ETSI ETS 300 807 ed.1 (1997-11)

Integrated Services Digital Network (ISDN); Audio characteristics of terminals designed to support conference services in the ISDN

DE/MTA-004043

Digitalno omrežje z integriranimi storitvami (ISDN) – Zvokovne lastnosti priključkov, zasnovanih za podporo konferenčnim storitvam v omrežjih ISDN

General Information

• Status: Published
• Publication Date: 04-Nov-1997
• Technical Committee: MTA - Multimedia Terminals & Applications
• Current Stage: 12 - Completion
• Due Date: 14-Nov-1997
• Completion Date: 05-Nov-1997
• Ref Project: SIST ETS 300 807 E1:2003 - Integrated Services Digital Network (ISDN); Audio characteristics of terminals designed to support conference services in the ISDN

Standards Content (Sample)

SLOVENSKI STANDARD
SIST ETS 300 807 E1:2003
01-december-2003
'LJLWDOQRRPUHåMH]LQWHJULUDQLPLVWRULWYDPL,6'1±=YRNRYQHODVWQRVWL
SULNOMXþNRY]DVQRYDQLK]DSRGSRURNRQIHUHQþQLPVWRULWYDPYRPUHåMLK,6'1
Integrated Services Digital Network (ISDN); Audio characteristics of terminals designed
to support conference services in the ISDN
Ta slovenski standard je istoveten z: ETS 300 807 Edition 1
ICS:
33.080 Digitalno omrežje z Integrated Services Digital
integriranimi storitvami Network (ISDN)
(ISDN)
35.180 Terminalska in druga IT Terminal and other
periferna oprema IT peripheral equipment
SIST ETS 300 807 E1:2003 en
2003-01.Slovenski inštitut za standardizacijo. Razmnoževanje celote ali delov tega standarda ni dovoljeno.

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SIST ETS 300 807 E1:2003

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SIST ETS 300 807 E1:2003
EUROPEAN ETS 300 807
TELECOMMUNICATION November 1997
STANDARD
Source: MTA Reference: DE/MTA-004043
Formerly: DE/TE-04043
ICS: 33.020
Key words: Audio, ISDN, conf, teleservice, terminal
Integrated Services Digital Network (ISDN);
Audio characteristics of terminals designed to support
conference services in the ISDN
ETSI
European Telecommunications Standards Institute
ETSI Secretariat
Postal address: F-06921 Sophia Antipolis CEDEX - FRANCE
Office address: 650 Route des Lucioles - Sophia Antipolis - Valbonne - FRANCE
X.400: c=fr, a=atlas, p=etsi, s=secretariat - Internet: [e-mail]
Tel.: [phone] - Fax: [phone]
Copyright Notification: No part may be reproduced except as authorized by written permission. The copyright and the
foregoing restriction extend to reproduction in all media.
© European Telecommunications Standards Institute 1997. All rights reserved.

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ETS 300 807: November 1997
Whilst every care has been taken in the preparation and publication of this document, errors in content,
typographical or otherwise, may occur. If you have comments concerning its accuracy, please write to
"ETSI Editing and Committee Support Dept." at the address shown on the title page.

