06 Tm51106en03gla3 Services
06 Tm51106en03gla3 Services
06 Tm51106en03gla3 Services
Contents
1
2
2.1
2.2
3
3.1
3.2
4
4.1
4.2
4.3
5
5.1
5.2
6
6.1
7
8
Module Objectives
3G evolution
3G service evolution
3G bit rate evolution
Services and Applications
Introduction to Mobile Applications
Service Platforms
Services and QoS classes
Services and Conversational QoS Class
Services and Streaming QoS Class
Services and Interactive/Background Class
Examples of UMTS Applications
Applications using Circuit Switched Service
Applications using Packet Switched Service
Appendix
Quality of Service QoS
Exercises
Solution
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2010 Nokia Siemens Networks
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2010 Nokia Siemens Networks
Module Objectives
The main objective of this module is to make the student able to explain the GSM
and UMTS mobile applications.
Topics to be covered in this module include the differentiation between UMTS
services and applications, a general discussion of the QoS, and an introduction to the
most Known services.
After completing this module, the participant should be able to:
TM51106EN03GLA3
2010 Nokia Siemens Networks
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2010 Nokia Siemens Networks
3G evolution
2.1
3G service evolution
Company data transmission (e.g. teleworking, mobile offices, virtual work groups,
etc.)
TM51106EN03GLA3
2010 Nokia Siemens Networks
2.2
GSM Phase 2+ GPRS (General Packet Radio Service) provides on one side a circuit
switched interface between the BSC and the MSC that corresponds to the normal
GSM Phase 2 circuit switched interface and on the other side a packet switched
interface between the BSC and the SGSN. GPRS uses 4 coding schemes, which are
different from the normal GSM Phase 2+. These coding schemes provide data-rates
between 9.05 kb/s up to 21.4 kb/s. In addition up to 8 channels can be combined at
the air interface. This results in a maximum data-rate of 171.2kb/s (theoretical). GSM
Phase 2+ EDGE (Enhanced Data rates for GSM Evolution) uses the GPRS
(8X21.4kb/s theoretical) for packet switched traffic and / or the HSCSD technology
(8X14.4kb/s theoretical) for circuit switched data transfer. In opposite to the other
GSM technologies, EDGE uses a different modulation on the air interface. With this
different modulation an approximately 3 times higher transmission rate can be
reached up to the maximum of 553.6kb/s (in combination with GPRS).
In the standard UMTS version data will be sent to the user equipment at a rate of up
to 2Mbps. In HSDPA (High Speed Downlink Packet Access), rates of 14,4Mbps may
be reached depending on the modulation (QPSK or 16-QAM) and coding scheme
used. Selection of the optimum modulation and coding scheme is performed based
on the downlink quality indicators provided by the UE.
TM51106EN03GLA3
2010 Nokia Siemens Networks
TM51106EN03GLA3
2010 Nokia Siemens Networks
3.1
In your everyday life, you are familiar with the concept of a mobile application.
For example, a mobile phone call is a mobile application. An SMS is another type of
mobile application. With time, the need for different types of applications is
increasing. Today the subscribers expect an increased number of applications and
greater value. For an operator with a large subscriber base, more usage time is one
way of ensuring continuing growth. However, the subscribers use mobile applications
on for a limited time. Therefore, when defining the 3 rd Generation (3G)
specifications, the emphasis is on the unlimited prospect of seamless services and
applications that can be provided.
One common misconception that people have is that applications have been
introduced in UMTS. However, this is not true as GSM already offers both integrated
network and Intelligent Networks (IN) applications. In today's networks, General
Packet Radio Service (GPRS) adds the facility of supporting packet data with
relatively quick set-up and transfer times, such as in the case of Internet.
UMTS Services
The term mobile application refers to services provided to the subscriber. Mobile
applications have not been standardized in UMTS. The GSM/UMTS network offers
service elements that are used by applications. The applications form the value
added for the subscriber. A set of services have been made available by UMTS,
which are:
Circuit Switched (CS) Services - These are the teleservices, such as speech call,
facsimile call, and CS data.
