INTERNATIONAL TELECOMMUNICATION UNION

ITU-T G.769/Y.1242
TELECOMMUNICATION (06/2004)
STANDARDIZATION SECTOR
OF ITU

SERIES G: TRANSMISSION SYSTEMS AND MEDIA,

DIGITAL SYSTEMS AND NETWORKS
Digital terminal equipments – Principal characteristics of
transcoder and digital multiplication equipment
SERIES Y: GLOBAL INFORMATION
INFRASTRUCTURE, INTERNET PROTOCOL ASPECTS
AND NEXT GENERATION NETWORKS
Internet protocol aspects – Architecture, access, network
capabilities and resource management

Circuit multiplication equipment optimized for

IP-based networks

ITU-T Recommendation G.769/Y.1242

 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 G.400–G.449
SYSTEMS ON RADIO-RELAY OR SATELLITE LINKS AND INTERCONNECTION WITH
METALLIC LINES
COORDINATION OF RADIOTELEPHONY AND LINE TELEPHONY G.450–G.499
TESTING EQUIPMENTS G.500–G.599
TRANSMISSION MEDIA CHARACTERISTICS G.600–G.699
DIGITAL TERMINAL EQUIPMENTS G.700–G.799
General G.700–G.709
Coding of analogue signals by pulse code modulation G.710–G.719
Coding of analogue signals by methods other than PCM G.720–G.729
Principal characteristics of primary multiplex equipment G.730–G.739
Principal characteristics of second order multiplex equipment G.740–G.749
Principal characteristics of higher order multiplex equipment G.750–G.759
Principal characteristics of transcoder and digital multiplication equipment G.760–G.769
Operations, administration and maintenance features of transmission equipment G.770–G.779
Principal characteristics of multiplexing equipment for the synchronous digital hierarchy G.780–G.789
Other terminal equipment G.790–G.799
DIGITAL NETWORKS G.800–G.899
DIGITAL SECTIONS AND DIGITAL LINE SYSTEM G.900–G.999
QUALITY OF SERVICE AND PERFORMANCE - GENERIC AND USER-RELATED G.1000–G.1999
ASPECTS
TRANSMISSION MEDIA CHARACTERISTICS G.6000–G.6999
DIGITAL TERMINAL EQUIPMENTS G.7000–G.7999
DIGITAL NETWORKS G.8000–G.8999

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

ITU-T Recommendation G.769/Y.1242

Circuit multiplication equipment optimized for IP-based networks

Summary
This Recommendation contains principles and examples of multiplication schemes of voice,
facsimile and voiceband data between the International Switching Centre (ISC) (exchanges) which
are connected via IP-based networks.

Source
ITU-T Recommendation G.769/Y.1242 was approved on 13 June 2004 by ITU-T Study Group 15
(2001-2004) under the ITU-T Recommendation A.8 procedure.

ITU-T Rec. G.769/Y.1242 (06/2004) i

 FOREWORD
The International Telecommunication Union (ITU) is the United Nations specialized agency in the field of
telecommunications. 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 received notice of intellectual property,
protected by patents, which may be required to implement this Recommendation. However, implementors
are cautioned that this may not represent the latest information and are therefore strongly urged to consult the
TSB patent database.

 ITU 2004
All rights reserved. No part of this publication may be reproduced, by any means whatsoever, without the
prior written permission of ITU.

ii ITU-T Rec. G.769/Y.1242 (06/2004)

 CONTENTS
Page
1 Scope ............................................................................................................................ 1
2 Normative references.................................................................................................... 1
3 Definitions .................................................................................................................... 3
4 Abbreviations................................................................................................................ 4
5 Target services to be supported .................................................................................... 5
6 Network reference model ............................................................................................. 5
6.1 Connection configuration ............................................................................... 5
6.2 Interfaces ........................................................................................................ 6
7 Functions of the IP-CME.............................................................................................. 7
7.1 Packetized transmission modes and their functions related to stream
handling function............................................................................................ 7
7.2 Trunk I/F access function ............................................................................... 9
7.3 ISC signalling I/F control function................................................................. 9
7.4 IP transmission channel I/F access function................................................... 10
7.5 Packetized IP-CME control signalling function............................................. 11
7.6 Transport of call control signalling function .................................................. 12
7.7 Multiplexing function..................................................................................... 12
7.8 Application transmission function.................................................................. 15
7.9 Multiplexing load control function................................................................. 15
7.10 System operation management function ........................................................ 16
7.11 QoS policy management function .................................................................. 16
7.12 Network management function ...................................................................... 16
8 Structure of the multiplexed packet.............................................................................. 16
8.1 PL indicator bit (X) ........................................................................................ 17
8.2 Payload Length (PL)....................................................................................... 17
8.3 IPP-ID indicator bit (Y).................................................................................. 17
8.4 IP port ID (IPP-ID)......................................................................................... 17
Annex A – IP transmission channel control procedure for packetized transmission
mode A ......................................................................................................................... 18
A.1 Conditions....................................................................................................... 18
A.2 Parameters ...................................................................................................... 18
A.3 Procedure........................................................................................................ 20
Annex B – IP transmission channel control procedure for packetized transmission
mode B.......................................................................................................................... 23
B.1 Introduction .................................................................................................... 23
B.2 Approach ........................................................................................................ 23
B.3 Multiplexing structure .................................................................................... 24
B.4 Short packet format for typical applications (details are for further study) ... 25

ITU-T Rec. G.769/Y.1242 (06/2004) iii

 Page
Appendix I – Functional architecture....................................................................................... 27
I.1 Functional implementation............................................................................. 27

iv ITU-T Rec. G.769/Y.1242 (06/2004)

ITU-T Recommendation G.769/Y.1242

Circuit multiplication equipment optimized for IP-based networks

1 Scope
This Recommendation contains principles and examples of multiplication schemes of voice,
facsimile and voiceband data between the International Switching Centre (ISC) (exchanges) (see
Note) which are connected via IP-based networks.
Circuit multiplication equipment may have integral echo control and A/µ-law converter functions.
The information in this Recommendation is compatible with the control procedures for such
devices.
NOTE – As circuit multiplication equipment may also be used in national networks, the signalling described
here could not only be used in International switching centres but also in national exchanges.
This Recommendation applies to digital circuit multiplication equipment optimized for IP-based
networks (IP-CME) and specifies the following aspects for IP-CME in order to achieve
interworking between them.
a) Network interface requirements
– connection configuration;
– trunk and bearer facility interface;
– IP-based networks interface;
– call control signalling;
– IP-CME control signalling which includes definition of coding types;
– echo control.
b) Functional requirements
– multiplication schemes optimized for IP-based networks;
– handling of the call signalling transmission between the ISCs;
– handling of the IP-CME control signalling between IP-CMEs;
– multiplexing load control of IP-transmission channels over IP-based networks;
– dynamic load control of calls in PSTN side;
– network management;
– management of voice, facsimile and voiceband data quality transported over IP-based
networks;
– system operation (capacity, overload strategy, maintenance, alarm).
c) Performance criteria of IP-CME system elements
– speech detector;
– facsimile detector;
– voiceband data detector;
– signalling detector.