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Contents
Foreword .7
1 Scope .9
2 Normative References .9
3 Definitions and abbreviations .11
3.1 Definitions .11
3.2 Abbreviations .11
4 System description.12
4.1 Audio facilities.13
4.2 Audio encoding .13
4.2.1 CCITT Recommendation G.711 encoding.13
4.2.1.1 A-law.13
4.2.1.2 μ-law .13
4.2.2 CCITT Recommendation G.722 encoding.14
4.2.3 CCITT Recommendation G.728 encoding.14
4.3 Audio decoding .14
4.3.1 CCITT Recommendation G.711 decoding.14
4.3.1.1 A-law.14
4.3.1.2 μ-law .14
4.3.2 CCITT Recommendation G.722 decoding.14
4.3.3 CCITT Recommendation G.728 decoding.14
4.4 Relative level.15
5 Speech transmission characteristics.15
5.1 Handset mode .15
5.2 Headset mode .15
5.3 Hands-free mode (single user) .15
5.4 Hands-free mode (multiple users) .15
5.4.1 Receiving volume control and sensitivity adjustments .16
5.4.1.1 Sending sensitivity adjustment .16
5.4.1.2 Receiving sensitivity adjustment.16
5.4.1.3 Receiving volume control.16
5.4.1.4 Adaptive gain control (optional) .16
5.4.2 Sensitivity-frequency response.16
5.4.2.1 Sending.16
5.4.2.2 Receiving.17
5.4.3 Loudness rating.18
5.4.3.1 Sending.18
5.4.3.2 Receiving.18
5.4.4 Terminal coupling loss.19
5.4.4.1 TCL .19
w
5.4.4.2 Stability loss.19
5.4.5 Distortion .19
5.4.5.1 Sending.19
5.4.5.2 Receiving.20
5.4.6 Out-of-band signals.20
5.4.6.1 Discrimination against out-of-band input signals (sending).20
5.4.6.2 Spurious out-of-band (receiving) .20
5.4.7 Noise .21
5.4.7.1 Sending.21
5.4.7.2 Receiving.21
5.4.8 Delay .21
Annex A (normative): Test methods.22

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A.1 General conditions for testing. 22
A.1.1 Testing environment. 22
A.1.1.1 Test room . 22
A.1.1.2 Testing arrangements. 22
A.1.1.2.1 Sending. 23
A.1.1.2.2 Receiving . 23
A.1.1.2.3 Terminal coupling loss . 24
A.1.2 Test equipment interface.24
A.1.3 Test equipment requirements . 24
A.1.3.1 Electroacoustic equipment. 24
A.1.3.2 Test signals and spectrum measurements. 24
A.1.3.2.1 Standard test signals. 24
A.1.3.2.2 Composite source signal. 25
A.1.3.3 Test signals levels . 25
A.1.3.3.1 Sending. 25
A.1.3.3.2 Receiving . 25
A.1.3.4 Test equipment for the digital interface . 25
A.1.3.4.1 Codec specifications . 25
A.1.3.4.2 Analogue interface . 26
A.1.3.4.3 Definition of 0 dBr point. 26
A.1.4 Accuracy of calibrations . 26
A.2 Testing of transmission requirements . 26
A.2.1 Sensitivity-frequency response . 26
A.2.1.1 Sending. 26
A.2.1.2 Receiving . 27
A.2.2 Loudness rating. 27
A.2.2.1 Sending Loudness Rating. 27
A.2.2.2 Receiving Loudness Rating . 28
A.2.3 Terminal Coupling Loss . 28
A.2.3.1 Weighted Terminal Coupling Loss. 28
A.2.3.2 Stability loss. 28
A.2.4 Distortion . 28
A.2.4.1 Sending. 28
A.2.4.2 Receiving . 29
A.2.5 Out-of-band signals. 29
A.2.5.1 Discrimination against out-of-band input signals (sending) . 29
A.2.5.2 Spurious out-of-band (receiving) . 29
A.2.6 Noise . 29
A.2.6.1 Sending. 29
A.2.6.2 Receiving . 29
A.2.7 CCITT Recommendation G.725 encoding. 29
Annex B (informative): Test Methods for Delay measurement. 30
B.1 Introduction. 30
B.2 Cross-correlation method . 30
B.2.1 Sending . 30
B.2.2 Receiving. 31
B.2.3 Total delay. 31
B.3 Method based on group delay . 31
B.3.1 Sending . 31
B.3.2 Receiving. 32
B.3.3 Total delay. 32
Annex C (informative): Audio alignment and practical installation guides . 33
C.1 Introduction. 33
C.2 Audio alignment and sensitivity-frequency characteristics . 33
C.2.1 Sending . 33

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C.2.1.1 Sensitivity adjustment(s) .33
C.2.1.2 In-site frequency response.34
C.2.2 Receiving .34
C.2.2.1 Sensitivity adjustment.34
C.2.2.2 In-site frequency response.34
C.3 Practical installation criteria.34
C.3.1 Maximum talker to microphone distance.34
C.3.1.1 Background noise level constraints.34
C.3.1.2 Reverberation constraints .35
C.3.1.3 Preferred maximum talker to microphone distance .36
C.3.2 Sound insulation .36
Annex D (informative): Bibliography.37
History.38

