I n t e r n a t i o n a l T e l e c o m m u n i c a t i o n U n i o n

ITU-T G.1028
TELECOMMUNICATION (04/2016)
STANDARDIZATION SECTOR
OF ITU

SERIES G: TRANSMISSION SYSTEMS AND MEDIA,

DIGITAL SYSTEMS AND NETWORKS
Multimedia Quality of Service and performance – Generic
and user-related aspects

End-to-end quality of service for voice over 4G

mobile networks

Recommendation ITU-T G.1028

 ITU-T G-SERIES RECOMMENDATIONS
TRANSMISSION SYSTEMS AND MEDIA, DIGITAL SYSTEMS AND NETWORKS

INTERNATIONAL TELEPHONE CONNECTIONS AND CIRCUITS G.100–G.199

GENERAL CHARACTERISTICS COMMON TO ALL ANALOGUE CARRIER- G.200–G.299
TRANSMISSION SYSTEMS
INDIVIDUAL CHARACTERISTICS OF INTERNATIONAL CARRIER TELEPHONE G.300–G.399
SYSTEMS ON METALLIC LINES
GENERAL CHARACTERISTICS OF INTERNATIONAL CARRIER TELEPHONE SYSTEMS G.400–G.449
ON RADIO-RELAY OR SATELLITE LINKS AND INTERCONNECTION WITH METALLIC
LINES
COORDINATION OF RADIOTELEPHONY AND LINE TELEPHONY G.450–G.499
TRANSMISSION MEDIA AND OPTICAL SYSTEMS CHARACTERISTICS G.600–G.699
DIGITAL TERMINAL EQUIPMENTS G.700–G.799
DIGITAL NETWORKS G.800–G.899
DIGITAL SECTIONS AND DIGITAL LINE SYSTEM G.900–G.999
MULTIMEDIA QUALITY OF SERVICE AND PERFORMANCE – GENERIC AND G.1000–G.1999
USER-RELATED ASPECTS
TRANSMISSION MEDIA CHARACTERISTICS G.6000–G.6999
DATA OVER TRANSPORT – GENERIC ASPECTS G.7000–G.7999
PACKET OVER TRANSPORT ASPECTS G.8000–G.8999
ACCESS NETWORKS G.9000–G.9999

For further details, please refer to the list of ITU-T Recommendations.

 Recommendation ITU-T G.1028

End-to-end quality of service for voice over 4G mobile networks

Summary
Recommendation ITU-T G.1028 provides guidelines concerning the key aspects impacting end-to-end
performance of managed voice applications over LTE networks and how they can be properly assessed
using current elements of knowledge.
Some typical end-to-end scenarios are described, involving cases with LTE access at both sides of the
communication, or with a different access technology at one side (wireless or wireline access). These
scenarios are based on typical reference connections defined in this Recommendation, composed of
various segments, including: terminal, wireless access, backhaul network, core network.
Considerations regarding the sharing of the budget of some key parameters and the location where
they can be assessed across these segments are provided.

History
Edition Recommendation Approval Study Group Unique ID*
1.0 ITU-T G.1028 2016-04-06 12 11.1002/1000/12748

* To access the Recommendation, type the URL http://handle.itu.int/ in the address field of your web
browser, followed by the Recommendation's unique ID. For example, http://handle.itu.int/11.1002/1000/1
1830-en.

Rec. ITU-T G.1028 (04/2016) i

 FOREWORD
The International Telecommunication Union (ITU) is the United Nations specialized agency in the field of
telecommunications, information and communication technologies (ICTs). The ITU Telecommunication
Standardization Sector (ITU-T) is a permanent organ of ITU. ITU-T is responsible for studying technical,
operating and tariff questions and issuing Recommendations on them with a view to standardizing
telecommunications on a worldwide basis.
The World Telecommunication Standardization Assembly (WTSA), which meets every four years, establishes
the topics for study by the ITU-T study groups which, in turn, produce Recommendations on these topics.
The approval of ITU-T Recommendations is covered by the procedure laid down in WTSA Resolution 1.
In some areas of information technology which fall within ITU-T's purview, the necessary standards are
prepared on a collaborative basis with ISO and IEC.

NOTE
In this Recommendation, the expression "Administration" is used for conciseness to indicate both a
telecommunication administration and a recognized operating agency.
Compliance with this Recommendation is voluntary. However, the Recommendation may contain certain
mandatory provisions (to ensure, e.g., interoperability or applicability) and compliance with the
Recommendation is achieved when all of these mandatory provisions are met. The words "shall" or some other
obligatory language such as "must" and the negative equivalents are used to express requirements. The use of
such words does not suggest that compliance with the Recommendation is required of any party.

INTELLECTUAL PROPERTY RIGHTS

ITU draws attention to the possibility that the practice or implementation of this Recommendation may involve
the use of a claimed Intellectual Property Right. ITU takes no position concerning the evidence, validity or
applicability of claimed Intellectual Property Rights, whether asserted by ITU members or others outside of
the Recommendation development process.
As of the date of approval of this Recommendation, ITU had not received notice of intellectual property,
protected by patents, which may be required to implement this Recommendation. However, implementers are
cautioned that this may not represent the latest information and are therefore strongly urged to consult the TSB
patent database at http://www.itu.int/ITU-T/ipr/.

 ITU 2016
All rights reserved. No part of this publication may be reproduced, by any means whatsoever, without the prior
written permission of ITU.

ii Rec. ITU-T G.1028 (04/2016)

 Table of Contents
Page
1 Scope............................................................................................................................. 1
2 References..................................................................................................................... 1
3 Definitions .................................................................................................................... 3
3.1 Terms defined elsewhere ................................................................................ 3
3.2 Terms defined in this Recommendation ......................................................... 3
4 Abbreviations and acronyms ........................................................................................ 3
5 Conventions .................................................................................................................. 6
6 Brief introduction on Voice over LTE and assumptions .............................................. 6
7 Hypothetical reference models ..................................................................................... 8
7.1 LTE-LTE communication on the same network ............................................ 10
7.2 LTE-LTE communication between two interconnected networks
(includes roaming) .......................................................................................... 10
7.3 LTE-3G communication ................................................................................. 10
7.4 LTE-PSTN communication ............................................................................ 11
8 Aspects of quality of service for voice over LTE services ........................................... 11
9 Relevant indicators and quality targets ......................................................................... 13
10 Recommendations for QoS monitoring and troubleshooting ....................................... 19
10.1 Recommended QoS monitoring points .......................................................... 19
10.2 Monitoring strategy ........................................................................................ 24
10.3 Available tools in ITU-T standards ................................................................ 27
Annex A – List of degradations encountered by end-users of VoLTE service and their
potential causes ............................................................................................................. 31
A.1 QoS problem linked to call session performance ........................................... 31
A.2 QoS problem linked to perceived speech quality ........................................... 32
Bibliography............................................................................................................................. 34

Rec. ITU-T G.1028 (04/2016) iii

Recommendation ITU-T G.1028

End-to-end quality of service for voice over 4G mobile networks

1 Scope
This Recommendation works on the assumption that Voice over LTE (VoLTE) is a so-called
''managed'' voice service, in opposition to over the top (OTT) applications without use of session
initiation protocol/IP multimedia subsystem (SIP/IMS) signalling and with no prioritized traffic.
Video-telephony over LTE (ViLTE) is another service which will be addressed in a specific
Recommendation.
This Recommendation describes the key aspects impacting end-to-end performance of managed voice
applications over LTE networks, for the most common call cases (IMS tromboning is not considered,
nor single radio voice call continuity (SRVCC), nor mobility under wireless local access network),
and how they can be properly assessed using current elements of knowledge.
Relevant quality of service (QoS) mechanisms used to manage the voice service, such as robust
header compression (RoHC), transmission time interval (TTI) bundling, semi-persistent scheduling
(SPS), discontinuous transmission (DTX) and reception (DRX), service domain selection (SDS) or
SIP preconditions, are not considered in this Recommendation as a mandatory part of a VoLTE
service, but their impact on end-to-end perceived quality will be taken into account.
Analysis of the impact of VoLTE on quality of supplementary services (such as data streaming) or
on device features (battery life) is outside the scope of this Recommendation.

