-RCS
interoperability testing and certication. The SatLabs Group is lead by ESA
with the participation of many manufacturers, operators and service providers

in the eld of satellite communications.
DVB-S2
The DVB-S2 [22] standard for satellite transmission supports ACM, which
enables high data-throughput eciency. ACM is applicable in networks where
a return channel allows transmission of information concerning the reception
quality from the satellite receiver to the satellite uplink station. The standard
denes the reception quality parameter and its binary coding. The transport
of this parameter back to the uplink station is not in the scope of the standard
and is specied separately for the dierent return channel systems. This has
been done already for satellite return channel in the current release of the
DVB-RCS standard [20].
Another potential application for ACM in DVB-S2 is hybrid satellite-
terrestrial networks for high-speed Internet access. In this kind of networks, a
user terminal receives data over satellite and transmits data over a terrestrial
dial-up connection. The more ecient use of satellite capacity could make
such hybrid networks more attractive and therefore enable a larger market for
DVB-S2 receiver chips with the interactive services prole implemented.
Applicability of DVB-S2-like ACM as a countermeasure to fading due to
terminal mobility is also a possibility. ACM does not help against the fast
fading that occurs in land mobile scenarios due to multipath and, further,
against typically short shadowing and blocking events. The adoption of ACM
in DVB-S2 is intended to counteract rain fading; therefore, it is important
to investigate how terminal mobility changes the time variability of rain fade
events and, hence, the eciency of ACM, e.g., when a car or a high-speed
train travel through a rain cell.
DVB-S2 extension for mobile usage

Current expectations of users are to access the Internet and to receive
multimedia contents while on the move. This is the reason way there is
interest in evolving the DVB-S2/-RCS standard to allow the mobile usage
(possible scenarios are: users on plains, trains and in land masses). This
extension need to address many challenging issues such as [23]: stringent
frequency regulations (Ku band), Doppler eect, frequent handovers, and
impairments in synchronization acquisition and maintenance. In addition to
this, the railway scenario is aected by shadowing, fast fading (due to mobility,
there are deep and frequent fades caused by the poles of the electried lines)
and long blockages (presence of tunnels and large train stations with non-LoS
propagation conditions to the satellite). The new standard should address
important issues that are outlined below.
Spectrum spreading techniques: the stringent regulations for Ku band
mobile terminals require a careful study for the possible use of spreading
techniques, especially in the return link for terminals with small antennas.
The adoption of spectrum spreading is a possible solution to reduce the
EIRP, while preserving the required SNR, at the expenses of reduced
spectral eciency. In the forward link, the introduction of spreading
requires the design of a new DVB-S2 receiver. In the return link, each
terminal could in principle implement direct spreading within the assigned
time and frequency slot (MF-TDMA approach).
Fading countermeasures: the more challenging propagation conditions of
the non-LoS scenario can be mitigated by adopting advanced techniques
such as diversity and higher layer FEC schemes. Moreover, new synchro-
nization acquisition and maintenance procedures need to be employed to
cope better with frequent fades.
Resource management techniques: ecient RRM schemes need to be
adopted to account for mobility, such as: impact of spreading on the
MF-TDMA allocation process (DVB-RCS); support of handover requests
with suitable protocols; interworking with terrestrial networks in shadowed
areas (e.g., tunnels, cities, etc.) where gap llers can be used; adaptive
schedulin techniques for the forward link that are aware of the physical
layer behavior.
All these innovative aspects require a cross-layer system design aiming at

optimizing the choices made at dierent layers. The DVB-TM is now working
to specify the modications that are needed for the mobile extension of the
DVB-S2 standard [23]. The SatNEx II project [24] is actively involved in this
standardization process.
DVB-H
The broadcast of digital television signals was originally targeted to xed