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Foreword
This European Telecommunication Standard (ETS) has been produced by the Terminal Equipment (TE)
Technical Committee and later the Multimedia Terminals and Applications (MTA) Project of the European
Telecommunications Standards Institute (ETSI).
Transposition dates
Date of adoption: 24 October 1997
Date of latest announcement of this ETS (doa): 28 February 1998
Date of latest publication of new National Standard
or endorsement of this ETS (dop/e): 31 August 1998
Date of withdrawal of any conflicting National Standard (dow): 31 August 1998

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1 Scope
This European Telecommunication Standard (ETS) specifies the audio characteristics of terminals
designed to support the audiographic conference teleservice as specified in ETS 300 675 [1]. The same
audio requirements of this ETS are also applicable to terminals supporting the Integrated Services Digital
Network (ISDN) videoconference teleservice.
This ETS does not specify the terminal procedures, both with respect to in-band signalling and to ISDN
signalling on the D channel. Also the procedures and protocols for data exchange and conference control
are outside the scope of this ETS.
2 Normative References
This ETS incorporates by dated and undated reference, provisions from other publications. These
normative references are cited at the appropriate places in the text and the publications are listed
hereafter. For dated references, subsequent amendments to or revisions of any of these publications
apply to this ETS only when incorporated in it by amendment or revision. For undated references the latest
edition of the publication referred to applies.
[1] ETS 300 675: "Integrated Services Digital Network (ISDN); Audiographic
conference teleservice; Service description".
[2] I-ETS 300 245-2 (1996): "Integrated Services Digital Network (ISDN); Technical
Characteristics of Telephony Terminals, Part 2: PCM A-law, handset telephony".
[3] I-ETS 300 245-3 (1995): "Integrated Services Digital Network (ISDN); Technical
Characteristics of Telephony Terminals, Part 3: Pulse Code Modulation (PCM)
A-law, Loudspeaking and Handsfree telephony".
[4] I-ETS 300 245-5 (1996): "Integrated Services Digital Network (ISDN); Technical
Characteristics of Telephony Terminals, Part 5: Wideband (7 kHz) Handset
Telephony".
[5] I-ETS 300 245-6 (1996): "Integrated Services Digital Network (ISDN); Technical
Characteristics of Telephony Terminals, Part 6: Wideband (7 kHz)
Loudspeaking and Handsfree telephony".
[6] I-ETS 300 245-8 (1996): "Integrated Services Digital Network (ISDN); Technical
Characteristics of Telephony Terminals, Part 8: Speech transmission
characteristics when using Low Delay Code-Excited Linear Prediction (LD-
CELP) coding at 16 kbit/s".
[7] I-ETS 300 302-1 (1996): "Integrated Services Digital Network (ISDN);
Videotelephony teleservice; Part 1: Electroacoustic characteristics for 3,1 kHz
bandwidth handset terminals".
[8] I-ETS 300 302-2 (1995): "Integrated Services Digital Network (ISDN)
Videotelephony teleservice; Part 1: Electroacoustic characteristics for 3,1 kHz
bandwidth loudspeaking and handsfree terminals".
[9] I-ETS 300 302-3 (1996): "Integrated Services Digital Network (ISDN);
Videotelephony teleservice; Part 3 Audio aspects-Wideband and Handset".
[10] I-ETS 300 144 (1996): "Integrated Services Digital Network (ISDN); Audiovisual
services, Frame structure for a 64 to 1 920 kbit/s channel and associated syntax
for in-band signalling".
[11] I-ETS 300 143 (1994): "Integrated Services Digital Network (ISDN); Audiovisual
services, In-band signalling procedures for audiovisual terminals using digital
channels up to 2 048 kbit/s".