2 References
The following ITU-T Recommendations and other references contain provisions which, through
reference in this text, constitute provisions of this Recommendation. At the time of publication, the
editions indicated were valid. All Recommendations and other references are subject to revision;
users of this Recommendation are therefore encouraged to investigate the possibility of applying the
most recent edition of the Recommendations and other references listed below. A list of the currently
valid ITU-T Recommendations is regularly published. The reference to a document within this
Recommendation does not give it, as a stand-alone document, the status of a Recommendation.
[ITU-T E.800] Recommendation ITU-T E.800 (2008), Definitions of terms related to
quality of services.
[ITU-T E.804] Recommendation ITU-T E.804 (2014), Quality of service aspects for
popular services in mobile networks.
[ITU-T G.107] Recommendation ITU-T G.107 (2015), The E-model: a computational
model for use in transmission planning.
[ITU-T G.107.1] Recommendation ITU-T G.107.1 (2015), Wideband E-model
[ITU-T G.109] Recommendation ITU-T G.109 (1999), Definition of categories of speech
transmission quality.
[ITU-T G.114] Recommendation ITU-T G.114 (2003), One-way transmission time
[ITU-T G.711] Recommendation ITU-T G.711 (1988), Pulse code modulation (PCM) of
voice frequencies.
[ITU-T G.1000] Recommendation ITU-T G.1000 (2001), Communications quality of
service: A framework and definitions.

Rec. ITU-T G.1028 (04/2016) 1

[ITU-T P.10] Recommendation ITU-T P.10/G.100 (2006), Vocabulary for performance
and quality of service.
[ITU-T P.563] Recommendation ITU-T P.563 (2004), Single-ended method for objective
speech quality assessment in narrow-band telephony applications.
[ITU-T P.564] Recommendation ITU-T P.564 (2007), Conformance testing for voice over
IP transmission quality assessment models.
[ITU-T P.800.1] Recommendation ITU-T P.800.1 (2016), Mean opinion score (MOS)
terminology.
[ITU-T P.862] Recommendation ITU-T P.862 (2001), Perceptual evaluation of speech
quality (PESQ): An objective method for end-to-end speech quality
assessment of narrow-band telephone networks and speech codecs.
[ITU-T P.863] Recommendation ITU-T P.863 (2014), Perceptual objective listening
quality assessment.
[ITU-T P.863.1] Recommendation ITU-T P.863.1 (2014), Application guide for
Recommendation ITU-T P.863.
[ITU-T Y.1540] Recommendation ITU-T Y.1540 (2016), Internet protocol data
communication service – IP packet transfer and availability performance
parameters.
[ITU-T Y.1541] Recommendation ITU-T Y.1541 (2011), Network performance objectives
for IP-based services.
[3GPP TS 22.105] 3GPP TS 22.105 v12.1.0 (2014), 3rd Generation Partnership Project;
Technical Specification Group Services and System Aspects; Services and
service capabilities (Release 12).
[3GPP TS 23.203] 3GPP TS 23.203 v 13.5.1 (2015), 3rd Generation Partnership Project;
Technical Specification Group Services and System Aspects; Policy and
charging control architecture (Release 13).
[3GPP TS 24.229] 3GPP TS 24.229 v 13.3.1 (2015), 3rd Generation Partnership Project;
Technical Specification Group Core Network and Terminals; IP multimedia
call control protocol based on Session Initiation Protocol (SIP) and Session
Description Protocol (SDP); Stage 3 (Release 13).
[3GPP TS 26.114] 3GPP TS 26.114 v 13.1.0 (2015), 3rd Generation Partnership Project;
Technical Specification Group Services and System Aspects; IP Multimedia
Subsystem (IMS); Multimedia Telephony; Media handling and interaction
(Release 13).
[3GPP TS 26.131] 3GPP TS 26.131 v 13.1.0 (2015), 3rd Generation Partnership Project;
Technical Specification Group Services and System Aspects; Terminal
acoustic characteristics for telephony; Requirements (Release 13).
[ETSI TS 101 563] ETSI TS 101 563 v 1.4.1 (2015), Speech and multimedia Transmission
Quality (STQ); IMS/PES/VoLTE exchange performance requirements.
[ETSI TR 103 219] ETSI TR 103 219 v1.1.1 (2015), Speech and multimedia Transmission
Quality (STQ); Quality of Service aspects of voice communication in an
LTE environment.
[IETF RFC 6076] IETF RFC 6076 (2011), Basic Telephony SIP End-to-End Performance
Metrics.

2 Rec. ITU-T G.1028 (04/2016)

[IETF RFC 3095] IETF RFC 3095 (2001), Robust Header Compression (RoHC): Framework
and four profiles: RTP, UDP, ESP, and uncompressed.

3 Definitions

3.1 Terms defined elsewhere

This Recommendation uses the following terms defined elsewhere:
3.1.1 call set-up time [ITU-T E.800]: The period starting when the address information required
for setting up a call is received by the network (recognized on the calling user's access line) and
finishing when the called party busy tone, or ringing tone or answer signal is received by the calling
party (i.e., recognized on the calling user's access line). Local, national and service calls should be
included, but calls to Other Licensed Operators should not, as a given operator cannot control the
QoS delivered by another network.
3.1.2 speech quality [ITU-T P.10]: Quality of spoken language as perceived when acoustically
displayed. Result of a perception and assessment process, in which the assessing subject establishes
a relationship between the perceived characteristics, i.e., the auditory event, and the desired or
expected characteristics.
3.1.3 service accessibility performance [ITU-T E.800]: The ability of a service to be obtained,
within specified tolerances and other given conditions, when requested by the user.
3.1.4 mean one-way propagation time [ITU-T P.10]: In a connection, the mean of the
propagation times in the two directions of transmission.

3.2 Terms defined in this Recommendation

None.

4 Abbreviations and acronyms

This Recommendation uses the following abbreviations and acronyms:
2G Second Generation of radio access network (also called GSM)
3G Third Generation of radio access network (also called UMTS)
4G Fourth Generation of radio access network (also called LTE)
ACK Acknowledgment
AEC Acoustical Echo Cancellation
AF Application Function
AGC Automatic Gain Control
AGW Access Gateway
AMR Adaptive Multi Rate coding
AMR-WB Wide-Band Adaptive Multi Rate coding
BGCF Border Gateway Control Function
BTS Base Transceiver Station
CDR Call Detail Record
CNG Comfort Noise Generation
CS Circuit Switched

Rec. ITU-T G.1028 (04/2016) 3

CSFB Circuit Switched Fall Back
DL Downlink
DRX Discontinuous Reception
DTX Discontinuous Transmission
EEC Electrical Echo Cancellation
E-NodeB Enhanced Node B
EPC Evolved Packet Core
E-UTRAN Evolved UMTS Terrestrial Radio Access Network
EVS Enhanced Voice Services
GBR Guaranteed Bit Rate
GERAN GSM/Edge Radio Access Network
GGSN Gateway GPRS (General Packet Radio Service) Service Node
GSM Global System for Mobile Communications
GSMA The GSM Association
GW Gateway
HARQ Hybrid Automatic-Repeat-Request
HD voice High Definition voice
HO Hand Over
HSS Home Subscriber Server
I-CSCF Interrogating Call Session Control Function
IMS IP Multimedia Subsystem
IPDV Internet Packet Delay Variation
IPER Internet Packet Error Ratio
IPLR Internet Packet Loss Ratio
IPTD Internet Packet Transfer Delay
IRA Ineffective Registration Attempt
KPI Key Performance Indicator
LTE Long Term Evolution (of radio access networks)
MGCF Media Gateway Controller Function
MGW Media Gateway
MME Mobility Management Entity
MOS Mean Opinion Score
MOS-LQ Mean Opinion Score – Listening Quality
MTAS Multimedia Telephony Application Server
MTSI Multimedia Telephony Service for IMS
NB Narrow Band (voice spectrum)
NER Network Error Rate