reception, although mobile reception is also feasible with current digital
television standards (DVB-T, DVB-S2). The Commercial Module of DVB
decided to launch commercial requirements for the production of ad hoc
specications able to provide broadcasting to one specic niche of the mobile
receivers: handheld terminals. This is the aim of the DVB-Handheld (DVB-H)
standard.
Conditional access is important in all broadcast radio/satellite networks
to prevent unauthorized access to the broadcast content by eaves-dropping.
In DVB-H, an IP-based Conditional Access System (IP-CAS) can provide
link-layer encryption (scrambling) for DVB-H services. CAS messages are
delivered over IP and may take advantage of time-slicing to save power at
a receiver. The DVB common scrambling algorithm on Transport Stream
packets is also employed (DVB-CAS): it uses entitlement control messages to
send keys to receivers and entitlement management mode messages to deliver
management messages.
10.6.5 International Telecommunication Union
ITU is an international organization of the United Nations where governments

and industries coordinate global telecom networks and services. ITU is divided
in three sectors: ITU-T that aims at the denition of high-quality standards
covering all elds of telecommunications; ITU-R that plays a fundamental role
in the management of the radio-frequency spectrum, physical layer issues, and
satellite orbits; and ITU-D, dealing with Telecommunications Developments.
ITU-R is charged with determining the technical characteristics and
operational procedures for a huge and growing range of wireless services.
This Sector also plays a vital role in the management of the radio-frequency
spectrum, a nite natural resource that is increasingly in demand due to the
rapid development of new radio-based services and the enormous popularity
of mobile communication technologies.
In its role as global spectrum coordinator, ITU-R develops and adopts
the Radio Regulations, a voluminous set of rules that serve as a binding
international treaty governing the use of the radio spectrum for dierent
services around the world. ITU-R also acts, through its Bureau, as a central
registrar of international frequency use, recording and maintaining the Master
International Frequency Register, which currently includes around 1,265,000
terrestrial frequency assignments, 325,000 assignments servicing 1,400 satellite
networks, and another 4,265 assignments related to satellite Earth stations.
Moreover, ITU-R is responsible for coordinating eorts to ensure that commu-
nication, broadcasting and meteorological satellites in the worlds increasingly
crowded skies can co-exist without causing harmful interference each other.
The Union facilitates agreements between both operators and governments,
and provides practical tools and services to help frequency spectrum man-
agers.
The portion of the radio-frequency spectrum suitable for communications
is divided into blocks, the size of them varying according to individual
services and their requirements. These blocks are called frequency bands
and are allocated to services on an exclusive or shared basis. The full list of
services and frequency bands allocated in dierent regions forms the Table of
Frequency Allocations, which is a part of the radio regulations.
10.7 Conclusions
A range of cross-layer optimization techniques have been proposed and

evaluated in this book for three dierent scenarios (i.e., DVB-S/DVB-RCS
via GEO bent-pipe satellite, S-UMTS via GEO bent-pipe satellite, and LEO
constellation with regenerating satellites). The most signicant techniques
have been summarized in this Chapter to provide nal guidelines for both
standardization eorts and further research directions.
Cross-layer methods have been categorized, considering: (i ) either explicit
signaling or an implicit scheme with a joint optimization of dierent protocol