2 Normative 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;

ITU-T Rec. G.769/Y.1242 (06/2004) 1

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 Recommendation G.109 (1999), Definition of categories of speech transmission
quality.
– ITU-T Recommendation G.168 (2004), Digital network echo cancellers.
– ITU-T Recommendation G.177 (1999), Transmission planning for voiceband services over
hybrid Internet/PSTN connections.
– ITU-T Recommendation G.701 (1993), Vocabulary of digital transmission and
multiplexing and pulse code modulation (PCM) terms.
– ITU-T Recommendation G.703 (2001), Physical/electrical characteristics of hierarchical
digital interfaces.
– ITU-T Recommendation G.704 (1998), Synchronous frame structures used at 1544, 6312,
2048, 8448 and 44 736 kbit/s hierarchical levels.
– ITU-T Recommendation G.711 (1988), Pulse code modulation (PCM) of voice frequencies.
– ITU-T Recommendation G.711 Appendix I (1999), A high quality low-complexity
algorithm for packet loss concealment with G.711.
– ITU-T Recommendation G.711 Appendix II (2000), A comfort noise payload definition for
ITU-T G.711 use in packet-based multimedia communication systems.
– ITU-T Recommendation G.723.1 (1996), Speech coders: Dual rate speech coder for
multimedia communications transmitting at 5.3 and 6.3 kbit/s.
– ITU-T Recommendation G.723.1 Annex A (1996), Silence compression scheme.
– ITU-T Recommendation G.726 (1990), 40, 32, 24, 16 kbit/s Adaptive Differential Pulse
Code Modulation (ADPCM).
– ITU-T Recommendation G.729 (1996), Coding of speech at 8 kbit/s using conjugate-
structure algebraic-code-excited linear prediction (CS-ACELP).
– ITU-T Recommendation G.729 Annex B (1996), A silence compression scheme for G.729
optimized for terminals conforming to Recommendation V.70.
– ITU-T Recommendation G.763 (1998), Digital circuit multiplication equipment using
G.726 ADPCM and digital speech interpolation.
– ITU-T Recommendation G.957 (1999), Optical interfaces for equipments and systems
relating to the synchronous digital hierarchy.
– ITU-T Recommendations I.233.x (1991), Frame mode bearer services.
– ITU-T Recommendation I.363.1 (1996), B-ISDN ATM Adaptation Layer specification:
Type 1 AAL.
– ITU-T Recommendation I.363.2 (2000), B-ISDN ATM Adaptation Layer specification:
Type 2 AAL.
– ITU-T Recommendation I.363.5 (1996), B-ISDN ATM Adaptation Layer specification:
Type 5 AAL.
– ITU-T Recommendation 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.

2 ITU-T Rec. G.769/Y.1242 (06/2004)

– ITU-T Recommendation Q.2 (1988), Signal receivers for automatic and semi-automatic
working, used for manual working.
– ITU-T Recommendation Q.50 (2001), Signalling between Circuit Multiplication Equipment
(CME) and International Switching Centres (ISC).
– ITU-T Recommendation Q.50.1 (2001), Signalling between International Switching
Centres (ISC) and Digital Circuit Multiplication Equipment (DCME) including the control
of compression/decompression.
– ITU-T Recommendation Q.52 (2001), Signalling between international switching centres
and stand-alone echo control devices.
– ITU-T Recommendation Q.400 (1988), Forward line signals.
– ITU-T Recommendation Q.931 (1998), ISDN user-network interface layer 3 specification
for basic call control.
– ITU-T Recommendation T.30 (2003), Procedures for document facsimile transmission in
the general switched telephone network.
– IETF RFC 1661 (1994), The Point-to-Point Protocol (PPP).
– IETF RFC 1812 (1995), Requirements for IP Version 4 Routers.
– IETF RFC 2131 (1997), Dynamic Host Configuration Protocol.
– IETF RFC 2427 (1998), Multiprotocol Interconnect over Frame Relay.
– IETF RFC 2460 (1998), Internet Protocol, Version 6 (IPv6) Specification.
– IETF RFC 2719 (1999), Framework Architecture for Signalling Transport.
– IETF RFC 2833 (2000), RTP Payload for DTMF Digits, Telephony Tones and Telephony
Signals.
– IETF RFC 2960 (2000), Stream Control Transmission Protocol.
– IETF RFC 3550 (2003), RTP: A Transport Protocol for Real-Time Applications.
– IEEE 802 (2001), IEEE Standards for Local and Metropolitan Area Networks: Overview
and Architecture.

3 Definitions
Definitions relating to the IP-CME are as follows:
3.1 IP-based CME (IP-CME): IP-CME constitutes a general class of equipment that permits
concentration of a number of IP ports on a reduced number of transmission channels over IP-based
networks.
3.2 low rate encoding (LRE): The speech-coding methods with bit rates less than 64 kbit/s,
e.g., the 32 kbit/s transcoding process defined in ITU-T Rec. G.726 applied to speech coded
according to ITU-T Rec. G.711.
Furthermore, in VoIP systems, coding-decoding device ("codec") that generate encoded blocks of
voice signals in each periodical frame are usually adopted. For example, codecs such as the G.729
Annexes and G.723.1 are common in the VoIP field, and the basic intervals of their frames are
usually multiples of 10 ms.
3.3 speech activity ratio: The ratio of the time speech and corresponding hangover occupies
the trunk to the total measuring time, averaged over the total number of trunks carrying speech.

ITU-T Rec. G.769/Y.1242 (06/2004) 3

3.4 trunk: A bidirectional connection consisting of a forward channel and a backward channel
between the SW (the International Switching Centre). Each channel in the trunk interface is
identified by Trunk channel ID.
3.5 IP port: A bidirectional call stream between the IP-CMEs. An IP port in an IP transmission
channel is distinguished by the IP port ID (IPP-ID) based on UDP port number and mapped to
correspondent trunk.
3.6 IP transmission channel: A bidirectional multiplexed IP/UDP/RTP stream channel
between the IP-CME that transmits speech data and VBD over IP-based networks.
3.7 freeze-out: The temporary condition when a trunk channel becomes active and cannot
immediately be assigned to an IP transmission channel, due to lack of available transmission
capacity and so on.
3.8 freeze-out fraction: The ratio of the sum of the individual channel freeze-outs to the sum
of the active signals and their corresponding hangover times and front end delays, for all trunk
channels over a fixed interval of time, e.g., one minute.

