EN 62365:2005

SLOVENSKI SIST EN 62365:2006
STANDARD
januar 2006
Digitalni avdio – Digitalni vhodno/izhodni vmesnik – Prenos digitalnih avdio
signalov prek omrežij z asinhronim prenosnim načinom (ATM) (IEC
62365:2004)
(istoveten EN 62365:2005)
Digital audio – Digital input-output interfacing – Transmission of digital audio over
asynchronous transfer mode (ATM) networks (IEC 62365:2004)
ICS 33.160.30 Referenčna številka
©  Standard je založil in izdal Slovenski inštitut za standardizacijo. Razmnoževanje ali kopiranje celote ali delov tega dokumenta ni dovoljeno

EUROPEAN STANDARD EN 62365
NORME EUROPÉENNE
EUROPÄISCHE NORM September 2005

ICS 35.200; 33.160
English version
Digital audio –
Digital input-output interfacing –
Transmission of digital audio over asynchronous
transfer mode (ATM) networks
(IEC 62365:2004)
Audionumérique –  Digitalton –
Interface numérique d'entrée-sortie – Digitale Ein-/Ausgangsschnittstellen –
Transmission de l'audionumérique Übertragung von Digitalton
sur les réseaux à mode de transfert über ATM-Netzwerke
asynchrone (ATM) (IEC 62365:2004)
(CEI 62365:2004)
This European Standard was approved by CENELEC on 2005-08-01. CENELEC members are bound to
comply with the CEN/CENELEC Internal Regulations which stipulate the conditions for giving this European
Standard the status of a national standard without any alteration.

Up-to-date lists and bibliographical references concerning such national standards may be obtained on
application to the Central Secretariat or to any CENELEC member.

This European Standard exists in three official versions (English, French, German). A version in any other
language made by translation under the responsibility of a CENELEC member into its own language and
notified to the Central Secretariat has the same status as the official versions.

CENELEC members are the national electrotechnical committees of Austria, Belgium, Cyprus, Czech
Republic, Denmark, Estonia, Finland, France, Germany, Greece, Hungary, Iceland, Ireland, Italy, Latvia,
Lithuania, Luxembourg, Malta, Netherlands, Norway, Poland, Portugal, Slovakia, Slovenia, Spain, Sweden,
Switzerland and United Kingdom.

CENELEC
European Committee for Electrotechnical Standardization
Comité Européen de Normalisation Electrotechnique
Europäisches Komitee für Elektrotechnische Normung

Central Secretariat: rue de Stassart 35, B - 1050 Brussels

© 2005 CENELEC - All rights of exploitation in any form and by any means reserved worldwide for CENELEC members.

Ref. No. EN 62365:2005 E
Foreword
The text of the International Standard IEC 62365:2004, prepared by IEC TC 100, Audio, video and
multimedia systems and equipment, was submitted to the formal vote and was approved by
CENELEC as EN 62365 on 2005-08-01 without any modification.
The following dates were fixed:
– latest date by which the EN has to be implemented
at national level by publication of an identical
national standard or by endorsement (dop) 2006-08-01
– latest date by which the national standards conflicting
with the EN have to be withdrawn (dow) 2008-08-01
Annex ZA has been added by CENELEC.
__________
Endorsement notice
The text of the International Standard IEC 62365:2004 was approved by CENELEC as a European
Standard without any modification.
In the official version, for Bibliography, the following note has to be added for the standard indicated:
IEC 60027-2 NOTE Harmonized as HD 60027-2:2003 (not modified).
__________
- 3 - EN 62365:2005
Annex ZA
(normative)
Normative references to international publications
with their corresponding European publications
The following referenced documents are indispensable for the application of this document. For dated
references, only the edition cited applies. For undated references, the latest edition of the referenced
document (including any amendments) applies.
NOTE Where an international publication has been modified by common modifications, indicated by (mod), the relevant
EN/HD applies.
Publication Year Title EN/HD Year
1) 2)
-
IEC 60958-1 Digital audio interface EN 60958-1 2004
Part 1: General
1) 2)
IEC 60958-4 - Part 4: Professional applications (TA4) EN 60958-4 2003
1)
ITU-T - B-ISDN asynchronous transfer mode - -
Recommendation functional characteristics
I.150
1)
ITU-T - B-ISDN ATM Adaptation Layer specification - -
Recommendation : Type 5 AAL
I.363.5
1)
ITU-T - Digital Subscriber Signalling System No. 2 – - -
Recommendation User-Network Interface (UNI) layer 3
Q.2931 specification for basic call/connection control

1)
Undated reference.
2)
Valid edition at date of issue.