Packet Switched (PS) Services - These are based on the PS connectivity provided
by Packet Data Protocol (PDP) contexts.
Message Services - These include Short Message Service (SMS) and Multimedia
Message Service (MMS).
The services speech call, facsimile, and SMS are both services and applications. CS
data is only a service because the subscriber allocated with a CS bearer for data
transport. The bearer itself adds no value to the subscriber. The subscriber requires
the CS data bearer to run a data application, where content is for instance
exchanged between two entities, for example, between the handheld device and an
application related content server. The same is true for PS services that are used to
establish a PS bearer. Again, the PS bearer alone adds no value to the subscriber.
However, when the subscriber can use the bearer in combination with an application,
then value is added for the subscriber For example, a subscriber can use a PS
bearer between the handheld device and the Internet to gain content through the
application HTTP and TCP/IP.
Consequently, the GSM/UMTS services must be selected in such a way, that the
application running on top of it can be served in the best possible way.
TM51106EN03GLA3
2010 Nokia Siemens Networks
3.2
Service Platforms
Service platforms are entities, which offer the implementation methods for
applications. A service platform is a logical entity often containing several pieces of
equipment. Following are the majority of existing applications adopted from GSM:
1. Voice Mail System (VMS) for Voice Call Completion.
2. Service delivery platform enabling servers that support different types of
applications. A typical example is the Short Message Service Centre (SMSC) for
short message delivery.
3. Service creation and execution platform is built upon the principles of IN and is
almost obligatory to provide the envisioned services.
Following figure shows the elements of Core network service platform.
The new WCDMA radio interface will improve the quality and convenience of these
applications. It will also enable higher packet data rates, which is highly important for
the new e-mail and Internet services. The circuit connections can initially be made to
the GSM switches to provide speech and other circuit switched services of up to 64
Kbps.
IN/SCE
APSE
PSTN/ISDN
HLR
GSM
BSC
MSS
A
Gateway Control
Server (GCS)
H.248
Other
PLMN
Iu-CS
MGW
MGW
WCDMA
IP/ATM
Backbone
RNC
Iu-PS
Ext. IP
Networks
SGSN
GGSN
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2010 Nokia Siemens Networks
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11
Delay: It is the elapsed time for a packet to traverse the network from the source to
the destination. At the network layer, the end-to-end packet latency is the sum of
processing delay, transmission delay, queuing delay and propagation delay.
However, late data arrivals make data useless, resulting in receiver buffer
underflow, and early arrival can lead to receiver buffer overflow.
Loss Rate Loss rate refers to the percentage of data loss among all the delivered
data in a given transmission time interval, which can be evaluated in frame level or
packet level. Loss rate requirements apply to all classes of applications. In
general, real-time applications might tolerate a limited amount of data lost,
depending on the error resiliency of the decoder, and the type of application. On
the other hand, non-real-time applications typically have much more strict
requirement on data loss.
12
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2010 Nokia Siemens Networks
Delay
Delay Variation
Throughput
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13
14
TM51106EN03GLA3
2010 Nokia Siemens Networks
Service Classes
Traffic Parameters
QoS Parameters
TM51106EN03GLA01
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15
4.1
At least in the beginning of UMTS the most common service using conversational
QoS class will still be the speech service. QoS related to Speech service
implemented using the Core Network CS domain inherited from GSM, do not
represent an issue as such CS domain elements have already QoS "built into" the
equipment. From the UMTS point of view, the situation is a bit more complicated
because the speech coding algorithms used in GSM and UMTS are different.
UMTS, however, emphasizes the interoperability aspect between itself and GSM.
Because the interoperability is implemented with inter-system handovers it leads to
the situation where the speech coding method changes during the call. In order to
maintain the quality of the connection the system may momentarily buffer the bit
stream representing speech.