4 Abbreviations
This Recommendation uses the following abbreviations.
CME Circuit Multiplication Equipment
CRTP Compressed RTP (Real-time Transport Protocol)
DHCP Dynamic Host Configuration Protocol
DTMF Dual Tone Multi-Frequency
ECRTP Enhanced CRTP
GSTN General Switched Telephone Network
IETF Internet Engineering Task Force
IFP Internet Facsimile Protocol
IP Internet Protocol
IP-CME Circuit Multiplication Equipment optimized for IP-based networks
IPP-ID IP port ID
ISC International Switching Centre
ITU International Telecommunication Union
MoIP Modem over IP
MUX RTP Multiplexing RTP
PCM Pulse Code Modulation
PPP Point-to-Point Protocol
PSTN Public Switched Telephone Network
ROHC RObust Header Compression
RTCP RTP Control Protocol
RTP Real-time Transport Protocol
SIGTRAN SIGnalling TRANsport
SNMP Simple Network Management Protocol

4 ITU-T Rec. G.769/Y.1242 (06/2004)

SPH Short Packet Header
SPRT Simple Packet Relay Transport protocol
SS7 Signalling System No. 7
SW Switch
TCP Transmission Control Protocol
TDM Time Division Multiplexing
TFO Tandem Free Operation
UDP User Datagram Protocol
UDPTL Facsimile UDP Transport Layer protocol
VAD Voice Activity Detection
VBD VoiceBand Data
VoIP Voice over IP

5 Target services to be supported

All kinds of telephony services, such as speech, facsimile (includes ITU-T Rec. T.30) and VBD
shall be supported by IP-CME. Facsimile demodulation/remodulation is optionally supported.
NOTE – How to support Modem signal and Tone signal is for further study at this moment.

6 Network reference model

6.1 Connection configuration

Multiplexing/demultiplexing the circuits communicating between an originator and a terminator are
on the same or different General Switched Telephone Network (GSTN) by the IP-CMEs located at
both ends of the transit IP-based networks. The single connection configuration is shown in
Figure 1. The multipoint connection configuration is also shown in Figure 2. And the multipoint
connection configuration is a set of single one.

G.769_F01

Figure 1/G.769/Y.1242 – Single connection configuration of the IP-CME

ITU-T Rec. G.769/Y.1242 (06/2004) 5

 . .
. .
. .
. .
. .
. .
. .
. .
. .

G.769_F02

Figure 2/G.769/Y.1242 – Multipoint connection configuration of the IP-CME

6.2 Interfaces
IP-CMEs are connected with the GSTN switches (International Switching Centres (ISCs)). The
following two connection configurations are supported by IP-CME.
When the call control signalling is transmitted over IP-based networks via IP-CMEs, there are five
interfaces that the IP-CMEs should have, as shown in Figure 3. On the other hand, in the
configuration using SS7 networks to transmit the call control, there are four interfaces as shown in
Figure 4 below.
SW signalling I/F

. IP-based .
ISC . IP-CME
networks
IP-CME
.
ISC
. .
. .

IP transmission channel I/F G.769_F03

Figure 3/G.769/Y.1242 – Network connection interfaces of the IP-CME

SS7
networks
SW signalling I/F

ISC . IP-CME
IP-based
IP-CME . ISC
. networks .
. .
. .

G.769_F04
IP transmission channel I/F

Figure 4/G.769/Y.1242 – Network connection interfaces of the IP-CME using SS7 networks

6 ITU-T Rec. G.769/Y.1242 (06/2004)

6.2.1 Trunk I/F
This interface, such as T1, E1, T3, and E3, should be used to transmit the voice, facsimile and
VBD signals between the IP-CME and ISC.
– Trunk side interface at 1544 kbit/s;
– Trunk side interface at 2048 kbit/s;
– Trunk side interface at 34 368 kbit/s;
– Trunk side interface at 44 736 kbit/s.
6.2.2 ISC signalling I/F
This is the signalling interface between ISC and IP-CME and supports the signals to control the
IP-CME from ISC.
6.2.3 Packetized IP-CME control signalling I/F
This interface provides the IP-CME control signals between them via IP-based networks.
6.2.4 Packetized call control signalling I/F
This interface provides the call control signals between ISCs via IP-based networks.
6.2.5 IP transmission channel I/F
This interface provides the bearer signals between IP-CMEs via IP-based networks.

7 Functions of the IP-CME

The functions of IP-CME are shown as follows. The architecture of the stream handling functions
of both modes is shown in Figure 5. A more detailed functional model of the IP-CME is shown in
Figure 6.

7.1 Packetized transmission modes and their functions related to stream handling
function
There are several standard VoIP applications defined in Annex B. The IP-CME supports two
packetized transmission modes: packetized transmission mode A and packetized transmission
mode B, in order to transmit those VoIP applications. The definitions of the modes are as follows:
1) Packetized transmission mode A
A transmission mode without using the application header based on the VoIP application
defined in Annex B.
2) Packetized transmission mode B (Optional)
A transmission mode using the application header based on the VoIP application defined in
Annex B.

ITU-T Rec. G.769/Y.1242 (06/2004) 7

 1) Packetized transmission mode switching function
a) Packetized transmission mode A
b) Packetized transmission mode B (Optional)
– RTP/RTCP
Signal processing – T.38
function – V.150.1
Trunk – RTP Compression
I/F 2) Short packet packetizer/depacketizer
.
. function 3) Short packets multiplexer/demultiplexer
. 4) RTP layer
.
.
.

Control function

G.769_F05
Signalling
Data stream

Figure 5/G.769/Y.1242 – Basic functions related to audio stream handling

7.1.1 Trunk I/F function

This function provides connection to PSTN network and distribution of the TDM channels for
signal processing.
7.1.2 Signal processing function
This function processes the voice, facsimile and VBD signals of the call. The signal processing
function list may include voice compression, signal analysis, fax relay or bypass, modem relay or
bypass, echo canceller, DTMF detector, etc. The signal processing function generates frames of
information that are applied to the packetizer function.
7.1.3 Packetized transmission mode switching function
The following are two packetized transmission modes. This function switches the modes.
a) Packetized transmission mode A;
b) Packetized transmission mode B (Optional).
This mode is used for the transmission of the signals with the application header defined in
Annex B.
7.1.4 Short packets packetizer function
This function builds short packets consisting of one or more frames of the signals transmitted.
7.1.5 Short packets multiplexer/demultiplexer function
This function combines short packets into one multiplexed structure packet. It demultiplexes
combined packets on the receiver side using the short packet header and sends the short packets to
the appropriate channel of the signal processing function.
7.1.6 RTP layer function
This function provides RTP capability between two IP-CMEs (IETF RFC 3550, IETF RFC 2833).
The UDP/IP interface function consists of the UDP functionality and IP packets layer-3
functionality/layer-2 protocol functionality and layer-1 physical interface.

8 ITU-T Rec. G.769/Y.1242 (06/2004)

 Packetized call
control signalling I/F

Packetized IP-CME
control signalling I/F

G.769_F06

Figure 6/G.769/Y.1242 – Functions of the IP-CMEs

The functional units of Figure 6 are briefly described in the following clauses.

7.2 Trunk I/F access function

7.2.1 Layer 1
Layer-1 protocols may include any of the following:
– ITU-T Recs G.703, G.704, G.957 and IEEE 802.
7.2.2 TDM signalling interface
TDM Bearer signalling is accomplished by means of signalling on the TDM bearer interface. TDM
signalling interfaces supported by this Recommendation should conform to national standards and
are for future study.
If SS7 links are used, then signalling on the TDM bearer interface is not used.
Support of the following and other signalling types is for further study:
– SS7 Signalling;
– ITU-T Rec. Q.931;
– R1 Signalling System – ITU-T Q.300 series Recommendations;
– R2 Signalling System – ITU-T Q.400 series Recommendations;
– Channel Associated as per ITU-T Rec. G.704.