INTERNATIONAL IEC
STANDARD 62365
First edition
2004-11
Digital audio –
Digital input-output interfacing –
Transmission of digital audio
over asynchronous transfer mode
(ATM) networks
 IEC 2004  Copyright - all rights reserved
No part of this publication may be reproduced or utilized in any form or by any means, electronic or
mechanical, including photocopying and microfilm, without permission in writing from the publisher.
International Electrotechnical Commission, 3, rue de Varembé, PO Box 131, CH-1211 Geneva 20, Switzerland
Telephone: +41 22 919 02 11 Telefax: +41 22 919 03 00 E-mail: inmail@iec.ch Web: www.iec.ch
PRICE CODE
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International Electrotechnical Commission
МеждународнаяЭлектротехническаяКомиссия
For price, see current catalogue

– 2 – 62365  IEC:2004(E)
CONTENTS
FOREWORD.3
INTRODUCTION.5
1 Scope.7
2 Normative references .7
3 Terms and definitions .7
4 Format of audio data in ATM cells .8
4.1 Format of audio samples .8
4.2 Packing of sample data into cells .11
4.3 Formats.12
4.4 ATM adaptation layer .12
4.5 ATM-user-to-ATM-user indication .13
5 Switched virtual circuits .13
5.1 Addresses .13
5.2 Audio call connection: SETUP and ADD PARTY messages .13
5.3 Call disconnection .15
6 Coding of audio formats .16
6.1 Qualifying information.16
6.2 Subframe format.16
6.3 Packing of subframes into cells .17
6.4 Sampling frequency.17
7 Permanent virtual circuits .18
8 Management interface .18
8.1 Call connection: SETUP messages.18
8.2 Message encapsulation .20
8.3 Message format and action to be taken by recipient .20
8.4 Message types .20

Annex A (informative) Data protection.26
Annex B (informative) Application identifier values.28

Bibliography.29
Table 1 – Fields contained in a subframe.9
Table 2 – Combinations of subframe format and packing scheme .12
Table 3 – Default port number and packing for certain VCIs.18
Table 4 – Status enquiry message.21
Table 5 – Audio connection request message .21
Table 6 – Audio disconnection request message.22
Table 7 – Input port status message .23
Table 8 – Output port status message .24
Table 9 – Other status messages.25
Table 10 – Vendor-specific messages.25
Table A.1 – Sequence number protection field values.26
Table B.1 – Application identifier (octets 9 to 12) values in the BHLI IE.28

62365  IEC:2004(E) – 3 –
INTERNATIONAL ELECTROTECHNICAL COMMISSION
___________
DIGITAL AUDIO – DIGITAL INPUT-OUTPUT INTERFACING –
TRANSMISSION OF DIGITAL AUDIO OVER ASYNCHRONOUS
TRANSFER MODE (ATM) NETWORKS
FOREWORD
1) The International Electrotechnical Commission (IEC) is a worldwide organization for standardization comprising
all national electrotechnical committees (IEC National Committees). The object of IEC is to promote
international co-operation on all questions concerning standardization in the electrical and electronic fields. To
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with the International Organization for Standardization (ISO) in accordance with conditions determined by
agreement between the two organizations.
2) The formal decisions or agreements of IEC on technical matters express, as nearly as possible, an international
consensus of opinion on the relevant subjects since each technical committee has representation from all
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3) IEC Publications have the form of recommendations for international use and are accepted by IEC National
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between any IEC Publication and the corresponding national or regional publication shall be clearly indicated in
the latter.
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equipment declared to be in conformity with an IEC Publication.
6) All users should ensure that they have the latest edition of this publication.
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expenses arising out of the publication, use of, or reliance upon, this IEC Publication or any other IEC
Publications.
8) Attention is drawn to the Normative references cited in this publication. Use of the referenced publications is
indispensable for the correct application of this publication.
9) Attention is drawn to the possibility that some of the elements of this IEC Publication may be the subject of
patent rights. IEC shall not be held responsible for identifying any or all such patent rights.
International Standard IEC 62365 has been prepared by technical area 4: Digital systems
interfaces and protocols, of IEC technical committee 100: Audio, video and multimedia
systems and equipment.
The text of this standard is based on the following documents:
CDV Report on voting
100/753/CDV 100/838/RVC
Full information on the voting for the approval of this standard can be found in the report on
voting indicated in the above table.
This publication has been drafted in accordance with the ISO/IEC Directives, Part 2.