The way to solve this problem is to use an AMR (Adaptive Multi-Rate) coding
technique as defined in the 3GPP specifications.
The AMR codec is able to produce several source bit rates: 12.2, 10.2, 7.95, 7.40,
6.70, 5.90, 5.15 and 4.75 kb/s. From these rates, the 12.2 kb/s stream corresponds
to GSM-EFR (Enhanced Full Rate) equal to the GSM EFR codec.
Codec
Full Rate
Relative Encode
Relative Decode
1.0
1.0
22.0
5.4
Half Rate
20.3
8.1
AMR 12.2
21.9
6.9
AMR 10.2
20.3
7.0
AMR 7.95
21.2
6.7
AMR 7.4
19.8
6.7
AMR 6.7
20.8
6.8
AMR 5.9
17.5
9.0
AMR 5.15
15.6
8.6
AMR 4.75
18.8
6.8
Some of the other bit rates produced are also compatible with other 2G digital
systems like US- TDMA and Japanese PDC. As far as speech service is concerned,
the AMR speech codec can be considered like a "QoS converter" between different
radio access technologies.
16
TM51106EN03GLA3
2010 Nokia Siemens Networks
The next step is to convert the traditional speech service to a packet switched
service, VoIP. The 3GPP R99 specifications assume that the speech service is still
used traditionally, i.e. over circuit switched connections. The 3GPP R4/R5
specifications introduce completely different, universal CN structures using packet
switching. If the UMTS network is implemented according to 3GPP R4/R5, the
speech service is carried like packet switched from the terminal through the network
up to the Media Gateway (MGW).
The MGW contains all the required and relevant mechanisms for different
conversions; for instance, AMR is located there. Because surrounding networks are
traditional circuit switched like PSTN (Public Switched Telephony Network) and ISDN
(Integrated Services Digital Network) the MGW also contains some conversion
mechanisms for these, too. It should be noted that the conversion mechanisms are
not only limited to the information flow, signaling between different networks must be
converted, also. As far as VoIP is concerned, there are basically two alternatives for
the call control signaling, H.323 specification and SIP (Session Initiation Protocol).
TM51106EN03GLA3
2010 Nokia Siemens Networks
17
4.2
Services using conversational QoS class do not transfer any files from one end to
another literally. The QoS class allowing this kind of transfer is the streaming class.
Because complete file download takes time and thus causes delay, there must be
mechanisms to open and handle files when they are not completely transferred from
the source to the destination. This is what the streaming QoS class covers. Typical
services or applications using streaming class are those handling big files but
playing, or showing, a limited part of it.
Also the services offering a multicast type of service use streaming class if delay is
not an issue. To minimize the possible delay effects the streaming class services are
mostly unidirectional; delay exists but it does not cause any harm because the
interactivity is missing.
.Example: An application located in the UE sends the user's request concerning a
certain MP3 file to the network through which the request finally arrives at the desired
service provider. As an acknowledgement, the service provider starts downloading
the desired MP3 file. When using streaming QoS class the whole MP3 file is not
required in the UE at the same time. The point here is that the packet format data
arrives fluently enough to the UE and the packets are in the correct order.
The data stream is buffered in two places for two different purposes:
In UTRAN the data is buffered in order to control the radio interface access
and its load.
In the UE the streaming class application constructs buffers from the data
stream in order to present the data to the user "real-time-like" (continuously
without any cuts and jitter).
When the application in the UE has buffered enough data to start the playback, the
data is played to the user through the selected application. At the same time the
streaming class application continues to download more data to its buffer from the
service provider.
The data download is continued until the file is completed or the user interrupts the
process.
.