7.3 ISC signalling I/F control function

This signalling is used for dynamic load control of calls in PSTN side and control of echo
cancelling mechanism.
See ITU-T Recs Q.50, Q.50.1 and Q.52.

ITU-T Rec. G.769/Y.1242 (06/2004) 9

7.4 IP transmission channel I/F access function
7.4.1 Layer 3
Layer-3 protocols may include any of the following:
– DHCP – IETF RFC 2131;
– IPv4 Router – IETF RFC 1812;
– Support of IPv6 – IETF RFC 2460.
7.4.2 Layer 2
Layer-2 protocols may include any of the following:
– PPP – IETF RFC 1661;
– Frame Relay – ITU-T Rec. I.233;
– ATM – ITU-T Recs I.363.1, I.363.2 and I.363.5;
– IP over PPP, as per IETF RFC 1661;
– IP over Frame Relay, as per IETF RFC 2427;
– IP over ATM.
7.4.3 Layer 1
Layer-1 protocols may include any of the following:
– ITU-T Recs G.703, G.704, G.957 and IEEE 802.
7.4.4 IP transmission channel control procedure
Annex A provides details about the IP transmission channel control procedure for the relevant
multiplex structure.
Figure 7 illustrates the configuration of the IP transmission channel between the IP-CMEs.

IP transmission channel 1

IP transmission channel 2

IP transmission channel 3

IP-CME 1 IP-CME 2

IP transmission channel N

G.769_F07
A set of IP ports
A set of trunks A multiplexed IP/UDP/RTP Stream

Figure 7/G.769/Y.1242 – Configuration of IP transmission channel between the IP-CMEs

10 ITU-T Rec. G.769/Y.1242 (06/2004)

7.5 Packetized IP-CME control signalling function
7.5.1 Definition of the IP-CME profile
This signalling is used for exchanging the profile of IP-CME. The profile includes the following
information.
a) IP network information;
One IP-CME has following three IP addresses:
1) For Packetized IP-CME control signalling I/F and Packetized call control
signalling I/F;
2) For IP transmission channel I/F;
3) For management use.
NOTE 1 – IP-CME(s) under DHCP environment should automatically collect the IP addresses.
NOTE 2 – The number of UDP port numbers of the IP transmission channel depends on the number
of the IP transmission channels between IP-CME(s).
b) Coding types of IP transmission channels (Optional);
c) Range of call ID values (Optional);
d) Range of trunk ID values (Optional);
e) Selected multiplication algorithms;
f) Information of Network management;
Access speed (bandwidth), average one-way delay time between IP-CMEs based on SNMP.
g) Information of QoS policy management.
IP-CME should have the following parameters which are calculated by Sender/Receiver
reports of RTCP:
– MeanDelay (Sender report of RTCP):
Mean value of delay for the current reported time interval;
– Max_ MeanDelay (Sender report of RTCP):
Largest value of MeanDelay;
– CumulativeNumberOfPacketsLost (Receiver report of RTCP):
Cumulative number of packets lost of last sent RTCP Receiver Report;
– MeanJitter (Receiver report of RTCP):
Average value of CalculatedJitter calculated from all sent RTCP Receiver Reports for
the current reported time interval;
– Max_MeanJitter (Receiver report of RTCP):
Largest value of MeanJitter.
7.5.2 Procedure of the profile exchange
Profile information should be exchanged, if required, when the new channel profiles are
implemented in IP-CME and diagnosis of own functionalities is initiated.
(Details are for further study.)
7.5.2.1 Off-line procedure
This issue is for further study.

ITU-T Rec. G.769/Y.1242 (06/2004) 11

7.5.2.2 On-line procedure
The profiles shall be exchanged once before the IP-CME starts operation. The transmission protocol
for the profile shall be FTP.

7.6 Transport of call control signalling function

7.6.1 Transmission over IP-based networks
Both of the channels (TDM time slots) in an ISC, such as call signalling channel and bearer
channels (voice, facsimile and VBD channels), are connected to IP-CME. In short, the call
signalling messages are passed transparently between IP-CMEs over IP-based networks using the
following means. See Figure 3.
1) SIGTRAN transmission (IETF RFC 2719, IETF RFC 2960);
2) Clear channel (64 kbit/s) transmission.
7.6.2 Transmission over SS7 networks
The call control signalling messages are sent to the existing SS7 networks and only the bearer
channels' signals (voice, facsimile and VBD signals) are connected to the IP-CME. See Figure 4.

7.7 Multiplexing function

The following items should be taken into account for the multiplexing function.
– Triggering algorithms for multiplexing.
– Conditions of QoS policies and network management should be also considered in the
algorithms.
– A buffer control mechanism for composing/decomposing the multiplexed RTP/UDP/IP
packets.
– The Voice Activity Detection (VAD) mechanism for rescheduling the multiplexing scheme.
– A mechanism for detecting the payload types of the GSTN streams such as voice, facsimile
and VBD signals for selecting the multiplexing scheme, and switching the multiplexing
schemes ON/OFF.
– Scheduling mechanism for controlling the multiplexed packet streams between the
IP-CMEs.
7.7.1 Algorithms for generating multiplexed packets
For multiplexing schemes, several algorithms can be used to determine the length and emission
timings of multiplexed packets. The following subclauses describe selectable multiplication
algorithms. The circuit multiplication schemes may be achieved in one of the following schemes.
Table 1 summarizes the variants of the scheme considered with respect to the algorithms for
generating the multiplexed packets.

Table 1/G.769/Y.1242 – Variants of the schemes

Variants Features Parameters Implementation
Scheme 1 Fixed threshold on packet payload length L Mandatory
Scheme 2 Dynamic threshold on packet payload length L (M, A) Option
Scheme 3 Periodic packet emission T Mandatory
Scheme 4 Combination of schemes 1 and 3 L and T Option

12 ITU-T Rec. G.769/Y.1242 (06/2004)

7.7.1.1 Scheme 1: Triggering by fixed payload length threshold
A multiplexed packet is constructed by packet emission triggering based on a fixed payload length
threshold. The following set of steps gives the procedure of the algorithm where the parameter L
indicates a pre-specified threshold value of the multiplexed packet payload length in bytes. Figure 8
depicts a sketch of the packetization process in this scheme.
Step 1) Set a threshold value L for a packet payload;
Step 2) If the total amount of short packets generated and collected for a multiplexed packet
becomes equal to or greater than the threshold L, then send out the multiplexed packet. It
should be noted that the delay of short packets waiting for transmission might vary largely
with the traffic load change. For instance, when only a few number of voice streams are in
progress, then the delay becomes longer.