– 4 – 62365  IEC:2004(E)
The committee has decided that the contents of this publication will remain unchanged until
the maintenance result date indicated on the IEC web site under "http://webstore.iec.ch" in
the data related to the specific publication. At this date, the publication will be
• reconfirmed;
• withdrawn;
• replaced by a revised edition, or
• amended.
62365  IEC:2004(E) – 5 –
INTRODUCTION
This International Standard describes means for the transmission of professional audio across
digital networks, including metropolitan- and wide-area networks, to provide the best
performance with regard to latency, jitter, and other relevant factors.
Current-generation wide-area communications are based on two very similar systems,
synchronous optical network (SONET) and synchronous digital hierarchy (SDH), SONET being
used in the United States and SDH in Europe. On top of them are run integrated services digital
network (ISDN), asynchronous transfer mode (ATM), and Internet protocol (IP).
ISDN provides telephone call connections of a fixed capacity that carry one 8-bit value per
125 µs; when a call is set up, its route through the system is chosen, and the switches that
route the data are configured accordingly. Each link, between switches or between switch and
end equipment, is formatted into frames that take 125 µs to transmit, and each data byte is
identified by its position in the frame.
ATM, also called broadband ISDN, provides a service similar to ISDN, but with the capacity of
each call being specified by the caller. Links are formatted into cells, which consist of a
header and 48 data bytes; the header is typically 5 bytes long, and most of it is taken up with
the virtual channel identifier (VCI) that shows to which call the cell belongs. Call set-up,
routeing, and switching are done in the same way as in ISDN, but with calls not being
restricted to 1 byte every 125 µs.
IP provides a very different service, not designed for continuous media such as audio and
video. There is no call set-up, and each packet contains enough information within itself to
allow it to be routed to its destination. This means that the header is much larger than in the
case of ATM, typically 74 bytes, and packets will also typically be much larger, if only
because otherwise the overheads would be excessive. Each packet is liable to be routed
separately, so two packets that are part of the same flow may well take different routes. This
can mean that the one that was sent first does not arrive first.
For many professional audio applications, a round-trip time from the microphone through the
mixing desk and back to the headphones of no more than 3 ms is required. Allowing 0,5 ms
each for conversion from analog to digital and back again, it follows that the network
connections to and from the mixing desk must have a latency of less than 1 ms each. For
distances of more than about 200 km, the transmission delay alone will exceed 1 ms, but
within a metropolitan area the transmission delay should be no more than 0,25 ms (equivalent
to about 50 km), leaving 0,75 ms for packetization, queuing within switches, and
resynchronization within the receiving equipment.
Packetization delays are proportional to the size of the transmission unit (frame, cell, or
packet), and resynchronization delays depend on how evenly spaced the transmission units
are when they arrive at their destination. Both classes of delay are thus small for ISDN and
large for IP. Using the format specified in this standard to carry dual-channel IEC 60958-4
audio with a 48 kHz sampling frequency over ATM results in an inter-cell time of 125 µs, at
which ATM will have similar delays to ISDN. A higher sampling frequency or a larger number
of channels would reduce the inter-cell time and hence also the delays.
The queuing time within each ISDN switch is likely to be around one frame time or 125 µs.
The ATM documents limit the queuing time in an ATM switch to approximately the inter-cell
time for the call, which, as with the other delays, translates into performance similar to that of
ISDN for dual-channel 48 kHz IEC 60958-4 audio and better for higher sampling frequencies
or larger numbers of channels.

– 6 – 62365  IEC:2004(E)
The queuing time within an IP router for normal, best effort, Internet traffic is unbounded, and
if the router is congested, packets may simply be thrown away. Resource reservation protocol
(RSVP) (see Annex A) allows capacity to be reserved for a particular traffic flow, but it does
not guarantee that the packets will actually be routed over the links on which the capacity has
been reserved; if the flow is re-routed, it will only get a best effort service until a reservation
has been made on the new route, and it may not even be possible to make a reservation on
the new route at all.
ATM has therefore been chosen as providing a more convenient service than ISDN and
significantly better performance than IP, even when RSVP is used.
This standard does not specify a physical interface to the network because one of the
features of ATM is its ability to make a seamless connection between interfaces operating at a
wide variety of data rates and with different ways of encoding the ATM cells. Commonly used
interfaces provide 25,6 Mbit/s over category 3 structured wiring and 155,52 Mbit/s over
category-5 structured wiring or fibre-optic cable.
The physical layer section description and unique ATM abbreviations can be found in ATM
forum approved specifications. See the Bibliography.