18
TM51106EN03GLA3
2010 Nokia Siemens Networks
4.3
The interactive class can afford quite remarquable and variable delays. The services
utilizing this QoS class are traditional request-response-type of services like web
surfing (request: URL address of a desired web page, response: the contents of the
web page), WAP services and other services requiring opening of server
connections. A new set of services called location based services utilize this QoS
class. The background QoS class provides the very basic connectivity level, where
delay and Bit Error Rate (BER) are, in practice, meaningless if the network is in a
position to provide error correction functionality. The services utilizing this QoS class
are typically various procedures including file downloads. For instance, e-mail
receiving is one of these.
TM51106EN03GLA3
2010 Nokia Siemens Networks
19
20
TM51106EN03GLA3
2010 Nokia Siemens Networks
TM51106EN03GLA3
2010 Nokia Siemens Networks
21
5.1
Applications are the end subscriber services. They are no longer standardized. It is
up to operators and value-added service providers to determine the need for an
application and implement them. GSM/UMTS offer the bearer and call control to
exchange content and content-related signaling information between the mobile
device and the application driven content server.
The applications that have been planned for the implementation of GSM/UMTS are
as follows:
22
TM51106EN03GLA3
2010 Nokia Siemens Networks
5.2
One of the main reasons for the implementation of UMTS networks is the anticipated
demand for data services. There are different types of PS services and requirements
for the services.
5.2.1
Voice Over IP
The well known use of voice telecommunication is telephony speech, for example,
GSM, but with Internet and multimedia, a number of new applications, for example
Voice Over IP(VoIP) and video conferencing tools, will require this scheme. Real-time
conversation is always performed between peers or groups of live or human endsubscribers. This is the only scheme where the required characteristics are strictly
given by human perception or senses.
5.2.2
PoC is a direct, real-time voice communications service. The principle of this service
is to just push to talk. The calls can be started to both individuals and groups with just
a push of a key because of a direct connection. The half-duplex or the one way at a
time, call connection is almost instant.
This technology uses the capabilities of the IP Multimedia Subsystem (IMS) as
specified by 3GPP. PoC is based on a half-duplex, always-on VoIP service over the
second generation GSM/GPRS network. Push to talk uses the Session Initiation
Protocol (SIP) service architecture as SIP messaging, which makes new applications,
such as voice chat and group chat messaging possible. Groups can also be created
using SMS, which is familiar and easy to control for the subscriber.
5.2.3
Videophone implies a full-duplex system, carrying both video and audio, and is
intended for use in a conversational environment. For this technology, the same
delay requirements as for conversational voice will apply. The added requirement is
that the audio and video must be synchronized within certain limits to provide the lipsynch, which means the synchronization of the speakers lips with the words being
heard by the end-subscriber. Due to the long delays in even the latest video codecs,
it will be difficult to meet these requirements.
The human eye is tolerant to some loss of information therefore, certain degree of
packet loss is acceptable depending on the specific video coder and amount of error
protection used. It is expected that the latest video codecs will provide acceptable
video quality with frame erasure rates up to 1%.
TM51106EN03GLA3
2010 Nokia Siemens Networks
23
5.2.4
5.2.5
When the subscriber is looking at a video or listening to audio, the scheme streams
apply. The real-time data flow is always aiming at a live or human destination. It is a
one-way transport called unidirectional continuous stream.
This scheme is new the world of data communication, which gives rise to a number of
new requirements for telecommunication and data communication systems.
Audio streaming is expected to provide better quality than conventional telephony,
and requirements for information loss in terms of packet loss will be tighter.
Similar to voice messaging, there is no conversational element involved and delay
requirements are flexible even more than for voice messaging. An example of audio
streaming is the web radio station.
The main distinguishing feature of one-way video is that there is no conversational
element involved, which means that the delay requirement is not be very stringent,
and can follow that of streaming audio. An example of one-way video is monitoring
your home using the Internet.
24
TM51106EN03GLA3
2010 Nokia Siemens Networks
5.2.6
Web Browsing
In this category, we will refer to retrieving and viewing the HTML component of a web
page. Other components, for example, images, audio, or video clips, are dealt with
under separate categories. From the subscribers perspective, the main performance
factor is how fast a page appears after it is requested. A value of 2 to 4 seconds per
page is proposed.