Channel 1

Channel 2

. 3
Channel
..
.. Trigger
Channel N

A block of the coded signal in each fixed codec frame

G.769_F08

Figure 8/G.769/Y.1242 – Multiplexing and packetization

by triggering of fixed payload length threshold

7.7.1.2 Scheme 2: Triggering by dynamic payload length threshold

In order to have a more flexible threshold method, an algorithm in which the payload length
threshold is dynamically changed as the function L (M,A) is defined. In this function, M represents
the time-varying number of voice streams in progress through the multiplexing device, and A is a
constant that may be used to represent the speech activity ratio in the streams. The following set of
steps gives the procedure, and a sketch of the packetization process in this scheme is shown in
Figure 9.
Step 1) Set a constant A;
Step 2) Update the value M when a voice stream is newly setup or released to calculate the current
value of L (M,A);
Step 3) If the total amount of short packets generated and collected for a multiplexed packet
becomes equal to or greater than the threshold L (M,A), then send out the multiplexed
packet. For example, the function L (M,A) = 10 × M × A is used assuming only a G.729
whose frame interval is 10 ms and is used as a low rate codec. By this function, it is
expected that L (M,A) gives an estimate of the amount of short packets generated during a
certain period such as a coding frame, and thus the variation of waiting delay for
transmission may be lessened.

ITU-T Rec. G.769/Y.1242 (06/2004) 13

 Channel 1

Channel 2

. 3
Channel
..
..
Channel N Trigger

A block of the coded signal in each fixed codec frame

G.769_F09

Figure 9/G.769/Y.1242 – Multiplexing and packetization by triggering of

dynamic payload length threshold

7.7.1.3 Scheme 3: Triggering by timer

In Scheme 3, a periodical timer is used in order to determine the timing to send out a multiplexed
packet. The basic scheme is to use a fixed timer value that is specified beforehand. The following
set of steps gives the procedure of the algorithm where the parameter T indicates a pre-specified
timer value. Figure 10 depicts a sketch of the packetization process in this scheme.
Step 1) Set T to determine the timing of a multiplexed packet to be constructed. The trigger is
activated periodically throughout the multiplexing operation;
Step 2) When a trigger is activated by T, collect the short packets generated and stored, up to this
moment, from the circuits concerned to construct the next multiplexed packet to be sent.

Channel 1

Channel 2

. 3
Channel
..
..
Channel N Trigger

A block of the coded signal in each fixed codec frame

Figure 10/G.769/Y.1242 – Multiplexing and packetization

by triggering of periodical timer

7.7.1.4 Scheme 4: Combination of schemes 1 and 3

The scheme 4 that we consider is based on an algorithm that is derived by combining those of
schemes 1 and 3, that is, a combined use of triggering by a periodic timer and by a fixed payload
length threshold.
NOTE – This scheme is based on a multiplexing algorithm whose triggering mechanism is a combination of
triggering by timer and triggering by fixed payload length threshold and, especially, pursues the shortening
of packetization delay under the heavy channel load condition by controlling the generation of packets with a
lengthy RTP payload.

14 ITU-T Rec. G.769/Y.1242 (06/2004)

However, from the viewpoint of reduction of the header overhead ratio, the scheme might be
disadvantageous because shorter RTP packets tend to be generated by triggering by timer just after
RTP packets are generated by triggering by a fixed payload length threshold, especially under the
heavy channel load condition as shown in Figure 11. The evaluation results using a prototype
system of the comparisons of the header overhead in Figure 12 also show this trend.
5 blocks 5 blocks 5 blocks
1 block 3 blocks 2 blocks

Channel 1

Channel 2

Channel
. 3
..
..
Channel N G.769_F11

Triggering by timer

Figure 11/G.769/Y.1242 – The packetization mechanism in scheme 4

0.6

0.5
Header overhead ratio

0.4

0.3

0.2

G.769_F12
0.1
0 20 40 60 80 100
Number of Channels N

Figure 12/G.769/Y.1242 – Header overhead ratio

7.8 Application transmission function

The short packets are constructed by this function. Two distinctive short packet formats are
provided based on the packetized transmission modes in 7.1.
The short packets are constructed with the short packet header in the packetized transmission
mode A. The additional application header is set next to the short packet header in the packetized
transmission mode B (optional).

7.9 Multiplexing load control function

This function provides interworking between the multiplexing function and the functions of
QoS policy management or network management.
(Details are for further study.)

ITU-T Rec. G.769/Y.1242 (06/2004) 15

7.10 System operation management function
Management mechanisms for equipment faults and bearer interface faults on the
GSTN side/IP network side and for maintenance operations are provided by this function.
(Details are for further study.)

7.11 QoS policy management function

7.11.1 QoS requirements and measures
To achieve QoS requirements, following measures shall be executed:
a) Clarity measurements
The objective quality measurement methods such as ITU-T Rec. P.862 Perceptual
evaluation of speech quality (PESQ) should be implemented.
b) Voice activity measurements
The chopping of the voice stream due to packet losses and other impairment factors should
be measured.
c) Delay measurement
Delay does not affect the intelligibility, but rather the character, of a speech conversation.
The measurement mechanism should be implemented.

7.12 Network management function

This issue is for further study.

8 Structure of the multiplexed packet

There are alternative methods with respect to structuring the short packet payload, header and
IP packet payload. Figure 13 depicts a short packet header and the IP packet structure in
multiplexing schemes.

G.769_F13

Figure 13/G.769/Y.1242 – IP packet structure and short packet header elements

The length of the short packet header, which has information to reconstruct the original
RTP/UDP/IP header, is set to be either 2-, 3- or 4-bytes based on the application types for
multiplexing. Figure 14 shows the format of the short packet header. Following are the entries and
their meanings.

16 ITU-T Rec. G.769/Y.1242 (06/2004)

 0 0 0 0 0 0 0 0 0 0 1 1 1 1 1 1 1 1 1 1 2 2 2 2 2 2 2 2 2 2 3 3
0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1

X Payload Length; Y IP Port ID;

1 PL (7 bits) 1

X PL (7 bits) Y IPP-ID (15 bits)

1 0

X PL (15 bits) Y IPP-ID (7 bits)

0 1

X PL (15 bits) Y IPP-ID (15 bits)

0 0
G.769_F14

Figure 14/G.769/Y.1242 – Format of the short packet header

8.1 PL indicator bit (X)

This bit indicates the length of the PL.
1: It means that the length of the PL is 7 bits.
0: It means that the length of the PL is 15 bits.

8.2 Payload Length (PL)

This field should indicate the exact short packet size including short packet header, except for the
case of X-bit is set to 1 and all PL bits are set to 1. This exceptional case is applied only for short
packet size of 162 bytes consisting of two bytes for short packet header and 160 bytes for payload.

8.3 IPP-ID indicator bit (Y)

This bit indicates the length of the IPP-ID.
1: It means that the length of the IPP-ID is 7 bits.
0: It means that the length of the IPP-ID is 15 bits.

8.4 IP port ID (IPP-ID)

The IPP-ID is used to identify the stream (call) at the IP-CME. The IP-CME can simultaneously
support several IP/UDP/RTP connections called "IP transmission channel".
When the Y-bit is set to 1, the length of the IPP-ID is 7 bits, and if Y-bit is set to 0, the length of the
IPP-ID is 15 bits.