layers; (ii ) the denition at higher layers of requirements to be used for ap-
propriate settings at lower layers or, vice-versa, the lower layers progressively
determining the requirements at higher layers. As for explicit cross-layer, we
have described dierent mechanisms for the exchange of internal protocol state
information between non-adjacent protocol layers, thus violating the classical
ISO/OSI layered philosophy.
We have proved that the cross-layer techniques can improve the overall
end-to-end quality of service, while optimizing the eciency in utilizing the
scarce satellite radio resources. However, standardization fora have not yet
signicantly addressed cross-layer issues. To this aim, there is a need for a
new framework, as well as the strong cooperation of dierent standardization
bodies. One of the aims of this book has been to provide some useful insights
that may promote new standardization activities on cross-layer air interface
design for satellite communication networks.
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[3] S. Floyd, V. Jacobson, Random Early Detection Gateways for Congestion
Avoidance, IEEE/ACM Transactions on Networking, Vol. 1, No. 4, pp.
397-413, August 1993.
[4] M. van der Schaar, S. Shankar, Cross-Layer Wireless Multimedia Transmission:
Challenges, Principles, and New Paradigms, IEEE Wireless Communications
Magazine, Vol. 12, No. 4, pp. 50-58, August 2005.
[5] Q. Wang, M. A. Abu-Rghe, Cross-Layer Signalling for Next-Generation
Wireless Systems, in Proc. of IEEE Wireless Communications and Networking
Conference 2003 (IEEE WCNC 2003), New Orleans, USA, pp. 1084-1089,
March 2003.
[6] The Internet Engineering Task Force (IETF); Web page with URL:
http://www.ietf.org.
[7] European Telecommunications Standards Institute (ETSI); Web page with
URL: http://www.etsi.org.
[8] International Telecommunication Union; Web page with ULR:
http://www.itu.int/home/index.html.
[9] MoSSA, Advanced Satellite Mobile Systems-Task Force Specic
Support Action, Project IST-507557, Deliverable Survey on
Standardization and Regulatory Activities; Web site with URL:
http://asms1.wss.bcentral.com/mossa/default.htm.
[10] ETSI TC-SES working group; Web page with URL: http://portal.etsi.org/ses/.
UMTS/IMT2000; G-family; Part 1: Physical channels and mapping of transport
channels into physical channels (S-UMTS-A 25.211), TS 101 851-1.
UMTS/IMT2000; G-family; Part 2: Multiplexing and channel coding (S-UMTS-
A 25.212), TS 101 851-2.
UMTS/IMT2000; G-family; Part 3: Spreading and modulation (S-UMTS-A
25.213), TS 101 851-3.
UMTS/IMT2000; G-family; Part 4: Physical layer procedures (S-UMTS-A
25.214), TS 101 851-4.
UMTS/IMT2000; G-family; Part 5: UE Radio Transmission and Reception (S-
UMTS-A 25.101), TS 101 851-5.
UMTS/IMT2000; G-family; Part 6: Space Segment Radio Transmission and
Reception (S-UMTS-A 25.104), TS 101 851-6.
[17] IST-MAESTRO project, Mobile Applications & sErvices based on Satellite &
Terrestrial inteRwOrking; Web site with URL: http://ist-maestro.dyndns.org,
2006.
[18] ETSI, Evaluation of the OFDM as a Satellite Radio Interface Satellite Earth
Stations and Systems (SES); Satellite Component of UMTS/IMT-2000, TR
102 433, 2006.
[19] Digital Video Broadcasting (DVB) Project; Web page with URL:
http://www.dvb.org.
[20] ETSI, Digital Video Broadcasting (DVB); Interaction channel for Satellite
Distribution Systems, EN 301 790.
[21] SatLabs ocial Web site with URL: http://www.satlabs.org/.
[22] ETSI, Digital Video Broadcasting (DVB); Second Generation Framing
Structure, Channel Coding and Modulation Systems for Broadcasting,
Interactive Services, News Gathering and other Broadband Satellite
Applications, EN 302 307.
[23] S. Scalise, G. E. Corazza, C. P
arraga Niebla, P. Chan, G. Giambene, F. Hu, A.
Vanelli-Coralli, M. A. V
azquez Castro, Towards the Revision of DVB-S2/RCS
Standard for the Full Support of Mobility, SSC Newsletter, Vol. 17, No. 2,
November 2006.
[24] SatNEx II Web site with URL: http://www.satnex.org.
Index
A Cross-layer approach, 34, 156, 164, 256,