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[12] CCITT Recommendation G.711 (1988): "Pulse code modulation (PCM) of voice
frequencies".
[13] CCITT Recommendation G.722 (1988): "7 kHz audio coding within 64 kbit/s".
[14] CCITT Recommendation G.728 (1992): "Coding of speech at 16 kbit/s using
low-delay code-excited linear prediction".
[15] CCITT Recommendation G.725 (1988): "System aspects for the use of the
7 kHz audio codec within 64 kbit/s".
[16] ITU-T Recommendation P.57 (1996): "Artificial Ears".
[17] ITU-T Recommendation P.10 (1993): "Vocabulary of terms on telephone
transmission quality and telephone sets".
[18] ITU-T Recommendation G.701 (1993): "Vocabulary of digital transmission and
multiplexing and pulse code modulation (PCM) terms".
[19] ITU-T Recommendation P.51 (1996): "Artificial Mouths".
[20] ITU-T Recommendation P.79 (1993): "Calculation of loudness ratings for
telephone sets".
[21] ITU-T Recommendation P.64 (1993): "Determination of sensitivity/frequency
characteristics of local telephone systems".
[22] CCITT Recommendation P.76 (1988): "Determination of loudness rating;
fundamental principles".
[23] TBR 3 (1995): "Integrated Services Digital Network (ISDN); Attachment
requirements for terminal equipment to connect to an ISDN using ISDN basic
access".
[24] ISO 3 (1973): "Preferred numbers-Series of preferred numbers".
[25] ITU-T Recommendation P.340 (1996): "Transmission characteristics of hands
free telephones".
[26] ITU-T Recommendation G.122 (1993): "Influence of national systems on
stability talker echo in international connections".
[27] IEC Publication 651 (1979): "Sound level meters".
[28] ITU-T Recommendation P.310 (1996): "Transmission characteristics for
telephone band (300-3400 Hz) digital telephones".
[29] ITU-T Recommendation P.50 (1993): "Artificial Voices".
[30] ITU-T Recommendation G.101 (1996): "The transmission plan".

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3 Definitions and abbreviations
3.1 Definitions
For the purposes of this ETS, the definitions provided in the referenced standards and the following
definitions apply:
Acoustic Reference Level (ARL): Acoustic level which gives -10 dBm0 at the digital interface.
audiographic terminal: Terminal supporting the audiographic teleconference service.
Hands Free Reference Point (HFRP): A point located on the axis of the Artificial Mouth, at 50 cm from
the lip ring, where the level calibration is made in free field. It corresponds to the measurement point n.11,
as defined in ITU-T Recommendation P.51 [19].
reference sphere: Sphere of radius 1 metre where the anechoic conditions of the acoustic testing
environment are verified.
lip synchronization delay: The delay introduced in the sending and receiving audio paths in order to
align the audio signals with the moving pictures respectively transmitted and received by the terminal.
digital interface: For the purposes of this ETS, the digital interface refers to the B channels available at
the coincident S and T reference points at an ISDN basic access.
3.2 Abbreviations
For the purposes of this ETS, the abbreviations used in ITU-T Recommendations G.701 [18], P.10 [17],
P.51 [19], P.57 [16], P.64 [21], P.76 [22] and P.79 [20] and the following abbreviations apply:
ARL Acoustic Reference Level
ERP Ear Reference Point
F Correction factor for receiving measurements (annex A, subclause A.1.1.2.2)
r
F Correction factor for sending measurements (annex A, subclause A.1.1.2.1)
s
F Correction factor for terminal coupling loss measurements (annex A,
tcl
subclause A.1.1.2.3)
HFRP HandsFree Reference Point
ISDN Integrated Services Digital Network
LD-CELP Low Delay-Code Excited Linear Prediction
MRP Mouth Reference Point
PCM Pulse Code Modulation
PSTN Public Switched Telephone Network
RLR Receiving Loudness Rating
rms root mean square
S/D Signal to Distortion
SB-ADPCM Sub Band-Adaptive Differential Pulse Code Modulation
SLR Sending Loudness Rating
TCL Terminal Coupling Loss
TCLw Weighted Terminal Coupling Loss
TEUT Telephone Equipment Under Test