ITU-T Rec. G.769/Y.1242 (06/2004) 17

 Annex A

IP transmission channel control procedure for

packetized transmission mode A

This annex provides details about the control procedure of IP transmission channel and structure of
the multiplexed packet for packetized transmission mode A, defined in 7.1.

A.1 Conditions
The following are conditions for the control of IP transmission channels.
An IP transmission channel is established or released depending on the following conditions:
1) the number of the streams of the IP ports within an IP transmission channel;
2) the type of the codec that is used in each call;
3) the requirements of the QoS of calls.
Furthermore, when the type of the codec is changeable in the same call (e.g., speech to facsimile), a
detection mechanism of the type of the codec is needed and the following are examples of the
detectors.
VBD/End-of-VBD signals detector, FAX/End-of-FAX signals detector, speech detector.
An IP transmission channel may accommodate streams of the IP ports having the same coding type
in order to simplify the triggering mechanisms of multiplexing schemes and reduce packetization
delay.
A call has two directional streams such as from IP-CME A to IP-CME B, and vice versa. When a
coding type of one direction of a call is different from the other, each stream can be accommodated
by a different IP transmission channel.
The maximum number of calls multiplexed onto one IP transmission channel is pre-assigned and
the IP port ID (IPP-ID) identifies every call in the channel. When the number of IPP-ID exceeds the
maximum number, a new IP transmission channel shall be established.
The ID of the IP transmission channel is defined as a pair of numbers of the UDP ports on both
sides of the IP-CMEs. Furthermore, a combination of the IPP-ID and ID of the IP transmission
channel distinguishes a call.
When the number of calls in a channel falls to zero, and after a timer interval T, the IP transmission
channel is released.

A.2 Parameters
The Trunk ID and Call ID versus the coding types is presented as shown in Table A.1. A Call ID
distinguishes a call that is a voice stream connected on an IP transmission channel through the
IP-CME. The maximum number of the Call ID depends on the number of trunks on the PSTN I/F
"trunk channel number". Each Trunk is distinguished by the Trunk ID.

Table A.1/G.769/Y.1242 – Coding types of a call

Trunk ID Call ID Coding type
101 1 0000
102 2 0011
... ... ...

18 ITU-T Rec. G.769/Y.1242 (06/2004)

Table A.2 shows the coding features such as the algorithm name, compression bit rate and voice
transfer structure. The call ID is related to Trunk ID which is shown in Table A.1.
NOTE – The coding algorithms that are shown in Table A.2 are nothing but the examples. A variety of
coding algorithms such as the higher low bit rate codings and the variable bit rate ones should be supported
in compliance with the future requirements.


m 

= 1~12
Table A.2/G.769/Y.1242 – Coding features

Compression bit rate Voice transfer

Coding type Algorithm name
(kbit/s) structure (Octets)
0000 PCM A-law (G.711) 64 40 × m
0001 56 35 × m
0010 48 30 × m
0011 PCM µ-law (G.711) 64 40 × m
0100 56 35 × m
0101 48 30 × m
0110 ADPCM (G.726) 40 25 × m
0111 32 20 × m
1000 24 15 × m
1001 16 10 × m
1010 LD-CELP (G.728) 16 10 × m
1011 CS-ACELP (G.729) 8 10 × m
1100 MP-MLQ (G.723.1) 6.3 24
1101 ACELP (G.723.1) 5.3 20
1110 GSM-EFR 13 20
1111 – – –

Table A.3 shows a mapping of the coding type and ID of the IP transmission channel.


m 

= 1~12
Table A.3/G.769/Y.1242 – ID of IP transmission channels

Coding type allocation on an IP transmission channel

Coding type
8 kbit/s 16 kbit/s 32 kbit/s 64 kbit/s 128 kbit/s
0000 1 2 3,4,5 6,7,8...9 10
1101 – – 11,12,13 14,15...20 21
... ... ... ... ... ...

Call ID

ITU-T Rec. G.769/Y.1242 (06/2004) 19

Table A.4 shows relation between ID of the IP transmission channel and the call ID.

Table A.4/G.769/Y.1242 – Attributes of IP transmission channels

Maximum The total

UDP port UDP port
ID of IP ID of call stream number of
number of number of
transmis- other number of an calls on an
Transmit Receive Call ID IPP-ID
sion party IP IP
part of the part of
channel IP-CME transmission transmission
IP-CME IP-CME
channel channel
1 15001 1 16001 64 40 1,2,12, 1,2,3,
10... 4...
2 15002 2 16002 64 60 3,7,9, 1,2,3,
11… 4...

3 15003 3 16003 32 10 4,13… 1,2…

... ... ... ... ... ... ... ...

A.3 Procedure
The following steps show a control procedure for the IP transmission channel. Figure A.1 is a
conceptual sketch of the control.
NOTE – Signalling part of Call control (A-1, B-1) in IP-CME is required when the call control signalling is
transmitted over IP-based networks.
A.3.1 Transmit side of IP-CME A
Step 1) The switching section of the IP-CME A (A-4) distributes the TDM side call control
signalling and data stream to the Signalling part of Call control (A-1) and Coding part
Channel #n (A-5-n), respectively.
Step 2) The signalling is forwarded to the signalling sections of the call control in the IP-CME B
(B-1) via signalling sections of the call control of the IP-CME A (A-1).
The data stream is transmitted to the coding section channel #n (A-5-n) to set the proper
encoding scheme and requests the packetization section channel #n (A-6-n) of the
IP transmission channel to set the proper short packet header information and to operate the
scheduled multiplexing scheme.
The packetized transmission mode A is selected in this procedure (A-7). The short packets
are constructed without adding the application header in each packetization section channel
#n (A-6-n) of the IP transmission channel.
The proper encoding scheme is provided by IP-CME profile.
Step 3) The management part of the IP transmission channels and IP-CME control (A-2) checks the
coding features in the Table #A2 which is defined based on the IP-CME profile and updates
the Table #A1. The A-2 also records the information of the Call ID and the coding type in
Table #A1.
Step 4) The A-2 checks the coding type and the maximum call stream number of the
IP transmission channel in Table #A3 and requests A-4 to assign the call stream to the
proper IP transmission channel. The A-2 also sends information on the IPP-ID, encoding
type.
Step 5) The A-2 updates Table #A4 and sends the modified information of Table #A4 to the other
party IP-CME (IP-CME B) via the signalling section of the IP transmission channels and
IP-CME control (A-3).

20 ITU-T Rec. G.769/Y.1242 (06/2004)

A.3.2 Receive side of IP-CME B
Step 1) The B-1 receives the signalling and forwards it to the switching section (B-4).
Step 2) The signalling section of the IP transmission channel and IP-CME control (B-3) receives
the updated information of Table #B4 and forwards it to the management section of
IP transmission channel and IP-CME control (B-2). The B-2 updates Tables #B4 and #B1.
Step 3) The B-4 chooses a trunk and sends a signalling message to the PSTN. The signalling
messages received at B-1 are also forwarded to B-2.
Step 4) The B-2 checks the coding type and requests B-4 to set the proper decoding scheme at
B-5-n taking into account the information of Table #B4.
Step 5) The depacketization section Channel #n (B-6-n) receives the multiplexed stream and checks
the short packet header information such as the IPP-ID and forwards it to B-4. The
application header is not checked when the packetized transmission mode A is selected
(B-7). The B-4 distributes the signal of the short packets to the proper trunk based on the
IPP-ID and information such as the Call ID provided by B-2.