Access protocol, 120, 132 314
Adaptive algorithms, 209, 295 Bottom-up approach, 316
Adaptive coding and modulation, 16, Hybrid approach, 317
24, 106, 139, 208, 316 Top-down approach, 316
Asynchronous transfer mode, 109 Cross-layer design, 35, 36, 45, 96, 105,
214, 221, 256, 270, 313, 314
B Explicit cross-layer, 35, 36, 133, 145,
156, 164, 300, 315
Broadband satellite multimedia, 28, 31,
Implicit cross-layer, 35, 36, 45, 95,
69, 98, 256
217, 270, 290, 315, 317
Broadcast and multicast services, 5, 80,
152, 160
D
C Delayed real-time services, 83
Demand assignment multiple access, 18,
CAC, 45, 51, 100, 110, 177, 179, 184,
189, 199, 257 290, 298, 300
Complete partitioning, 52, 179 Access delay, 290, 298
Complete sharing, 52, 179 Rate-based dynamic capacity, 20,
Call handover, 53, 189, 195, 214, 233 211, 221, 249, 299
Inter-satellite handover, 54, 190, 194, Volume-based dynamic capacity, 20,
214 211, 221, 249, 299, 303
Intra-satellite handover, 54, 190, 191, DiServ, 36, 77, 107, 183, 246
214 DVB-S, 16, 80, 105, 187, 326
CDMA, 14 DVB-RCS, 17, 81, 186, 211, 249, 289,
Channel quality indicator, 138 298, 323, 327
Channel utilization, 162, 305 Adaptive coding, 218
Combined free/demand assignment Implementation issues, 218, 228
multiple access, 48, 101, 256 MF-TDMA scheme, 15, 17, 184,
Commercial solutions, 82, 89 211, 251, 302, 327
Communications architecture, 314, 320
Cross-layer signaling, 320 E
Congestion control, 164, 186 ETSI TC SES S-UMTS working group,
Control-theoretic approach, 213 15, 121, 325
336 Index
Explicit congestion notication, 107 Performance enhancing proxies, 29, 99,

293
F Power allocation and control, 50
Closed loop, 50
FDMA, 13
Feedback loop, 50
Open loop, 50
G Proactive algorithms, 210
GEO satellite systems, 4, 10, 68, 71,
125, 131, 141, 184, 209, 265, 290
Q
QoS classes, 156, 165
H QoS for multimedia services, 68
Handover algorithms, 51 Background services, 76
Handover queuing, 53 Conversational services, 70
Predictive resource reservation, 54 Interactive services, 73
HSDPA, 16, 108, 138, 139, 141, 144, 148 Performance requirements, 70, 73, 74,
Hybrid satellite networks, 265 76
Erasure codes, 265 QoS based IP models, 76
QoS, 266 Streaming services, 74
WiFi networks, 267 QoS mapping, 98, 256, 260
I R
Innitesimal perturbation analysis, 99, Radio resource management, 43, 54, 96,
216, 257 101, 119, 177, 289, 303, 318
IntServ, 36, 77, 107, 183, 244 Cross-layer approach, 45, 60, 96, 99,
101, 104, 214, 295, 303, 305
L Joint optimization, 95, 97100
LEO satellite systems, 4, 10, 54, 68, 71, MAC-centric approach, 105
132, 141, 189, 192, 195 Dynamic allocation, 20, 47, 49, 55,
99, 101, 110, 191, 198, 208, 211,
M 213, 214, 218, 233, 248, 251, 256,
299
MAC, 18, 97, 98, 105, 110, 119, 139,
248, 256, 298 Fairness, 44, 217, 232
MEO satellite systems, 4, 10, 48, 71, Reactive algorithms, 210
141 Receding horizon controller, 214
Modeling and simulation, 54 Resource allocation, 23, 46, 99, 121,
138, 158, 179, 184, 208, 225, 249,
299
N Frequency allocation, 46
NCC, 17, 178, 208, 298, 327 Space allocation, 46
Network layer, 243 Time allocation, 46
Node-B, 139
O S
S-UMTS, 15, 58, 108, 121, 131, 152, 325
OSI model, 34, 102 Satellite constellations/orbits, 4, 10, 275
Satellite digital multimedia
P broadcasting, 325
Packet scheduler, 134, 137, 140, 152, Satellite IP networks, 31, 69, 76, 109,
155, 164 248
Index 337
IP QoS, 76, 109, 183, 244, 248 MODCOD optimization, 294

Proportional DiServ, 249 TDMA, 13
Scheduling scheme, 134 Transport layer, 99, 289
Channel-aware scheduling, 135 Congestion control, 291
Exponential Rule scheduler, 138 TCP, 99, 273, 290, 294, 298
Maximum C/I scheduler, 138 UDP, 273, 290, 305
Proportional Fair scheduler, 138, 147
Service level agreement, 96, 222, 246 U
SI-SAP, 31, 98, 256 UMTS, 15, 58, 121
Smith predictor controller, 214
Standardization, 322
Static algorithms, 209
V
VLANs for LEO constellations, 270
Voice over IP, 50, 70, 262
T
TCP over satellite, 290 W
Cross-layer interactions, 294, 298 W-CDMA, 16, 46, 137

Resource Management in Satellite Networks

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What are some of the main applications of satellite communications discussed in the document?