4 Rec. ITU-T G.1028 (04/2016)

O-SDS Originating Service Domain Selection
OTT Over-The-Top
PCC Policy and Charging Control
PCEF Policy and Charging Enforcement Function
PCRF Policy and Charging Rule Function
P-CSCF Proxy Call Session Control Function
PDD Post Dialling Delay
PGW Packet Data Network Gateway
PLC Packet Loss Concealment
PLMN Public Land Mobile Network
PSTN Public Switched Telephone Network
QCI Quality Classification Identifier
QoS Quality of Service
RACH Random Access Channel
RoHC Robust Header Compression
RRC Radio Resource Control
RTCP Real-time Transport Control Protocol
RTP Real-time Transport Protocol
S-CSCF Serving Call Session Control Function
SCR Session Completion Ratio
SDP Session Description Protocol
SDS Service Domain Selection
SEER Session Establishment Effectiveness Ratio
SGSN Service GPRS (General Packet Radio Service) Support Node
SGW Serving Gateway
SIP Session Initiation Protocol
SIP-I Session Initiation Protocol with encapsulated ISUP
SPR Subscriber Profile Repository
SPS Semi-Persistent Scheduling
SRD Session Request Delay
SRVCC Single Radio Voice Call Continuity
SWB Super Wide Band (voice spectrum)
T-ADS Terminating Access Domain Selection
TDM Time Domain Multiplication
TrFO Transcoding Free Operation
TrGW Trunking Gateway
T-SDS Terminating Service Domain Selection

Rec. ITU-T G.1028 (04/2016) 5

TTI Transmission Time Interval
UDP User Datagram Protocol
UE User's Equipment
UL Uplink
UMTS Universal Mobile Telecommunication System
UTRAN UMTS Terrestrial Radio Access Network
VAD Voice Activity Detection
ViLTE Video-telephony over LTE
VoWiFi Voice over WiFi
VoLTE Voice over LTE
VQE Voice Quality Enhancement
WB Wide Band (voice spectrum)
WiFi Wireless Fidelity radio access network

5 Conventions
None.

6 Brief introduction on Voice over LTE and assumptions

This Recommendation makes the following assumptions, in line with the IMS profile for voice
defined by the global system for mobile communications association (GSMA) in [b-GSMA IR.92]
and the multimedia telephony service for IMS (MTSI) media handling procedures (voice part only)
defined by 3GPP in [3GPP TS 26.114], as far as Voice over LTE (VoLTE) is concerned:
– Voice calls are initiated and/or received by a terminal connected to a 4G radio access network
(evolved universal terrestrial radio access network, or E-UTRAN).
– Terminals connected to E-UTRAN embed a VoLTE client by default, either directly native
on the chip or as a dedicated application.
– E-UTRAN is connected to an evolved packet core (EPC) network. The connecting point is
an enhanced node B (e-NodeB).
– EPC enables the communication between 4G terminals and IMS core platforms, in order to
set up calls using the session initiation protocol (SIP). There is no other possibility to set up
calls.
– EPC supports quality of service (QoS) classification (between policy and charging
enforcement function (PCEF) and the terminal), as defined in [3GPP TS 22.105] (section 5)
and [3GPP TS 23.20]3 (section 6.1.7 and associated table partially reproduced in Table 1).
– SIP signalling is assigned with the QoS quality classification identifier (QCI) 5.
– Real time voice signal during calls is carried out with the real-time transport protocol/user
datagram protocol (RTP/UDP), and is assigned with the QCI 1.
– The application of QCI 5 for SIP signalling bearer and QCI 1 for real time voice bearer gives
them a high priority over all data bearer, resulting in no degradation of the voice service when
used in parallel with data session on the same device.
– Voice calls are possible seamless with any far end user of circuit switched (CS) voice
connected to public switched telephone network (PSTN) or to 2G or 3G mobile access. This

6 Rec. ITU-T G.1028 (04/2016)

 implies that the VoLTE service is assumed being deployed over the entire 4G network. Cases
of VoLTE terminals being under 4G but non-VoLTE coverage are not considered.
– Voice calls between two VoLTE clients can be placed using any bit rate and codec mode of
adaptive multi rate coding (AMR) and wide-band adaptive multi rate coding (AMR-WB), as
indicated in section 3.2.1 of [b-GSMA IR.92]. In the near future, the super wide band (SWB)
codec enhanced voice services (EVS) will also be supported. Entities ending the user plane
(terminal, media gateways (MGWs)) support transcoding free operation (TrFO).
– The feature TrFO on SIP session initiation protocol with encapsulated ISUP (SIP-I) is
activated which means that codec negotiation between VoLTE and the other networks is
performed in an end-to-end way.

Table 1 – Quality classification identifiers in use for voice over LTE

(source: [3GPP TS 23.203])
QCI Resource Priority Packet Packet Example services
type level delay error loss
budget rate
1 GBR 2 100 ms 10−2 Conversational voice
(Note)
5 Non-GBR 1 100 ms 10−6 IMS signalling
(Note)
NOTE – A delay of 20 ms for the delay between a PCEF and a radio base station should be subtracted from
a given packet delay budget to derive the packet delay budget that applies to the radio interface. This delay
is the average between the case where the PCEF is located "close" to the radio base station (roughly 10 ms)
and the case where the PCEF is located "far" from the radio base station, e.g., in case of roaming with home
routed traffic (the one-way packet delay between Europe and North America west coast is roughly 50 ms).
The average takes into account that roaming is a less typical scenario. It is expected that subtracting this
average delay of 20 ms from a given packet delay budget will lead to desired end-to-end performance in
most typical cases. Also, note that the packet delay budget defines an upper bound. Actual packet delays
– in particular for guaranteed bit rate (GBR) traffic – should typically be lower than the packet delay budget
specified for a QCI as long as the terminal has sufficient radio channel quality.
These elements are constitutive of a so-called "managed" voice service, in opposition to OTT
applications without use of SIP/IMS signalling and with no prioritized traffic.
Relevant QoS mechanisms used in 4G access network or in EPC and IMS platforms to manage the
voice service are not considered in this Recommendation as a mandatory part of a VoLTE service,
but their impact on end-to-end perceived quality will be taken into account as far as possible. The
most frequent of these features are:
– Robust header compression (RoHC, see [IETF RFC 3095]) is a method to shorten the size of
voice packets (header often represents 60% of the total of a packet, and can be reduced by a
factor of up to 20 thanks to RoHC) and thus limit the network bandwidth occupation.
– Transmission time interval (TTI) bundling is a mechanism to improve coverage at 4G cell
edge or in poor radio conditions, by transmitting each packet four times in a row
(without signalling overhead).
– Semi-persistent scheduling simplifies the allocation and re-allocation of radio resources in a
cell and thus can allow more simultaneous VoLTE calls and a lower consumption of battery
in devices.
– Discontinuous transmission (DTX) is a technique to reduce the bandwidth occupied by a
voice call, by not transmitting data during silent periods.

Rec. ITU-T G.1028 (04/2016) 7

– Discontinuous reception (DRX) puts the terminal into periodic repetition of sleep mode and
wake up mode (to listen to messages from the network), allowing saving battery
consumption.
– SIP preconditions, as specified in section 2.4.1 of [b-GSMA IR.92] and [3GPP TS 24.229],
can be added to session description protocol (SDP) messages exchanged between terminals
and IMS in order to ensure that the dedicated radio bearers are available before setting up the
call. This strengthens the call set-up process but makes it longer by up to several seconds.
– Terminating or originating service domain selection (T-SDS and O-SDS) allows selecting
between the CS or the IMS domain to provide the voice service to a VoLTE terminal under
VoLTE and CS network coverage. The mechanism for the selection of domain to
communicate call termination from the network to the terminal is called terminating access
domain selection (T-ADS).