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4 System description
For the purposes of this ETS a teleconference terminal can be represented by the functional block
diagram shown in figure 1.
Video I/O
Video Codec(s)
Audio Codec(s)
Audio I/O MUX
(incl. delay)
Network
&
Interface
Data I/O for
DEMUX
telematic aids
ISDN
End-to-end
signalling
SYSTEM
CONTROL
D-channel
signalling
Figure 1: Functional block diagram of a teleconference terminal
The following units compose a teleconference terminal:
- video I/O-includes camera(s), monitor(s) and video processing units providing functions like split
screen, picture superposition (e.g. for combining the pictures generated by telewriting and still
pictures) and text generation (e.g. for supporting conference management and text processing
features);
- audio I/O-inclusive of handset/headset and handsfree facilities, both for single users and for
multiple users;
- data I/O-used for interfacing telematic aids to the teleconference terminal (e.g. external PCs, fax
machines, telewriting facilities, electronic blackboards, etc.);
- system control-carrying out the following functions: network access through user-to-network
signalling, end-to-end in band signalling, control of audio and video codecs, control of data I/O
facilities, human interface with the user;
NOTE 1: End-to-end signalling is defined in I-ETS 300 143 [11] and I-ETS 300 144 [10].
- audio codec(s)-performing CCITT Recommendations G.711 [12], G.722 [13] and G.728 [14] coding
according to in band and D-channel signalling information;
- video codec(s)-performing moving and still picture coding according to the terminal characteristics;
- mux/demux unit-multiplexes transmitted video, audio, data and control signals into a single bit
stream and demultiplexes the received bit stream, as defined in I-ETS 300 144 [10];
- network interface-makes the necessary adaptation between the network and the terminal according
to the user-network interface requirements.

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NOTE 2: Teleconference terminals using up to 2 B channels can use the ISDN Basic Access,
while terminals designed for using more then 2 B channels can either be equipped with
more Basic Access interfaces or with a Primary Rate Interface.
The teleconference terminals can be connected through the ISDN both to other terminals, either of the
same type or of different types, or to Multipoint Conference Units. The behaviour of the terminals for both
type of connections shall be the same.
4.1 Audio facilities
The teleconference terminals are intended in principle for supporting communications between groups of
users. However, single-user terminals compliant with this ETS can also be designed for allowing a
single-user access to audiographic teleconferencing in order to permit, for instance, single users to enter
multipoint conferences or to connect end-to-end to multi-user terminals.
The following range of audio facilities can be provided by teleconference terminals:
- handset (single user);
- headset (single user);
- handsfree (single user);
- handsfree (multiple users).
Some implementations can provide audio facilities to groups of users by supplying each user with a
personal hands-free unit. The same requirements applicable to single user handsfree terminals apply to
this type of multiple user terminal.
4.2 Audio encoding
Audiographic terminals shall be able to operate both narrow band, CCITT Recommendations G.711 [12]
Pulse Code Modulation (PCM) and G.728 [14], and wide band CCITT Recommendation G.722 [13]
coding. The default mode for narrow band PCM coding is A-law, however μ-law coding shall also be
implemented.
4.2.1 CCITT Recommendation G.711 encoding
4.2.1.1 A-law
At the beginning of a call operation, mode 0F (CCITT Recommendation G.711 [12]) shall be used.
The default encoding shall be A-law.
When in mode 0U the requirements of I-ETS 300 245-2 [2] and I-ETS 300 245-3 [3] shall be met for
handset and handsfree operations respectively.
4.2.1.2 μ-law
If information is available to the terminal, either by configuration or by user input, as to whether the
destination is within a μ-law region, then this encoding law shall be used after the reception of the
ALERTING message or, if the ALERTING message is not received, the CONNECT message, or inband
signalling, as described in I-ETS 300 245-5 [4] shall be initiated. The information shall be encoded using
the μ-law at 64 kbit/s as defined in CCITT Recommendation G.711 [12].
It is the responsibility of the calling terminal to ensure that the correct encoding law is used. If no indication
on the coding law has been received during the D-channel signalling sequence or during the in-band
signalling sequence, the calling terminal shall use the default coding law wh
...