62365  IEC:2004(E) – 7 –
DIGITAL AUDIO – DIGITAL INPUT-OUTPUT INTERFACING –
TRANSMISSION OF DIGITAL AUDIO OVER ASYNCHRONOUS
TRANSFER MODE (ATM) NETWORKS
1 Scope
This International Standard specifies a means to carry multiple channels of audio in linear
PCM or IEC 60958-4 format over an ATM layer service conforming to ITU-T Recommendation
I.150. It includes a means to convey, between parties, information concerning the digital audio
signal when setting up audio calls across the ATM network.
It does not specify the physical interface to the network.
2 Normative references
The following referenced documents are indispensable for the application of this document.
For dated references, only the edition cited applies. For undated references, the latest edition
of the referenced document (including any amendments) applies.
IEC 60958-1, Digital audio interface – Part 1: General
IEC 60958-4, Digital audio interface – Part 4: Professional applications
ITU-T Recommendation I.150: B-ISDN asynchronous transfer mode functional characteristics
ITU-T Recommendation I.363.5, B-ISDN ATM Adaptation Layer specification: Type 5 AAL
ITU-T Recommendation Q.2931: Digital Subscriber Signalling System No. 2 – User-Network
Interface (UNI) layer 3 specification for basic call/connection control
3 Terms and definitions
For the purposes of this document, the following terms and definitions apply.
3.1
asynchronous transfer mode (ATM)
networking technology in which data are carried in 48-o cells
NOTE Octet (unit symbol, o) is defined as an 8-bit data element by IEC 60027-2, which is synonymous with byte
(unit symbol, B) whenever the term, byte, is restricted to 8-bit elements.
3.2
ATM adaptation layer (AAL)
protocol layer that allows different services, such as packet transfer, to be provided on an
ATM network
3.3
ATM signaling
protocol that conveys connection management and other messages between an ATM network
and equipment attached to it
– 8 – 62365  IEC:2004(E)
3.4
audio channel
path that carries one monophonic digital audio signal
3.5
audio port
physical or virtual connector that carries a fixed number of audio channels
3.6
information element (IE)
component of an ATM signalling message
3.7
MADI
serial multi-channel audio digital interface
3.8
organizationally unique identifier (OUI)
3-o code issued by a designated agency to form globally consistent bit strings as described in
OUI and company_id assignments
3.9
user-to-user indication (UI)
single bit in the ATM cell header that can be used by the ATM adaptation layer as a marker
for certain cells
3.10
virtual channel
communications channel that provides for the sequential unidirectional transport of ATM cells
on a link between two pieces of equipment
3.11
virtual channel identifier (VCI)
numerical tag occupying a 16-bit field in the ATM cell header that identifies the virtual channel
over which the cell is to travel
3.12
virtual circuit
route through a network formed by concatenating virtual channels
3.13
virtual path
group of up to 65536 virtual channels
3.14
virtual path identifier (VPI)
numerical tag occupying an 8-bit field in the ATM cell header that identifies the virtual path
which contains the virtual channel over which the cell is to travel
4 Format of audio data in ATM cells
4.1 Format of audio samples
4.1.1 Subframes
4.1.1.1 Each audio sample shall be encoded in a subframe that consists of a whole number
of octets. The subframe shall be stored in the cell in consecutive octets, with the first bit of the
subframe in the most significant bit of the first octet.