Mobile browsing delivers formatted Web pages to the subscribers terminal and
displays them on the screen, enabling interaction with active elements on the page,
such as links and forms. In the case of pull, the subscriber consumes the product by
clicking links and form buttons to request the next page. Mobile browsing also
supports push, which is an action initiated by the server to deliver content to the
terminal. subscribers may receive a Service Initiation push message, asking for
permission to display a page, or a Service Load push message, which depending on
the subscriber settings, can automatically load a page and then display it, or simply
have it ready in the cache for immediate display later.
5.2.7
Interactive Games
Requirements for interactive games depends on the specific game, but the
demanding applications require short delays and a value of 250 ms, which is
consistent with demanding interactive applications, is recommended.
5.2.8
E-mail is thought to be a store and forward service, which can tolerate very long
delays. However, it is important to differentiate the communications between the
subscriber and the local e-mail server and server-to-server transfer. When the
subscriber communicates with the local mail server, the expectation is that the mail
may not be transferred instantaneously but will definitely be transferred rapidly. The
proposed time is 2 to 4 seconds for the transfer, which is consistent with the research
findings on delay tolerance for web browsing.
TM51106EN03GLA3
2010 Nokia Siemens Networks
25
26
TM51106EN03GLA3
2010 Nokia Siemens Networks
Appendix
TM51106EN03GLA3
2010 Nokia Siemens Networks
27
6.1
28
TM51106EN03GLA3
2010 Nokia Siemens Networks
UMTS
TE
MT
UTRAN
CN Iu
EDGE
NODE
CN
Gateway
TE
External
Bearer
Service
Iu Bearer
Service
UTRA
FDD/TDD
Service
Physical Bearer
Service
CN Bearer
Service
Backbone
Bearer Service
TM51106EN03GLA01
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29
In another way, QoS can be seen as the degree of satisfaction of an end-user for a
delivered service.
The concept above leads to the basic idea of QoS, to distinct traffic into different
types, corresponding to their different features and different demands to the
networks, and to be delivered to the customers on different charges.
30
TM51106EN03GLA3
2010 Nokia Siemens Networks
TM51106EN03GLA01
1
Nokia Siemens Networks
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2010 Nokia Siemens Networks
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31
32
TM51106EN03GLA3
2010 Nokia Siemens Networks
Adherence to the
specified load
profile
Admission
Control
User
Service
Guarantee of
QoS Parameters
TM51106EN03GLA01
1
Nokia Siemens Networks
Actual Data
Traffic
Shaping
Shaped Data
Traffic
Policing
Yes
No
TM51107EN03GLA01
1
Nokia Siemens Networks
TM51106EN03GLA3
2010 Nokia Siemens Networks
33
34
TM51106EN03GLA3
2010 Nokia Siemens Networks
Exercises
TM51106EN03GLA3
2010 Nokia Siemens Networks
35
Exercise 1
Identify the UMTS services provided in the UMTS network? (Choose three)
CS Data Services
Web browsing
SMS
Speech call
VoIP call
VMS
Conversational class
Streaming class
Interactive class
Background class
Jitter
File size
Throughput
Delay
In the UE only
36
Traffic shaping
Traffic scheduling
Admission control
Resources reservation
TM51106EN03GLA3
2010 Nokia Siemens Networks
Solution
TM51106EN03GLA3
2010 Nokia Siemens Networks
37
Identify the UMTS services provided in the UMTS network? (Choose three)
CS Data Services
Web browsing
SMS
Speech call
VoIP call
VMS
Conversational class
Streaming class
Interactive class
Background class
Jitter
File size
Throughput
Delay
In the UE only
38
Traffic shaping
Traffic scheduling
Admission control
Resources reservation
TM51106EN03GLA3
2010 Nokia Siemens Networks