G.769_FA.1

Figure A.1/G.769/Y.1242 – Conceptual block diagram of IP transmission channel control

ITU-T Rec. G.769/Y.1242 (06/2004) 21

The examples of the characteristics of standard audio encodings are shown in Table A.5.

Table A.5/G.769/Y.1242 – Properties of audio encodings

Encoding format Bit rate (kbit/s) Sample/frame Bits/sample ms/frame
G.711 (A-law, µ-law) 64 Sample 8 –
G.723.1 5.3/6.3 Frame – 30
G.729 8 Frame – 10

22 ITU-T Rec. G.769/Y.1242 (06/2004)

 Annex B

IP transmission channel control procedure for

packetized transmission mode B
B.1 Introduction
This annex provides the detailed explanation about the packetized transmission mode B specified
in 7.1. This method supports multiplexing of different telephony services over the same multiplexed
structure (speech, unrestricted 64 kbit/s, N × 64 kbit/s, fax relay, etc.).
The multiplexing scheme supports using of such VoIP applications as:
– RTP transport;
– RTCP transport;
– RTP header compression;
– RTP redundancy (RFC 2198);
– DTMF digits relay, telephony tones and telephony events (RFC 2833);
– Facsimile demodulation/remodulation (ITU-T Rec. T.38);
– Modem relay (ITU-T Rec. V.150.1).
The open structure of the multiplexing method enables the addition of future IP/UDP applications.
Due to reuse of standard protocol applications, multiplexed and non-multiplexed flows may be
concurrently implemented in the same IP-CME.
Normative references relevant to and abbreviations used in this annex can also be found in clauses 2
and 3, respectively.

B.2 Approach
Using the channel multiplexing method solves two main problems of VoIP transmission: reducing
the number of transmitted packets and economizing on bandwidth required for VoIP transmission.
However, the requirement to retain the ability to use effective RTP processing opposes the
conditions for the channel multiplexing scheme.
This annex provides a multiplexing structure for transport of multiple telephony call flows between
IP-CMEs. This structure provides transmission of all application information (RTP, UDPTL for
FAX transmission or SPRT for MoIP) for each telephony channel. Application header transmission
allows effective use of all quality enhancement methods, which are developed for VoIP
transmission (such as packet loss concealment). This solution also minimizes additional processing
power for multiplexing, which is important for large-scale IP-CME. The scheme permits a
reduction in the number of transmitted packets up to theoretically minimal values, with packet delay
acceptable for large-scale IP-CME. The bandwidth reduction ratio depends on the payload length
within a packet and allows effective telephone signal transmission over the IP network. Additional
bandwidth compression may be achieved by using RTP header compression.
The multiplexing structure allows two separate quality-monitoring schemes based on
RTCP messaging, one scheme for aggregate RTP channel on IP transmission channel basis and
another for each channel (per-channel basis). RTCP packets of aggregate channel are transmitted in
non-multiplexed format. RTCP packets for per-channel basis quality monitoring are multiplexed in
the aggregate channel.

ITU-T Rec. G.769/Y.1242 (06/2004) 23

B.3 Multiplexing structure
The aggregate packet consists of the following fields:
1) IP Header – 20 bytes (as described in IETF RFC 791);
2) UDP Header – 8 bytes (as described in IETF RFC 768);
3) MUX RTP Header – 12 bytes (as described in IETF RFC 3550);
4) Channel multiplex structure, each channel information consists of:
a) Short packet header (as defined in Annex A);
b) Application header;
c) Application payload.
Figure B.1 shows the multiplexing and the short packet structures.

G.769_FB.1

Figure B.1/G.769/Y.1242 – Multiplexing and short packet structures

The multiplexing method sends multiple encapsulated packets under MUX RTP frame. As a result,
the RTP overhead per packet is reduced. The encapsulation adds two additional headers: a short
packet header and application header to each information VoIP packet.
The key idea is to concatenate multiple channel frames into a single MUX RTP frame by inserting a
length field before the beginning of channel information. The length field is used by the
demultiplexer to separate the channels within the multiplexed frame. Each encapsulated frame
within the multiplexed frame is called an RTP short packet.
B.3.1 Application header
Application header is an integral part of channel information. As a minimum, an IP-CME shall
support RTP/RTCP protocol as described in IETF RFC 3550. Optionally UDPTL (ITU-T
Rec. T.38) or SPRT (ITU-T Rec. V.150.1) protocols may be used for effective transmission of FAX
and voiceband data modem signals. Widely used RTP header compressed protocols (e.g., CRTP
(IETF RFC 2508), ECRTP (IETF RFC 3545) and ROHC (IETF RFC 3095)) significantly reduce
required bandwidth. The expected bandwidth efficiency depends on a number of factors. These
factors include multiplexing gain, expected packet loss rate across the network, and rates of change
of specific fields within the IP and RTP headers.

24 ITU-T Rec. G.769/Y.1242 (06/2004)

B.4 Short packet format for typical applications (details are for further study)
B.4.1 Short packet format for RTP application
A short packet containing a full RTP header (as described in IETF RFC 3550) is shown in
Figure B.2. Typically, a 12 bytes long RTP header is used. Other lengths of RTP header are
described in IETF RFC 3550. This format used for all types of RTP applications includes:
– Compressed and uncompressed voice;
– DTMF digits, telephony tones and telephony signals over IP networks, as described in
IETF RFC 2833;
– Redundant audio data as described in IETF RFC 2198.

G.769_FB.2

Figure B.2/G.769/Y.1242 – Short packet with full RTP header

B.4.2 Short packet format for RTCP application

The RTCP packets for per-channel based quality monitoring are multiplexed in the aggregate
packet without an application header. The short packet format for RTCP packets is shown in
Figure B.3.

G.769_FB.3

Figure B.3/G.769/Y.1242 – Short packet format for RTCP application

B.4.3 Short packet format for RTP header compression

RTP header compressed protocols CRTP (IETF RFC 2508), ECRTP (IETF RFC 3545) and ROHC
(IETF RFC 3095) may significantly reduce required bandwidth. The type of compression is
pre-configured. The format of the short packet with compressed RTP header is shown in Figure B.4.

G.769_FB.4

Figure B.4/G.769/Y.1242 – Short packet with reduced header

ITU-T Rec. G.769/Y.1242 (06/2004) 25

B.4.4 Short packet format for T.38 application
The short packet format for T.38 applications over UDP is shown in Figure B.5.

G.769_FB.5

Figure B.5/G.769/Y.1242 – Short packet format for T.38 over UDP

B.4.5 Short packet format for modem relay transmission

ITU-T Rec. V.150.1 describes procedures for transmission of V-series DCEs over IP networks. The
short packet format for modem relay application over UDP is shown in Figure B.6.