62365  IEC:2004(E) – 9 –
4.1.1.2 A subframe shall consist of the fields listed in Table 1, in the order in which they
appear.
Table 1 – Fields contained in a subframe
Field Specified in
Audio sample word 4.1.2
Ancillary data 4.1.3
Protocol overhead 4.1.4
4.1.2 Audio sample word
4.1.2.1 The audio sample shall be represented in linear 2’s complement form, with the most
significant bit first. If the source provides fewer bits than the size of this field, the unused least
significant bits shall be set to zero.
NOTE This specification is the same as in IEC 60958-4, except that the bit order is reversed.
4.1.2.2 The number of bits in the audio sample word shall be chosen in such a way that the
total number of bits in the subframe is 8, 16, 24, 32, or 48.
4.1.3 Ancillary data
4.1.3.1 This field shall either contain no bits or consist of four bits designated B, C, U, V, in
that order.
4.1.3.2 The C, U, and V bits shall be the channel status, user data, and validity bits
specified in IEC 60958-1.
4.1.3.3 The B bit shall be a 1 for the first subframe of the block specified in IEC 60958-1,
and a 0 for all other subframes.
NOTE The B bit affects the interpretation of the C bit, and possibly also of the U bit, but has no relation to the
grouping of samples specified in 4.2.
Where more than one audio channel is carried, the B bit shall be set at the start of the block in
every channel, not just in the first channel. The block starts may be unaligned.
4.1.4 Protocol overhead
This field shall either contain no bits or consist of a sequencing bit followed by three bits that
provide data protection.
4.1.4.1 Sequencing word
The sequencing word consists of the sequencing bits of all the subframes in a cell, in the
order in which the subframes appear in the cell.
4.1.4.1.1 Sequence number
The first four bits of the sequencing word shall contain a sequence number in the form of
a binary integer with the least significant bit first.
Except in the first cell transmitted on a virtual circuit, the value of this integer shall be 1 more
(modulo 16) than in the previous cell on the same virtual circuit.

– 10 – 62365  IEC:2004(E)
The value of this integer in the first cell transmitted on each virtual circuit shall be chosen
such that in the first cell of each block, as specified in 4.5, the least significant three bits shall
be zero.
The value of the most significant bit in the first cell transmitted on each virtual circuit shall not
be defined in this standard.
4.1.4.1.2 Sequence number protection
The fifth to seventh bits of the sequencing word shall contain the 1's complement of the
3 3
remainder of the division (modulo 2) by the generator polynomial x + x + 1 of the product x
multiplied by the sequence number. The coefficient of the x term in the remainder
polynomial is the fifth bit.
The eighth bit of the sequencing word shall be such that there are an even number of 1's in
the first eight bits.
NOTE Additional information is given in Annex A.
4.1.4.1.3 Second number
The ninth to twelfth bits of the sequencing word may contain a second number in the
form of a binary integer with the least significant bit first.
a) The value of this integer in the first cell transmitted on each virtual circuit shall be defined
in this standard only as in (b). Its value in a cell which is the first cell of a block (as
specified in 4.5) and has its user indication bit set to 1 shall be 1 more (modulo 16)
than in the previous cell on the same virtual circuit. Its value in each other cell shall be
equal to that in the previous cell on the same virtual circuit.
b) Where two ATM virtual circuits carry data from sources that use the same local clock as
specified in 4.5, there may be a defined relationship between the second number values
on the two connections which can allow co-temporal samples on the two connections to be
identified. The method by which the necessary information is conveyed to receiving
equipment is not specified in this standard.
NOTE The second number may be used to identify samples uniquely within a 16-second period.
c) If the sender does not support the inclusion of the second number, these four bits shall
be zero in every cell.
4.1.4.1.4 Remainder of sequencing word
Any further bits in the sequencing word shall be reserved and shall be set to zero on
transmission and ignored on reception.
4.1.4.2 Data protection bits
If the ancillary data field contains a V bit, the three data protection bits shall contain the
1's complement of the remainder of the division (modulo 2) by the generator polynomial x + x
+ 1 of the sum of the product x multiplied by the most significant nine bits of the audio
sample and the product x multiplied by the V-bit.
Otherwise, the three data protection bits shall contain the 1's complement of the remainder of
3 3
the division (modulo 2) by the generator polynomial x + x + 1 of the product x multiplied by
the most significant nine bits of the audio sample.
In either case, the coefficient of the x term in the remainder polynomial is the first (most
significant) of the three bits.
NOTE This protection scheme is appropriate for linear PCM audio samples. Other data types carried in these
streams may need to arrange additional protection within their codecs.