7 Hypothetical reference models

The global architecture supporting VoLTE services can be seen in Figure 1:

Figure 1 – Overall architecture of VoLTE services

Several blocks are present and form the basic elements of a reference model:
– The terminal
– The E-UTRAN

8 Rec. ITU-T G.1028 (04/2016)

– The EPC network
– The mobile IMS core network
– Other radio access networks (GERAN and UTRAN)
– Mobile core networks for 2G and 3G
– Interconnection with other EPC or fixed networks
The description above is simplified in Figure 2:

Figure 2 – Simplified architecture of VoLTE services

This Recommendation provides an overview of the impact of various issues on perceived quality,
together with an estimation of the quantization of this impact by building block of the hypothetical
reference model, for several end-to-end scenarios falling into the scope of this Recommendation.
A client of a VoLTE service can experience different types of calls:
– Basic call: Either with another VoLTE user connected to the same 4G network
(see clause 7.1) or with a user of another voice network (CS or PSTN, see clauses 7.3 and
7.4).
– Circuit switched fall back (CSFB): A call with another 4G terminal, when one of the two
ends must perform a fall back to a CS connection over 3G or 2G before call set up. From a
user point of view, CSFB is automatic and transparent, no action is required. The
performance of CSFB in terms of call set up is seen as a sub-part of basic call performance.
Once the CSFB is performed, the performance objectives in terms of integrity and call
retainability are similar to the ones of a basic call (see clauses 7.3 and 7.4).
– Interconnection: A call between two VoLTE terminals connected to two different
interconnected networks (see clause 7.2).
– IMS tromboning: When the VoLTE terminal is under CS coverage, signalling and user planes
go through the IMS domain. From the e2e performance perspective, this IMS tromboning
should only impact end-to-end delay and post dialling delay (PDD). This call case is outside
the scope of the present Recommendation.
Due to mobility, a call initiated under a 4G VoLTE coverage may have to hand over (HO) to CS
coverage in order to continue. This process, known as SRVCC, is also outside of the current scope of
this Recommendation and under consideration for a further revision.
A call initiated under a 4G VoLTE coverage may also have to hand over to wireless local access
network coverage. A 4G terminal may also directly start a voice call on an IMS platform under this
radio coverage. This use case, known as voice over WiFi (VoWiFi), is also outside of the current
scope of this Recommendation and under consideration for a further revision.
The most common scenarios, considered in this Recommendation, are detailed below.

Rec. ITU-T G.1028 (04/2016) 9

7.1 LTE-LTE communication on the same network
As shown in Figure 3, two clients of the same VoLTE service are in communication. It is assumed
that they are not located under the coverage of the same cell.

Figure 3 – Hypothetical reference model for an LTE-LTE communication on

the same network

7.2 LTE-LTE communication between two interconnected networks (includes roaming)

Figure 4 shows clients of VoLTE services of two different operators in communication. The IMS
core networks are interconnected, allowing both signalling and user plane continuity.
It is assumed that this interconnection is IP-based (time domain multiplication (TDM) solution is not
considered). Also TrFO on SIP SIP-I feature is assumed activated on both networks.

Figure 4 – Hypothetical reference model for an LTE-LTE communication between two

interconnected networks

Two different VoLTE interconnections cases can be addressed:

– The two networks are directly interconnected which yields the use of the same codec – as for
intra public land mobile network (PLMN) case – and low values of PDD and end-to-end
delay.
– The two networks are interconnected through a CS intermediate network that increases PDD
and end-to-end delay, reduces the codec rate, and might even cause the loss of high definition
voice (HD voice) (if WB AMR codec is not supported by the CS network).

7.3 LTE-3G communication

When a VoLTE client is in communication with a CS mobile client, the mobile IMS is connected to
the 3G mobile core network (see Figure 5), allowing both signalling and user plane continuity.

10 Rec. ITU-T G.1028 (04/2016)

 Figure 5 – Hypothetical reference model for an LTE-3G communication

7.4 LTE-PSTN communication

When a VoLTE client is in communication with a user on PSTN, the mobile IMS is connected to the
fixed IMS (see Figure 6), allowing both signalling and user plane continuity.

Figure 6 – Hypothetical reference model for an LTE-PSTN communication

All scenarios including interconnection to another network may include very long international paths.
This is considered as a separate case and does not lead to considerations for the general budgeting of
delays.

8 Aspects of quality of service for voice over LTE services

Figure 7 shows the QoS degradations that can be typically encountered on a VoLTE call. QoS is
understood here as defined in [ITU-T E.800]. The main elements of the network are depicted to show
the signalling and media elements as well as the connections with PSTN or mobile platforms.

Rec. ITU-T G.1028 (04/2016) 11

 Figure 7 – Typical degradations of VoLTE communications

From a customer point of view (QoS required and perceived, as defined in [ITU-T G.1000]), these
degradations are divided into the following categories:
– Call session performance
– Problems of registration to the service (IMS/SIP).
– Call set up issues (bad accessibility).
– Failed continuity (or retainability), including impact of mobility (radio hand-overs and
SRVCC events).
– Perceived speech quality during the call (integrity)
– Frequency content. This refers to the speech spectrum of signals presented to end users
(NB, WB or SWB) and its potential distortions.
– Interruptions. Concerns all events resulting in clipping of the speech signal during the
conversation.
– End to end delay (impact on conversation interactivity)
– Presence of unwanted noises, from whatever origin.

12 Rec. ITU-T G.1028 (04/2016)

A more detailed list of typical degradations encountered by end-users of VoLTE service, together
with potential causes, is provided in Annex A.
VoLTE can also have an impact on other services or functionalities of the device. For instance:
– data transfer performance is potentially influenced by the presence of VoLTE usage on the
same cell;
– battery autonomy can become critical if the usage of a given application (here VoLTE) is
consuming too much power.
These aspects are however outside the scope of the present Recommendation.
To report, monitor and troubleshoot these QoS issues experienced by customers, consistent
measurements architecture must be set up with appropriate metrics. This is the purpose of following
sections.

9 Relevant indicators and quality targets

From an end-to-end perspective, as perceived by an end-user located under a VoLTE compliant 4G
network coverage and originating calls, the relevant indicators are given in the Table 2, together with
the potential contributor network key performance indicator(s) (KPI(s)).

Table 2 – End-to-end quality indicators and corresponding network KPIs

End-to-end indicators Definition IP network KPIs
Registration success rate Rate of successful registration Registration success rate
attempts in the VoLTE KPI related to IMS and based on
service P-CSCF counters.
Equivalent to IMS registration Equivalent to 1 – ineffective
success ratio as defined in registration attempt (IRA) ratio, as
[ETSI TR 103 219] defined in [IETF RFC 6076]
Service availability End to end service availability in Network efficiency ratio
terms of capacity to establish calls Measures the ability of network, from
from, and to, a VoLTE customer. the service platform point of view, to
Equivalent to 1 – VoLTE deliver calls to the VoLTE customer
session set-up failure ratio as Based on SIP protocol, network error
defined in [ETSI TR 103 219] rate (NER) is equivalent to session
Equivalent to 1 – telephony establishment effectiveness ratio
service non-accessibility as (SEER), as defined in
defined in [ITU-T E.804] [IETF RFC 6076]
(clause 7.3.6.1)
Post dialling delay Time interval (in seconds) SIP session set-up time
between the end of dialling by the Interval between sending INVITE
caller and the reception back by message (with SDP) and ACK (180
him of the appropriate ringing or 200) message by the originating
tone or recorded announcement. side.
Equivalent to call set-up time, as Equivalent to successful session
defined in [ITU-T E.800] request delay (SRD), as defined in
Equivalent to telephony set-up [IETF RFC 6076]
time as defined in
[ITU-T E.804]
(clause 7.3.6.2)

Rec. ITU-T G.1028 (04/2016) 13

 Table 2 – End-to-end quality indicators and corresponding network KPIs
End-to-end indicators Definition IP network KPIs
Voice quality (MOS-LQ) Equivalent to speech quality Network quality index
as defined in [ITU-T P.10]. ([ITU-T G.107], [ITU-T P.564])
Models like those defined in IP Packet loss ratio (see definition
[ITU-T P.862] and of Internet packet loss ratio (IPLR)
[ITU-T P.863] provide an in [ITU-T Y.1540]): several
objective view on the quality possible measurement points.
of the voice signal as it may
be perceived by the customer.
Can be seen on a call basis or
on a sample basis
(see [ITU-T E.804]
clauses 7.3.6.3 and 7.3.6.4)
Mouth-to-ear delay The time it takes for the IP packet transfer delay
speech signal to go from the (see definition of Internet packet
mouth of the speaker to the transfer delay (IPTD) in
ear of the listener [ITU-T Y.1540])
Round trip time
Corresponds approximately to
twice the end-to-end delay
Can be measured based on RTCP
protocol messages
Call drop rate Service continuity in terms of Session completion rate
capacity to maintain calls to their KPI related to IMS and based on
normal end. P-CSCF counters
Equivalent to telephony Equivalent to session completion
cut-off call ratio as defined in ratio (SCR), as defined in
[ITU-T E.804] [IETF RFC 6076]
(clause 7.3.6.5)
Speech bandwidth (NB, WB or Measurement of the Codec statistics
SWB) bandwidth used (normal NB Information related to the selection
or WB, or even partial and of (AMR and AMR WB) codec and
unwanted bandwidth codec modes, as well as switch
limitation). between them, accessible on SIP
protocol messages.
Measurement points in VoLTE session set-up and completion procedures have been captured in
Figures 8 and 9:

14 Rec. ITU-T G.1028 (04/2016)

 Figure 8 – VoLTE session set-up measurement flow

Figure 9 – VoLTE session completion measurement flow

For most of the indicators in Table 2, a budget can be assigned to the various segments that compose
end-to-end paths as seen in clause 6. Tables 3 to 6 provide indications of target values that can be
reasonably reached on each of these segments for each of the hypothetical reference connections
depicted in clause 7. The total budget is not necessarily the exact sum of all individual budgets.
These targets are examples of realistic values that network operators may reach when using tools
complying with up-to-date standards. For instance, the mean opinion scores (MOS) in Tables 3 to 6
are meant as average values when applying [ITU-T P.863] with the right reference sentence (i.e.,
complying with [ITU-T P.863.1]) and doing a small drive test with state-of-the art devices. For longer

Rec. ITU-T G.1028 (04/2016) 15

drive test, with e.g., hand over (HO) events, then these values may be seen as a maximum. For lab
tests with clean stable environment, then this may be seen as minimum value. It should also be noted
that this could change in the future with new devices and network set-ups.
LTE-LTE communication on the same network

Table 3 – Quality budgets for LTE-LTE communication on the same network

End-to-end TOTAL Terminal E- EPC Mobile Transmission
indicators budget UTRAN IMS network
Registration 99.9% 99.9% 99.9% 100% 99.9% No target
success rate
Service 99% No target No target No No target No target
availability Note 1 target
Post dialling LTE-LTE: No target No target No No target No target
delay (PDD) 3.5 s target
(Note 2)
CSFB: 6 s
(Note 3)
Voice quality 4 No target No target No No target No target
(MOS- (Note 4) target
LQxSW)
Mouth-to-ear 400 ms 190 ms 80 ms on 50 ms 0 10 ms (may be
delay (Note 5) (sending + both sides bigger for large
receiving) countries)
(Note 6)
Call drop rate 2% No target No target No No target No target
target
NOTE 1 – Call processing performance objective according to [ETSI TS 101 563] is higher than 99.9%.
NOTE 2 – [ETSI TS 101 563] recommends 5.9 s, with 95% of probability below 2.4 s.
NOTE 3 – Only circuit switched fall back on mobile originating side is considered here.
NOTE 4 – Assumes the use of AMR-WB at a 23.85 kbit rate. The use of other codecs and/or bit rates will
result in other values.
NOTE 5 – [ITU-T G.114] specifies a preferred maximum value at 150 ms, impossible to reach currently;
some network operators are able to provide national calls with delays below 250 ms.
NOTE 6 – According to [3GPP TS 26.131].

LTE-LTE communication between two interconnected networks (includes roaming)

Table 4 – Quality budgets for LTE-LTE communication between two interconnected

networks
End-to-end TOTAL Terminal E- EPC Mobile (International)
indicators budget UTRAN IMS transmission
network
Registration 99.9% 99.9% 99.9% 100% 99.9% No target
success rate
Service 99% No target No target No target No No target
availability target

16 Rec. ITU-T G.1028 (04/2016)

 Table 4 – Quality budgets for LTE-LTE communication between two interconnected
networks
End-to-end TOTAL Terminal E- EPC Mobile (International)
indicators budget UTRAN IMS transmission
network
Post dialling LTE-LTE: 4 s No target No target No target No No target
delay (PDD) CSFB: 6 s target
(Note 1)
Voice quality 4 (if HD voice No target No target No target No No target
(MOS- + TrFO) target
LQxSW) (Note 2)
2.8 (otherwise)
Mouth-to-ear 400 ms 190 ms 80 ms on 50 ms 0 see typical values
delay (Note 3) (sending + both sides in [b-GSMA
receiving) IR.34] Table 7
(Note 4)
Call drop rate 2% No target No target No target No No target
target
NOTE 1 – Only circuit switched fall back on mobile originating side is considered here.
NOTE 2 – Assumes the use of AMR-WB at a 23.85 kbit rate. The use of other codecs and/or bit rates will
result in other values.
NOTE 3 – [ITU-T G.114] specifies a preferred maximum value at 150 ms, impossible to reach currently;
some network operators are able to provide national calls with delays below 250 ms.
NOTE 4 – According to [3GPP TS 26.131].

LTE-3G communication

Table 5 – Quality budgets for LTE-3G communication

End-to- TOTAL Terminal E-UTRAN EPC Mobile (Inter- Far end Far end
end budget IMS national) access terminal
indicators trans- network
mission
network
Registrati 99.9% 99.9% 99.9% 100% 99.9% No target No No
on target target
success
rate
Service 98% No target No target No No No target No No
availabilit target target target target
y
Post 4.5 s No target No target No No No target No No
dialling CSFB: for target target target target
delay further
(PDD) study

Rec. ITU-T G.1028 (04/2016) 17

 Table 5 – Quality budgets for LTE-3G communication
End-to- TOTAL Terminal E-UTRAN EPC Mobile (Inter- Far end Far end
end budget IMS national) access terminal
indicators trans- network
mission
network
Voice 3.8 (if HD No target No target No No No target No No
quality voice + target target target target
(MOS- TrFO)
LQxSW) 2.8
(otherwise)
Note 1

Mouth-to- 400 ms 190 ms 80 ms 50 ms 0 see typical UTRAN 0

ear delay Note 2 (sending values in : 90
+ [b-GSMA
receiving) IR.34]
Note 3 Table 7
Call drop 3% No target No target No No No target No No
rate target target target target
NOTE 1 – Assumes the use of AMR-WB at a 12.65 kbit rate. The use of other codecs and/or bit rates will
result in other values.
NOTE 2 – [ITU-T G.114] specifies a preferred maximum value at 150 ms, impossible to reach currently;
some network operators are able to provide national calls with delays below 300 ms.
NOTE 3 – According to [3GPP TS 26.131].

LTE-PSTN communication

Table 6 – Quality budgets for LTE-PSTN communication

End-to-end TOTAL Terminal E- EPC Mobile (International) PSTN
indicators budget UTRAN IMS transmission
network
Registration 99.9% 99.9% 99.9% 100% 99.9% No target No target
success rate
Service 99% No target No target No No No target 100 %
availability target target
Post dialling 4s No target No target No No No target No target
delay (PDD) CSFB: target target
for
further
study
Voice 3.1 No target No target No No No target No
quality (Note 1) target target degradati
(MOS- on
LQxSW)
Mouth-to- 400 ms 190 ms 80 ms 50 ms 0 see typical values 10 ms
ear delay (Note 2) (sending + in [b-GSMA
receiving) IR.34] Table 7
(Note 3)

18 Rec. ITU-T G.1028 (04/2016)

 Table 6 – Quality budgets for LTE-PSTN communication
End-to-end TOTAL Terminal E- EPC Mobile (International) PSTN
indicators budget UTRAN IMS transmission
network
Call drop 2% No target No target No No No target 0%
rate target target
NOTE 1 – Assumes the use of AMR at a 12.2 kbit rate. The use of other codecs and/or bit rates will result
in other values.
NOTE 2 – [ITU-T G.114] specifies a preferred maximum value at 150 ms, impossible to reach currently;
some network operators are able to provide national calls with delays below 250 ms.
NOTE 3 – According to [3GPP TS 26.131].

10 Recommendations for QoS monitoring and troubleshooting

10.1 Recommended QoS monitoring points

The measurement points are critical to locate the issues as economical and operational matters
required to limit the number of probes, robots or other tools.
The objective is to ensure a good representation of the covered territory in statistical terms for
reporting, monitoring and troubleshooting with associated measurements sources or tools
(probes, robots, counters and call detail records – CDRs).
Three kinds of measurements points can be envisaged (see Figure 10 below):
– At end points, where end-users access and experience the service (here A and J).
– At interfaces where the user plan is accessible, in general at demarcation points between
network trunks or transmission technologies: here B, D, E and G.
– At points of presence of serving elements concerning signalling plan: here points C, F, and
H (but also E).