G.769_FB.6

Figure B.6/G.769/Y.1242 – Short packet format for modem relay over UDP

26 ITU-T Rec. G.769/Y.1242 (06/2004)

 Appendix I

Functional architecture

This appendix provides an example of the functional implementation of IP-CME.

I.1 Functional implementation

Figures I.1 and I.2 show examples of the functional implementation of IP-CME for transmit side
and received side respectively. Since the IP packets are bidirectional, transmitted in the form of
full-duplex communication, each IP-CME shall have both the transmit side and receive side
functionalities.
I.1.1 Transmit side of IP-CME
IP transmission channel control functional unit (A-2) communicates with the functional unit (B-2)
in the destination side. IP-CME determines a specified coding type and other profiling parameters
for controlling IP transmission channels. Call control functional unit (A-1) communicates with the
functional unit (B-1) in the destination side. IP-CME connects the calls between the origin side and
destination side IP-CMEs.

Trunk
channel #1

Trunk
channel #N

SW
signalling
control
channel G.769_FI.1

Figure I.1/G.769/Y.1242 – Example of block diagram of multiplication

in transmit side of IP-CME

In the transmit side IP-CME, the voice, facsimile and VBD signals of the call via the
communication channel is applied to the coding functional unit (A-12). The coding functional unit
encodes the signal into encoded signal in accordance with one of the coding types determined by
coding type order functional unit (A-7) based on IP transmission channel control signalling (A-2).
The silence part of the voice signal is compressed and thus the active part signal is encoded.

ITU-T Rec. G.769/Y.1242 (06/2004) 27

The encoded signal from the coding functional unit (A-12) is applied to the buffer functional unit
(A-13) and temporarily stored therein. The short packet construction functional unit (A-15) gets a
short packet header for the encoded signal of the call from the short packet header generation
functional unit (A-8). The functional unit (A-15) also gets a payload length of the short packet for
the encoded signal from the payload length indication functional unit (A-9), and then extracts from
the buffer functional unit (A-16) a part of the encoded signal with the payload length as a segment.
The application header defined in Annex B is added next to the short packet header when the
packetized transmission mode B, defined in Annex B, is chosen by the packetized transmission
mode switching functional unit (A-17).
The short packet, composed of the short packet header (SPH) and the short packet payload (SPP), is
provided for each call. In the short packet header, an IP port number (IPP-ID) and a payload length
(PL) are provided. The short packet construction functional unit (A-15) transfers the constructed
short packet to the short packet buffer functional unit (A-16).
I.1.2 Receive side of IP-CME
The IP packets transmitted from the transmit side IP-CME are received at the packet transceiver
functional unit (B-4) in the receive side IP-CME. The received IP packet is transferred to the packet
disassembler functional unit (B-3).

Trunk
channel #1

Trunk
channel #N

SW
signalling
control
channel

G.769_FI.2

Figure I.2/G.769/Y.1242 – Example of block diagram of multiplication

in receive side of IP-CME

The disassembler functional unit (B-3) disassembles the received IP packet into short packets. Then
the functional unit B-3 reads out the communication channel numbers described in the short packet
headers of the respective short packets and transfers these short packets to the corresponding short
packet buffer functional units (B-6), respectively. The short packet disassembler functional unit

28 ITU-T Rec. G.769/Y.1242 (06/2004)

(B-7) extracts the short packet and disassembles it into a short packet header and a short packet
payload when the packetized transmission mode A, defined in Annex A, is chosen by the
packetized transmission mode switching functional unit (B-10).
Then, the disassembler functional unit (B-7) transfers the coded signals in the short packet payload
to the buffer functional unit (B-8). The buffer functional unit (B-8) inserts a fill-in signal such as a
signal indicating silence between the immediately preceding segment and the current segment. The
decoding functional unit (B-9-n (n; from 1 to N)) sequentially decodes the coded signals extracted
from the buffer functional unit (B-8) to convert into ISC signals for the telephone network.
In IP-CME, an optimum coding type for the content of the information signal can be selected for
each communication channel during communication of the call. The transmit side IP-CME has, as
shown in Figure I.1, N coding functional units (A-12-1) to (A-12-N) which operate different coding
algorithms for one communication channel, N buffer functional units (A-13-1) to (A-13-N), a
communication channel monitor functional Unit (A-10), an input switch functional unit (A-11), an
output switch functional unit (A-14) and a coding type order functional unit (A-7). On the other
hand, the receive side IP-CME has, in addition, N decoding functional units (B-9-1) to (B-9-N)
which operate different coding algorithms for one communication channel and N buffer functional
units (B-8-1) to (B-8-N).

ITU-T Rec. G.769/Y.1242 (06/2004) 29

 ITU-T Y-SERIES RECOMMENDATIONS
GLOBAL INFORMATION INFRASTRUCTURE, INTERNET PROTOCOL ASPECTS AND NEXT
GENERATION NETWORKS

GLOBAL INFORMATION INFRASTRUCTURE

General Y.100–Y.199
Services, applications and middleware Y.200–Y.299
Network aspects Y.300–Y.399
Interfaces and protocols Y.400–Y.499
Numbering, addressing and naming Y.500–Y.599
Operation, administration and maintenance Y.600–Y.699
Security Y.700–Y.799
Performances Y.800–Y.899
INTERNET PROTOCOL ASPECTS
General Y.1000–Y.1099
Services and applications Y.1100–Y.1199
Architecture, access, network capabilities and resource management Y.1200–Y.1299
Transport Y.1300–Y.1399
Interworking Y.1400–Y.1499
Quality of service and network performance Y.1500–Y.1599
Signalling Y.1600–Y.1699
Operation, administration and maintenance Y.1700–Y.1799
Charging Y.1800–Y.1899
NEXT GENERATION NETWORKS
Frameworks and functional architecture models Y.2000–Y.2099
Quality of Service and performance Y.2100–Y.2199
Service aspects: Service capabilities and service architecture Y.2200–Y.2249
Service aspects: Interoperability of services and networks in NGN Y.2250–Y.2299
Numbering, naming and addressing Y.2300–Y.2399
Network management Y.2400–Y.2499
Network control architectures and protocols Y.2500–Y.2599
Security Y.2700–Y.2799
Generalized mobility Y.2800–Y.2899

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

 SERIES OF ITU-T RECOMMENDATIONS

Series A Organization of the work of ITU-T

Series B Means of expression: definitions, symbols, classification

Series C General telecommunication statistics

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 Construction, installation and protection of cables and other elements of outside plant

Series M TMN and network maintenance: international transmission systems, telephone circuits,
telegraphy, facsimile and leased circuits

Series N Maintenance: international sound programme and television transmission circuits

Series O Specifications of measuring equipment

Series P Telephone transmission quality, telephone installations, local line networks

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 and open system communications

Series Y Global information infrastructure, Internet protocol aspects and Next Generation
Networks

Series Z Languages and general software aspects for telecommunication systems

Printed in Switzerland
Geneva, 2004