62365  IEC:2004(E) – 11 –
4.2 Packing of sample data into cells
4.2.1 Packing schemes
4.2.1.1 An ATM virtual circuit shall carry either a single audio channel or a group of audio
channels. In the latter case, all audio channels in the group shall use the same format and
share the same sample clock.
For the purpose of this description, audio channels shall be numbered from 1 upwards. In the
examples, the sample times are given letters, so for instance 2a is the first sample on audio
channel 2 and 2b is the second.
4.2.1.2 The number of samples per cell shall be 48 divided by the number of octets in a
subframe (see 4.1.1).
4.2.1.3 On each ATM virtual circuit, one of the packing schemes specified in 4.2.2, 4.2.3,
and 4.2.4 shall be used. To assist interoperability, temporal grouping should be used in
preference to grouping by channel.
NOTE Only certain combinations of subframe size and number of audio channels are possible; if necessary, an
application may leave some audio channels unused.
4.2.1.4 The audio sample data in every subframe of an unused audio channel shall be 0.
4.2.2 Temporal grouping
The number of samples per cell shall be divisible by the number of audio channels.
Co-temporal samples shall be grouped together. Samples within a group shall be in audio
channel number order and groups shall be in temporal order.
A block, for the purposes of 4.5, shall consist of eight cells.
EXAMPLE
2 channels, 12 samples per cell: 1a, 2a, 1b, 2b, 1c, 2c, 1d, 2d, 1e, 2e, 1f, 2f.
4.2.3 Multi-channel
The number of audio channels shall be divisible by the number of samples per cell.
Samples shall be in channel number order.
A block, for the purposes of 4.5, shall consist of eight sets of samples.
EXAMPLE
24 channels, 12 samples per cell: 1a . 12a in first cell; 13a . 24a in second; 1b . 12b in
third.
4.2.4 Grouping by channel
The number of samples per cell shall be divisible by the number of channels.
Samples on the same channel shall be grouped together; samples within a group shall be in
temporal order, and groups shall be in channel number order.

– 12 – 62365  IEC:2004(E)
A block (for the purposes of 4.5) shall consist of eight cells.
EXAMPLE
2 channels, 12 samples per cell: 1a, 1b, 1c, 1d, 1e, 1f, 2a, 2b, 2c, 2d, 2e, 2f.
NOTE If there is just one channel, this scheme is identical to temporal grouping; if the number of channels is
equal to the number of samples per cell, all three schemes are identical.
4.3 Formats
4.3.1 Only those subframe formats, packing schemes, and sampling frequencies that are
expressible in the notation of Clause 6 shall be used.
NOTE See additional restrictions in 4.1.2 and 4.2.1.
4.3.2 To ensure interoperability between equipment designed for different applications,
source equipment shall be capable of transmitting at least one of the formats indicated in
Table 2, and destination equipment shall be capable of receiving all of the formats indicated
in Table 2.
Table 2 – Combinations of subframe format and packing scheme
Code Status Subframe Audio Ancillary Protocol Grouping Number of
(NOTE 1) length sample word data overhead audio
bytes length bits bits channels
bits
56 02 NOTE 2 4 24 4 4 Temporal 2
56 01 NOTE 2 4 24 4 4 Not 1
applicable
06 02 NOTE 2 3 24 0 0 Temporal 2
06 01 NOTE 2 3 24 0 0 Not 1
applicable
56 85 NOTE 3 4 24 4 4 Multi- 60
channel
NOTE 1 The code column shows the encoding (in hexadecimals) of the second and third bytes of the AAL
parameters IE as specified in 5.2.2.1 and is informative only.
NOTE 2 Required for all equipment.
NOTE 3 Required for equipment that can convey at least 56 audio channels and has sufficient network capacity to
do so.
4.3.3 When conveying 56-channel MADI data, the format with 60 channels in Table 2 shall
be used, with the last four channels being unused.
4.3.4 Sampling frequencies shall be as specified in AES5.
4.3.5 Destination equipment shall support the 48 kHz sampling frequency.
4.4 ATM adaptation layer
Audio virtual circuits shall use a user-defined ATM adaptation layer.