Rec. ITU-T G.1028 (04/2016) 19

 Figure 10 – VoLTE QoS measurement points
10.1.1 Measurement points A1, A2 and A3
Measurement at the customer premises is absolutely necessary to answer the important need of
knowledge on how the service quality is felt by end users. It gives a real end-to-end visionas there is
no other place to obtain it.
Then comes of course the question of representation: how many measurement points, what
geographical repartition? The answer depends on the possible solutions at the concerned
measurement points.
Different implementation are possible:
A1 measurement point is intended for measurements from a reporting point of view (manual test
campaigns).
In fact, this point is not mandatory for supervision. But from an operator's point of view, a lot of QoS
evaluations with this methodology are done each year on the entire network for regulatory or
benchmark purposes. So, it is a way to check the QoS and to try to resolve encountered problems.
It enables to test end-to-end metrics on accuracy mostly, but also availability, and continuity. Such
methodology is generally based on:
– A good and representative coverage of country area (rural, towns of different sizes, etc.).
– A selection of representative situation of communications (indoor, street, car, train, etc.).
– A selection of representative devices on the operator network.
– A selection of representative day periods to perform calls.
– A selection of representative calls. Narrowband and wideband calls must be taken into
account.

20 Rec. ITU-T G.1028 (04/2016)

– Enough measurements to have a correct view of QoS of the operator and relevant statistics.
A2 and A3 measurement points are intended for measurements where tools replace the end user
present in A1 and model his/her behaviour and quality judgment. They are recommended for
reporting purposes, on end-to-end metrics on accuracy mostly, but also availability, continuity and
service speed. This is where intrusive robots, drive test tools, radio tool analysers and soft agents are
required. The cost of such solution stipulates that only a sampling strategy can be envisaged. The
selection of measurement locations, devices and time periods follow similar criteria as for A1, but
with less constraints since these are automatic tools.
A2 is inside the terminal. There soft agent can access decoded signal and/or information on how the
communication is being processed (call set-up, call drop, IP jitter buffer behaviour, etc.). Such
solutions are generally very cost-effective, can collect data on a very large panel of users and devices,
but may require a participation or feedback of end-users.
A3 is where intrusive tools are required. This can be either static solutions based on robots (the focus
will then be placed on service speed, accuracy and continuity) or mobile solutions (called drive test
tools, with a larger focus enclosing also radio network performance). As seen before, for cost and
operational reasons, only a sampling strategy can be envisaged with such tools (typically less than
ten measurement points for national network). Moreover if a problem is reported by a measurement
tool, it means that many users are probably impacted equally. As a consequence of the low number
of measurements resulting from this sampling strategy, such tools will be mostly used for reporting
purposes. Drive test tools are also commonly used for troubleshooting, when the enhancement of QoS
is linked with the optimization of network coverage.
It is important to understand that this end-to-end vision enables the detection or the confirmation of
a problem but rarely to know the source of the degradation, unless it comes from the device.
10.1.2 Measurement point B
The access part of the network is an important source of outages (biggest number of network
elements). Therefore, its supervision is also necessary. One of the main difficulties is to take into
account the radio part which can be clearly different according to the selected area.
The B point is localized in eNodeB, which is the first ingress point to the LTE network. This
measurement point assumes monitoring the interfaces on both radio and in the EPC direction. The
strategy assumes differentiation of VoIP traffic from other data traffic based on adequate traffic class
indicator (QCI).
It is intended for measurements of:
– radio performance in cells: It is recommended notably for troubleshooting purposes, on
metrics on network performance. Together with A2, the eNodeB point should enable a
correct view of radio coverage performances of area;
– signalling flows, to establish statistics about calls: It is highly recommended for reporting,
monitoring and troubleshooting purposes on end-to-end metrics on service availability,
continuity, accuracy, utilization and network performance.
Supervision tools from RAN providers can enable a global aggregation of those kinds of information
from all eNodeBs.
The measurement strategy for this point should be rather cheap in implementation (i.e., performance
counters accessible from the device) than exhaustive.
10.1.3 Measurement points C1, C2 and C3
Points C1 to C3 corresponds to the interfaces of EPC (in fact all are localized close to mobility
management entity (MME)) where most important signalling messages can be captured and analysed.
They are recommended for reporting, monitoring and troubleshooting purposes on end-to-end metrics

Rec. ITU-T G.1028 (04/2016) 21

on service availability, continuity, and utilization. An exhaustive access to all data for all session is
recommended, mostly through the usage of probes or network element call detail records (CDRs).
C1 covers signalling between MME and radio access (interface S1), C2 supervises the link to serving
GW (interface S11), whereas with C3 the Sgs interface with MSC server is also addressed.
Please note that the traffic captured on all interfaces covered by C points is tunnelled (GTP v2-C).
These measurement point are important especially for the analysis of:
– EPS bearer statistics.
– Mobility management (HO statistics, routing area update).
– Subscriber management (direct connection to HSS).
10.1.4 Measurement points D1 and D2
Like C1, C2 and C3 these points are located inside the EPC, but covers interfaces where real time
flows (using the RTP protocol) can be captured and analysed.
D1 is positioned at the egress between EPC and radio access (interface S1u of serving GW), whereas
D2 covers the link between serving and PDN GWs (interfaces S5-S8).
The serving GW constitutes the anchor point for intra-LTE mobility, as well as for mobility between
GSM/GPRS, 3G/HSPA and LTE. It supports transport level QoS through marking IP packets with
appropriate DiffServ code points based on the parameters associated with the corresponding bearer.
The PDN GW is the point of interconnect to external IP networks through the SGi interface. It also
has a key role in supporting QoS for end-user IP services.
With such critical measurement points, all information on the quality of transport of communication
is accessible. They are highly recommended for reporting, monitoring and troubleshooting purposes
on end-to-end metrics on service availability, continuity, accuracy, speed and network performance.
An exhaustive access to all data for all session is recommended, mostly through the usage of probes
or network element CDRs.
10.1.5 Measurement point E
Measurement point E is located at the border between EPC and 3G core network. The corresponding
equipment is the MSC server, which controls calls and mobility in 2G and 3G networks (e.g., in case
of SRVCC). The MSC server is also in charge of TFO and TrFO functions. In many cases, MSC
server is collocated with media gateway controller function (MGCF), in charge of creating and
releasing connections at MGWs between IMS and CS domains.
This point is therefore recommended for monitoring and troubleshooting purposes on most metrics
concerning communications where roaming between LTE and 2G or 3G occurs. The concerned
metrics fall in the service availability, continuity, and speed families mostly.
10.1.6 Measurement point F
The policy and charging rule function (PCRF) is the central entity in policy control making policy
and charging control (PCC) decisions. The decisions can be based on input from a number of different
sources, including:
– Operator configuration in the PCRF that defines the policies applied to given services.
– Subscription information for a given user, received from the subscriber profile repository
(SPR).
– Information about the service received from the application function (AF).
– Information from the access network about what access technology is used, etc.