What are some of the main applications of satellite communications discussed in the document?

What was the original idea for using satellites for telecommunications described in the document?

What was the original idea for using satellites for telecommunications described in the document?

Giovanni Giambene

Library of Congress Control Number: 2007922349

ISBN 0-387-36897-3 e-ISBN 0-387-53991-3

Printed on acid-free paper.

2007 Springer Science+Business Media, LLC

The volume Editor and the authors would like to acknowledge

through the SI-SAP (Satellite-Independent - Service Access Point) interface

I had the pleasure to coordinate the ja2430 activities, organizing 4 peri-

List of Contributors . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . xix

List of Acronyms and Abbreviations . . . . . . . . . . . . . . . . . . . . . . xxiii

Part I Resource Management Framework for Satellite

1.6.2 Vertical approach . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 34

2 ACTIVITY IN SATELLITE RESOURCE

3 QoS REQUIREMENTS FOR MULTIMEDIA

4 CROSS-LAYER APPROACHES FOR RESOURCE

Part II Cross-Layer Techniques for Satellite-Dependent Layers

5 ACCESS SCHEMES AND PACKET SCHEDULING

6 CALL ADMISSION CONTROL . . . . . . . . . . . . . . . . . . . . . . . . . . 177

6.5 Directions for further research . . . . . . . . . . . . . . . . . . . . . . . . . . . . 199

7 DYNAMIC BANDWIDTH ALLOCATION . . . . . . . . . . . . . . . 207

Part III Cross-Layer Techniques for Satellite-Independent Layers

8 RESOURCE MANAGEMENT AND NETWORK LAYER 243

8.7 Conclusions . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 282

9 RESOURCE MANAGEMENT AND TRANSPORT

10 CROSS-LAYER METHODS AND STANDARDIZATION

Rafael Asorey Cacheda Nedo Celandroni

Franco Davoli Javier Herrero S

Giada Mennuti Veronica Pasqualetti

3G 3rd Generation B-ISDN Broadband Integrated Services

CMF Control and Monitoring DVB-RCT DVB-Return Channel via

GPS Generalized Processor Sharing LEO Low Earth Orbit

PER Packet Error Rate SCr Service Credit

TTI Transmission Time Interval VPI/VCI Virtual Path Identier/

Resource Management Framework for Satellite

Editor: Giovanni Giambene1

Contributors: Paolo Chini1 , Giovanni Giambene1

1.1 Satellite communications

Support to mobile users: a mobile user, which is situated in the satellite

Very good books in the eld of satellite communications, providing excel-

Fig. 1.1: Description of satellite orbit types.

region or the whole Earth, thus forming a constellation.

L band from 1 to 2 GHz

These bands, composing the microwave spectrum, are actively used in

Note that L, S and C bands are already congested; X band is typically

of signicant signal impairment in the presence of bad wheatear conditions

Fig. 1.2: Basic satellite network architecture.

System Orbit type, Services Access Frequency

Table 1.1: Description of the characteristics of the main satellite communication

telephony and ISDN calls; Internet/Intranet connection; SMS and MMS;

System Orbit type, Services Access Frequency

Table 1.2: Description of the characteristics of the main satellite communication

of portable terminals. The enhanced system (BGAN-X, BGAN Extension

1.2 Basic issues in the design of satellite communication

Round Trip propagation Delay (RTD)

The eects of atmosphere (subdivided in troposphere and ionosphere) can be

Atmospheric gasses. Oxygen (dry air) and water vapor determine an