62365  IEC:2004(E) – 13 –
4.5 ATM-user-to-ATM-user indication
4.5.1 Cells shall be grouped into blocks as specified in 4.2.
4.5.2 The sender shall include a local clock, which ticks once per second.
NOTE This standard does not specify the accuracy of the local clock, nor to what (if anything) it is synchronized.
It need not be related to the audio sample clock.
4.5.3 For the first block transmitted after a clock tick, the ATM-user-to-ATM-user
indication (UI bit) in the cell header shall be set to 1 in the first and last cells and to 0 in
all other cells.
For all other blocks, the ATM-user-to-ATM-user indication shall be set to 1 in the last
cell and to 0 in all other cells.
NOTE If the state of the UI bit is latched as each cell is unpacked, the resulting signal can be a pulse train with
the leading edges of the pulses being evenly spaced with frequency f/8n, where f is the sampling frequency and n
the number of samples from the same channel in a cell, and with a double-width pulse once each second.
5 Switched virtual circuits
5.1 Addresses
5.1.1 The distinction between audio and other circuits, and between different types of ports,
shall be made using protocol and other information conveyed as specified in 5.2 and 8.1.
5.1.2 Source and destination ports may be different types; the calling party number (or
subaddress) shall identify a source port, and the called party number (or subaddress) shall
identify a destination port.
5.1.3 The distinction between different ports of the same type within an interface shall be
made using the Selector value.
5.1.4 An audio port may carry more than one audio channel; the protocol information
specifies how many channels are to be received. Ports with different numbers of channels
may be considered to be of different types, and the same physical port may be addressed in
more than one way; for instance, an AES3 output port may be addressed as a single stereo
port, two mono ports, a single mono port using double sampling frequency mode, or as
one of a group of three carrying a 5.1-channel signal.
5.2 Audio call connection: SETUP and ADD PARTY messages
5.2.1 Restrictions on connection requests
5.2.1.1 All audio connections shall be point to multipoint and shall be originated by the
equipment that contains the source port.
NOTE When connecting a new call, the caller sends a SETUP message and the destination equipment receives a
SETUP message; when adding a new destination, the caller sends an ADD PARTY message and the destination
equipment receives a SETUP message unless it already has a port that is a destination for that call, in which case it
receives an ADD PARTY message.
5.2.1.2 The AAL parameters, broadband high-layer information, and calling party number
information elements (IE), which are optional in the ATM signalling specification, shall be
required when connecting an audio call.

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5.2.1.3 If the called party number is not in the network service access point (NSAP) format
conforming to Table 4-12 of ITU Q.2931, the called party subaddress IE shall be required.
If the calling party number is not in the NSAP format, the calling party subaddress IE shall be
required.
5.2.2 Information elements in the SETUP and ADD PARTY messages
5.2.2.1 ATM adaptation layer parameters IE
Within the ATM adaptation layer parameters IE, the following coding shall be used.
a) The AAL type shall be coded as user-defined AAL (10 in octet 5).
b) The first octet of the User Defined AAL Information (octet 6) shall encode the
qualifying information specified in 6.1.
c) The second octet of the User Defined AAL Information (octet 6.1) shall encode the
subframe format as specified in 6.2.
d) The third octet of the User Defined AAL Information (octet 6.2) shall encode the
packing of subframes into cells as specified in 6.3.
e) The fourth octet of the User Defined AAL Information (octet 6.3) shall encode the
sampling frequency as specified in 6.4.
5.2.2.2 Broadband high-layer information IE
Within the broadband high-layer information IE, the following coding shall be used.
a) The high-layer information type shall be coded as vendor-specific application
identifier (83 in octet 5).
b) The OUI value shall be coded as 00 in octet 6, 0B in octet 7, and 5E in octet 8.
16 16 16
c) The first octet of the application identifier (octet 9) shall be coded as zero to
indicate the first edition of this standard.
d) The second octet of the application identifier (octet 10) shall be coded as zero to
indicate audio data.
The remaining octets of the application identifier (octets 11 to 12) are reserved. They
shall be coded as zero but ignored by the recipient unless specified by a means specified
outside this standard.
NOTE A different value for octet 10 is specified in 8.1.2.2. Encodings in the broadband high-layer information IE
are summarized in Annex C.
5.2.3 Destination response to SETUP and ADD PARTY messages
5.2.3.1 Destination response to SETUP message
5.2.3.1.1 A SETUP message received from the network shall be processed according to the
provisions of this standard for audio calls if it meets the following three criteria.
a) It contains a broadband high-layer information IE which conforms to 5.2.2.2 or does not
contain a broadband high-layer information IE but is in other respects consistent with
being an audio call conforming to this standard.
b) It is for a point-to-multipoint connection.
c) It contains ATM adaptation layer parameters IE indicating a user-defined AAL (10 in
octet 5).
5.2.3.1.2 If the first octet of the User Defined AAL Information (octet 6) does not
contain an encoding recognized by the equipment, the destination equipment shall reject the