22 Rec. ITU-T G.1028 (04/2016)

The measurement point F is meant for monitoring the Rx interface between PCRF and the IMS
domain proxy call session control function (P-CSCF). It is recommended for reporting, monitoring
and troubleshooting purposes on metrics on service utilisation, availability, continuity, and speed.
10.1.7 Measurement points G1, G2 and G3
This group of measurement points corresponds to media gateways (GWs). GWs are important
locations to have specific information about evaluation of core network status (IP packets losses,
delay and jitter), codec in core, transcoding and use of TrFO. These are points of interest for
monitoring and troubleshooting purposes on metrics such as service accuracy and network
performance.
Proper monitoring of media flows (RTP) not only for interworking of services between EPC and other
networks is supported by various types of GW:
– G1 – IMS MGW – this is a GW used for anchoring media legs between two VoLTE users
but also in case of SRVCC (HO from VoLTE to mobile CS) or VoLTE – legacy mobile CS
interworking. This gateway should also provide functionality for interworking between
VoLTE and fixed VoIP users (with AMR/AMR-WB – ITU-T G.711 transcoding).
– G2 – PSTN MGW – this is a place where monitoring, reporting and troubleshooting can be
applied (for network performance), for outgoing IP (from IP to PSTN) flows. This is also
(and this is its most interest) the place where the performance of cancellation of echoes
coming from PSTN is performed.
– G3 – Trunking gateway (TrGW) in case of interconnection with another IP domain operator.
10.1.8 Measurement Points H1, H2, H3 and H4
This is the group of measurement points inside IMS platforms. Monitoring information which can be
possessed here are related to the signalling traffic. Probes which can be deployed at the platform side
should be complementary to the information received from CDRs and counters. Monitoring
information which should be gathered at the corresponding measurement points are related to service
availability, continuity and utilization. According to the service (voicemail, conferencing, etc.),
accuracy and network performance supervisions can also be interesting to track audio problems.
Instead of having one single point, it is recommended to separate the capture of metrics at the level
of different entities:
– H1 – is located at the interface between EPC (PDN GW) and IMS (P-CSCF). This point
covers passive monitoring of Gm interface, which is the first interface where none tunnelled
SIP signalization coming from all users in the network is centralized;
– H2 – is a global monitoring point localized on IMS platform (comprising at least elements
such as I- and S-CSCF, AS and SCC-AS), ideal to make reporting on service utilization as
the analysis will be highly statistically reliable. Monitoring data gathered in this point are
focused on performance counters and billing data. As the most suitable platform element for
gathering statistics may be metric dependent, detailed information on collection interface will
be specified in metrics definitions;
– H3 – measurement point (on call server/MGCF) can be used for an exhaustive view on the
signalling protocol concerning all calls outgoing to PSTN network, since the call server is
involved in all call negotiations. Service availability metrics (and other with less priority
concerning continuity and network performance) are measurable at this point;
– H4 – controls IBCF, where the interconnection between two operator domains is performed
(including charging data records). SIP signalling information for interconnection can be
monitored here.

Rec. ITU-T G.1028 (04/2016) 23

10.1.9 Measurement point J
The J measurement point is the circuit-switched network counterpart of A point. The same metrics
can be measured for similar purposes (except radio network performance), and with the same
sampling strategy.
"Circuit switch", in this case means all possible interfaces: PSTN, 2G/3G PLMN as well as fixed
VoIP access behind an FXS port of a DSL router.

10.2 Monitoring strategy

This clause describes the location of the metrics that have to be measured according to their usage.
In particular, it aims at defining:
– what needs to be measured, how it will be measured (with what source of information, at
what frequency) and why these measurements are needed;
– what kind of measurement tools and where (locations description and number) these
measurements are needed.
10.2.1 Reporting
The reporting will provide a general view on the service QoS, potentially in a benchmark perspective
(between countries, or against domestic competitors), and on its evolution in time (longitudinal view).
It can also allow identification of categories of customers that have a degraded QoS.

24 Rec. ITU-T G.1028 (04/2016)

 Figure 11 – VoLTE QoS measurement points – reporting

All measurement points can actually provide data for reporting. Even not fully representative data,
like those gathered from intrusive measurements, can be valuable for this purpose.
The following dashboards can be built based on such measurements:
– General view of the service utilisation (number of customers, number of calls, call durations,
churn rate).
– Performance of service platforms and network equipment (service availability and
continuity).
– QoS counters (availability, PDD, mean opinion score (MOS), call continuity).

Rec. ITU-T G.1028 (04/2016) 25

10.2.2 Monitoring
All measurement points can provide data for monitoring, as long as they can provide a view on either
end-to-end QoS or the contribution of a network section to this QoS.
The hot mode monitoring (alarm raising) will provide real time information related to degradation of
the service QoS impacting a great number of customers.

Figure 12 – VoLTE QoS measurement points – monitoring

10.2.3 Troubleshooting
The troubleshooting will provide additional information by:
– In depth analysis of customer or access below the threshold, by trying to reproduce the causes
for the encountered problems and analysing data collected during the problem by different
sources (CDRs, probes).
– Remote testing/configuration checks, including trace capture and analysis at major network
nodes.

26 Rec. ITU-T G.1028 (04/2016)

– On site investigations, including trace capture at the customer premise or in the access
network.

Figure 13 – VoLTE QoS measurement points – troubleshooting

10.3 Available tools in ITU-T standards

Though based on new network technologies, VoLTE remains a telephony offer subject to all
requirements expressed by end users concerning its usage, and in particular when it comes to QoS.
Therefore, provisions already contained in existing ITU-T Recommendations concerning the
procurement and evaluation of perceived quality of voice services apply in most cases to VoLTE as
well.
10.3.1 Definition of key performance indicators
As can be seen in previous sections (in particular in Table 2), the metrics representative of the quality
of service as perceived by end users are not specific to VoLTE. Most of them are already defined in

Rec. ITU-T G.1028 (04/2016) 27

[ITU-T P.10], and these definitions apply for this context. Following are the details of these
references:
– Service availability: see service accessibility performance (clause 3.1.1.2.2 in
[ITU-T E.800]).
– Post dialling delay: see call set-up time (clause 3.1.1.2.1 in [ITU-T E.800]).
– Voice quality (MOS-LQ, see ''Speech Quality'' (S-28) in [ITU-T P.10] or the various
definitions provided in [ITU-T P.800.1].
– Mouth-to-ear delay: not defined as such in ITU-T Recommendations. See however mean
one-way propagation time (M-1) in [ITU-T P.10], as well as explanation in [ITU-T G.114].
– Call drop ratio: see telephony cut off call ratio in [ITU-T E.804].
– Speech bandwidth (NB, WB or SWB): amendment 4 of [ITU-T P.10] gives very detailed
definition of various audio bandwidth used in telephony.
Registration success rate for VoLTE is not defined in Recommendations. [ITU-T E.804]
(clause 7.2.2.1) defines a network selection and registration failure ratio, but this disregards the
specificity of IMS registration.
As far as the main QoS dimensions are concerned, the four viewpoints of QoS (see Figure 14) defined
in [ITU-T G.1000] are also useful in the context of VoLTE.

Figure 14 – The four viewpoints of QoS (source: [ITU-T G.1000])

As far as it concerns metrics representative of the underlying network layers, [ITU-T Y.1540]
provides information on IP-related metrics, while no ITU-T standards address radio metrics.
10.3.2 Tools and models for the measurement and prediction of voice quality
There are two approaches for the assessment of end-to-end voice quality:
– Parametric tools take advantage of the good correlation between technical information of a
connection and the corresponding end-to-end quality as perceived by end-users, to produce
a relatively accurate estimate at a cheap implementation cost. Such a tool can be envisaged
at edge points, close to the end-user, for a better prediction of individual quality, or inside the
network, for a good knowledge of the general impact of network performance on end-to-end
quality. [ITU-T P.564] describes a general class of parametric voice quality prediction
models that provide highly scalable voice quality estimation using information in the
IP/UDP/RTP header of packets. In addition, [ITU-T P.564] provides performance criteria for
models of this type that operate on narrowband speech.

28 Rec. ITU-T G.1028 (04/2016)

– Another type of parametric tool is the E-model described in [ITU-T G.107], which is a widely
used transmission planning tool. Most factors present in the model (now adapted for IP
transmission and WB telephony, see [ITU-T G.107.1]) apply for VoLTE.
– Signal-based models are much more complicated than parametric tools since they require a
capture and an analysis of the speech signal. This is why they are mostly used in a point-to-
point perspective, in order to measure very accurately the end-to-end quality of a voice
service at a given time and at a given location (and most of the time for a given service with
a given end device). ITU-T developed several such tools: [ITU-T P.862] and [ITU-T P.863]
for full-reference models, and [ITU-T P.563] for a single-ended implementation (restricted
to narrowband telephony).
10.3.3 Applicable acceptability thresholds and targets
In general, ITU-T do not specify acceptability targets for end-user's metrics. However, there is a
notorious exception for end-to-end delay, where [ITU-T G.114] specifies (in clause 4) a first threshold
at 150 ms below which most users do not notice the delay, and a second one at 400 ms above which
the quality is considered as unacceptable.
R factor values computed with the E-model from [ITU-T G.107] (and translated into MOS-CQE
scores, as shown on Figure 15) can also be compared with acceptability thresholds. Recommendation
[ITU-T G.109] defines such categories (for NB telephony only) reproduced in Table 7:

Table 7 – Definition of categories of speech transmission quality (source: [ITU-T G.109])