62365  IEC:2004(E) – 15 –
call with cause value call rejected (octet 6 = 95 in the cause IE), rejection reason
user specific and condition permanent (octet 7 = 81 ), and the user specific
diagnostic (octet 7.1) coded as a single octet with the value zero.
5.2.3.1.3 If the remaining octets of the user defined AAL information (octets 6.1 to
6.3) indicate a format or sampling frequency which the equipment does not support, the
destination equipment shall reject the call with cause value call rejected (octet 6 = 95 ),
rejection reason user specific and condition permanent (octet 7 = 81 ), and the first
user specific diagnostic octet coded as 01.
NOTE The destination equipment should take the sampling frequency from the AAL parameters IE and not
calculate it from the cell rate in the ATM traffic descriptor IE.
5.2.3.1.4 If the Selector value, in the context of the format indicated by the ATM
adaptation layer parameters IE, does not correspond to an output port or corresponds to an
output port which is disabled, the destination equipment shall reject the call with cause value
call rejected (octet 6 = 95 ), rejection reason user specific and condition
permanent if the port does not exist, transient if it is disabled (octet 7 = 81 or 82 ,
16 16
respectively), and the first user specific diagnostic octet coded as 02 .
5.2.3.1.5 If the Selector value (in the context of the format indicated by the ATM
adaptation layer parameters IE) corresponds to an output port which would conflict with an
existing connection, or completing the connection would require some other resource that has
been used up for other calls, the destination equipment shall reject the call with cause value
user busy (octet 6 = 91 ).
NOTE Receiving equipment can include two cause IEs in the RELEASE message to give information on two
different aspects of the reason for rejection of the call. If two-cause IEs are included, the cause value specified in
this subclause may be in either of them.
5.2.3.2 Destination response to ADD PARTY message
5.2.3.2.1 An ADD PARTY message received from the network shall only be processed
according to this standard if it relates to an audio call conforming to this standard.
5.2.3.2.2 The receiving equipment may process the ATM adaptation layer parameters IE in
the same way as for a SETUP message (see 5.2.3.1), or it may check that it is the same as
that received when the call was connected and reject the call with cause invalid
information element contents (octet 6 = E4 ) citing the ATM adaptation layer
parameters IE (octet 7 = 58 ) if it is not.
5.2.3.2.3 The Selector value is processed in the same way as for a SETUP message (see
5.2.3.1).
5.3 Call disconnection
Disconnection may be initiated by either the source or the destination equipment without
giving any warning to the other party.
NOTE Disconnection can also be initiated by the network in the event of various kinds of error or if it detects that
the other party has been switched off, reset, or unplugged from the network.
The cause shall be specified as normal call clearing (90 in octet 6 of the cause IE).
NOTE This value is not included in the table in 5.4.5.15 of the UNI 3.0 specification. There are no diagnostics
associated with it.
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6 Coding of audio formats
NOTE In this clause, bits are numbered in the same way as in the ATM UNI specifications. Thus, bit 8 is the most
significant bit of the octet and bit 1 the least significant.
6.1 Qualifying information
8 7 6 5 4 3 2 1
0 0 0 0 See 6.1.2 reserved
6.1.1 Bits 8 to 5 shall be zero to indicate the formats specified in this edition of this
standard.
NOTE Other formats, with bits 8 to 5 not all 0, are reserved.
6.1.2 Bit 4 shall be coded as
0 no information about the exact frequency of the sample clock is provided;
1 sample clock is frequency-locked to a global reference.
NOTE This standard does not specify to what reference the sample clock is locked if bit 4 is a 1.
6.1.3 Bits 3 to 1 are reserved and shall be coded as zero unless otherwise specified.
6.2 Subframe format
8 7 6 5 4 3 2 1
Ancillary Overhead Sample length

6.2.1 The ancillary field shall be coded as
00 no ancillary data bits;
01 4 ancillary data bits as specified in 4.1.3;
1x reserved.
6.2.2 The overhead field shall be coded as

00 no protocol overhead bits;
01 four protocol overhead bits as specified in 4.1.4;
1x reserved.
6.2.3 The sample length field shall be coded to indicate the number of bits in the audio
sample word as
000x reserved
0010 8
0011 12
0100 16
0101 20
0110 24
0111 28
1000 32
62365  IEC:2004(E) – 17 –
1001 reserved
1010 40
1011 reserved
11xx reserved
6.3 Packing of subframes into cells

8 7 6
...