Speech transmission
R-value range User satisfaction
quality category
90  R < 100 Best Very satisfied
80  R < 90 High Satisfied
70  R < 80 Medium Some users dissatisfied
60  R < 70 Low Many users dissatisfied
50  R < 60 Poor Nearly all users dissatisfied

Figure 15 – MOS-CQE as function of rating factor (source: [ITU-T G.107])

Recommendation [ITU-T Y.1541] also provides performance objectives for various QoS classes (see
Table 8) based on IP-network metrics defined and specified in [ITU-T Y.1540]. VoLTE falls into
class 0 or 1:

Rec. ITU-T G.1028 (04/2016) 29

 Table 8 – IP network QoS class definitions and network performance
objectives (source: [ITU-T Y.1541])

Network performance QoS classes

Nature of network performance objective
parameter Class 0 Class 1
IPTD Upper bound on the mean IPTD 100 ms 400 ms
–3
IPDV Upper bound on the 1  10 quantile of IPTD 50 ms 50 ms
minus the minimum IPTD
IPLR Upper bound on the packet loss probability 1 × 10–3 1 × 10–3
IPER Upper bound 1 × 10–4

30 Rec. ITU-T G.1028 (04/2016)

 Annex A

List of degradations encountered by end-users of VoLTE service and their

potential causes
(This annex forms an integral part of this Recommendation.)

A.1 QoS problem linked to call session performance

Table A.1 – degradations related to call session performance and their potential causes
Kind of degradation Possible reasons: Location
Identification Failure – Problem with MME, HSS or PCRF EPC
– Error in scheduling
– Radio resource control (RRC) connection set-up failure
(reception of RRC connection reject, or expiry of timer eUTRAN
T300, no RRC connection set-up complete sent after
reception of RRC connection set-up).
– Not available due to load (serving gateway (SGW) or packet
Unavailability of data network gateway (PGW))
basic call – Failed negotiation (no allocation of QCI, no codec match,
SIP preconditions unmet, etc.)
– Reception of several SIP error codes (e.g., 401 = EPC
Unauthorized, 405 = Method Not Allowed, etc.)
– Reception of SIP CANCEL from IMS
– TD internal timer expired, causing a
''SessionSetupFailureTimeout''
– Load
High post dialling – Interworking between systems
All
delay – Use of SIP preconditions
– CS fall back or IMS tromboning at call set-up
– Bad negotiation between two equipments of the network
Link failure eUTRAN/ EPC
during call establishment (bad codec management)
– Terminal is not able to code or decode speech while the
White call Terminal
signalling is OK for the communication
– Terminal bug, bad covered area, handover/SRVCC failures
due to cells neighbourhood problem, etc.
– RRC connection drop (at reception of RRC connection re- Terminal/
establishment reject, or expiry of timer T301 or in case RRC eUTRAN
connection release is received before new RRC connection
set-up attempt)
Call drop – Link failure: System failure, bad re-negotiation between two
equipments of the network during call.
– Reception of SIP status code 500 (Server Internal Error)
– No RTP packet received during a period longer than EPC
''SessionDropTimeout'' TD internal timer
– No SIP 200 OK on BYE is received within the time measured
by ''SessionHangupTimeout'' TD internal timer

Rec. ITU-T G.1028 (04/2016) 31

A.2 QoS problem linked to perceived speech quality

Table A.2 – Degradations related to perceived speech quality and their potential causes
Kind of Possible reasons: Location
degradation
– Disturbing comfort noise generation (CNG) due to bad noise
reduction.
– Noise due to bad electronic implementation on terminal (e.g.,
Noise analogue /digital conversion).
– Disturbing residual noise due to bad noise reduction.
Terminal
– Background noise (street, car, etc.).
– Additional noise due to eUTRAN configuration problem.
– Bad performance of acoustic echo cancellation (AEC)/ No AEC. As
reminder: Acoustic echo is the coupling between the loudspeaker
and the microphone of the phone terminal.
Echo – Bad performance of electric echo cancellation (EEC)/No EEC .
Reminder: Electrical echo is due to digital to analogue
Networks
transformation for a call between mobile terminal and PSTN (No
electrical echo for mobile to mobile call).
Low/high speech
– Bad performance of automatic gain control (AGC)/No AGC. Terminal
level
– Narrowband instead of wideband speech quality:
• Remote terminal not WB
• HO towards 2G Terminal/
• Call with PSTN, 2G, mobile platforms, etc. where wideband eUTRAN
is not deployed.
Encoding /
• Interworking with CS 3G not WB
decoding issues
– Lower WB-AMR bitrate/AMR (Loaded cell, autonomous mode,
etc.) leading to distortion on speech signal.
Terminal/
– Many transcodings (for example with call to voicemail) leading to
eUTRAN
distortion on speech signal
– Rebuffering and time scaling causing distortion
– Although WB-AMR codec is supported, the acoustical performance
of the terminal (on receiving and/or sending side) is not wideband
compliant.
Terminal Acoustic Terminal
– Not well-balanced acoustic terminal can lead to a sound which seems
too aggressive, too muffled, etc.
– Distortion due to transducers.
– Bad VAD/DTX/DRX implementation.
Terminal
– Problem with voice quality enhancement (VQE) algorithm.
Chopped – IP packet loss or jitter in network (congestion, QoS prioritization,
Conversation UL/DL scheduling delays, radio retransmissions, handover).
All
– Bad handling of IP packet loss and inter-arrival jitter by jitter buffers
or packet loss concealment (PLC) inside terminals
DTMF not
– Problem with in-band or out-band processing All
recognized

32 Rec. ITU-T G.1028 (04/2016)

 Table A.2 – Degradations related to perceived speech quality and their potential causes
Kind of Possible reasons: Location
degradation
– Network load.– Media handling (packet construction, jitter buffer
management)
– Speech processing in terminals
– Random access channel (RACH) upon receiving handover
command
E2E delay All
– RACH/contention procedure
– Additional RACH attempts
– Dynamic scheduling, link adaptation
– Radio link failure/re-establishment during handover
(possibly different cell)
– Network congestion (several causes : traffic load, distance from cell
RTP/IP packet centre causing activation of TTI bundling, for instance) EPC/
loss – Jitter buffers not adapted to actual jitter amount or packet size Terminal
(can depend on use of RoHC or not)
RTP/IP
– New route after a problem such as congestion EPC
Desequencing
Network Delay – Network congestion EPC/
Variation (Jitter) – Jitter buffers not adapted Terminal
– Limit of the cell coverage
– Interference
– Area not well covered (obstacle, etc.)
– Bad radio optimization
Radio
degradations – Radio loss profile eUTRAN
– Bad radio scheduling
– No or bad use of hybrid automatic-repeat-request (HARQ)
mechanisms
– Etc.
EPC/CS
Handover latency – Latency due to new route after HO or SRVCC
network

Rec. ITU-T G.1028 (04/2016) 33

 Bibliography

[b-GSMA IR.34] GSMA IR.34 v 9.1 (2013), Guidelines for IPX Provider networks.
[b-GSMA IR.92] GSMA IR.92 v 7.0 (2013), IMS Profile for Voice and SMS.

34 Rec. ITU-T G.1028 (04/2016)

 SERIES OF ITU-T RECOMMENDATIONS

Series A Organization of the work of ITU-T

Series D General tariff principles

Series E Overall network operation, telephone service, service operation and human factors
Series F Non-telephone telecommunication services

Series G Transmission systems and media, digital systems and networks

Series H Audiovisual and multimedia systems

Series I Integrated services digital network

Series J Cable networks and transmission of television, sound programme and other multimedia
signals
Series K Protection against interference
Series L Environment and ICTs, climate change, e-waste, energy efficiency; construction, installation
and protection of cables and other elements of outside plant
Series M Telecommunication management, including TMN and network maintenance
Series N Maintenance: international sound programme and television transmission circuits
Series O Specifications of measuring equipment

Series P Terminals and subjective and objective assessment methods

Series Q Switching and signalling

Series R Telegraph transmission

Series S Telegraph services terminal equipment

Series T Terminals for telematic services

Series U Telegraph switching

Series V Data communication over the telephone network

Series X Data networks, open system communications and security

Series Y Global information infrastructure, Internet protocol aspects and next-generation networks,
Internet of Things and smart cities

Series Z Languages and general software aspects for telecommunication systems

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Geneva, 2016