ETSI TS 102 924 V1.2.1 (2018-03)

Speech and multimedia Transmission Quality (STQ); Transmission requirements for Super-Wideband / Fullband handset and headset terminals from a QoS perspective as perceived by the user

RTS/STQ-208-1

General Information

• Status: Published
• Publication Date: 26-Mar-2018
• Technical Committee: STQ - Speech and multimedia Transmission Quality
• Current Stage: 12 - Completion
• Due Date: 30-Apr-2018
• Completion Date: 27-Mar-2018

Standards Content (Sample)

TECHNICAL SPECIFICATION
Speech and multimedia Transmission Quality (STQ);
Transmission requirements for Super-Wideband / Fullband
handset and headset terminals from a QoS perspective
as perceived by the user
2 ETSI TS 102 924 V1.2.1 (2018-03)

Reference
RTS/STQ-208-1
Keywords
QoS, terminal
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3 ETSI TS 102 924 V1.2.1 (2018-03)
Contents
Intellectual Property Rights . 5
Foreword . 5
Modal verbs terminology . 5
Introduction . 5
1 Scope . 6
2 References . 6
2.1 Normative references . 6
2.2 Informative references . 8
3 Definitions and abbreviations . 8
3.1 Definitions . 8
3.2 Abbreviations . 9
4 Applications and coder considerations . 9
4.1 Applications . 9
4.2 Coder considerations . 10
4.2.0 Premise . 10
4.2.1 Super-wideband (SWB) . 10
4.2.2 Fullband (FB). 11
5 Test considerations and test equipment . 12
5.0 Introduction . 12
5.1 IP half channel measurement adaptor . 12
5.2 Environmental conditions for tests . 12
5.3 Accuracy of measurements and test signal generation . 13
5.4 Network impairment simulation . 13
5.5 Acoustic environment . 14
5.6 Verification of the environmental conditions . 15
5.7 Influence of terminal delay on measurements . 15
5.8 Specific test considerations . 15
5.8.0 Premise . 15
5.8.1 Loudness rating and Loudness . 16
5.8.1.1 Loudness Rating . 16
5.8.1.2 Loudness . 16
5.8.2 Binaural listening . 16
6 Requirements considerations and associated measurement Methodologies . 16
6.1 Considerations . 16
6.2 Test setup. 17
6.2.1 General . 17
6.2.2 Setup for handsets and headsets . 17
6.2.3 Position and calibration of HATS . 18
6.2.4 Test signal and test signal levels . 18
6.2.5 Setup of background noise simulation . 18
6.2.6 Setup of variable echo path . 19
6.3 Coding independent parameters . 20
6.3.1 Send frequency response . 20
6.3.2 Send Loudness Rating (SLR). 23
6.3.3 Mic mute . 23
6.3.4 Linearity range of SLR . 24
6.3.5 Send Distortion . 24
6.3.5.1 Signal to harmonic distortion versus frequency . 24
6.3.5.2 Signal to harmonic distortion for higher input level . 25
6.3.6 Send Noise . 26
6.3.7 Sidetone Masking Rating STMR (Mouth to ear) . 26
6.3.8 Sidetone delay . 27
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4 ETSI TS 102 924 V1.2.1 (2018-03)
6.3.9 Terminal Coupling Loss (TCL) . 27
6.3.10 Stability loss. 28
6.4 Receive parameters. 29
6.4.1 Equalization . 29
6.4.2 Receive Frequency response . 29
6.4.3 Receive Loudness Rating (monaural reproduction) . 32
6.4.4 RLR for stereo/dichotic reproduction . 32
6.4.5 Loudness . 32
6.4.6 Receive Distortion . 32
6.4.7 Minimum activation level and sensitivity in Receive direction . 33
6.4.8 Receive Noise . 33
6.4.9 Automatic level control in receiving . 34
6.4.10 Double talk performance . 34
6.4.10.1 General . 34
6.4.10.2 Attenuation range in send direction during double talk A . 34
H,S,dt
6.4.10.3 Attenuation range in receive direction during double talk A . 35
H,R,dt
6.4.10.4 Detection of echo components during double talk . 36
6.4.11 Switching characteristics . 37
6.4.11.1 Note . 37
6.4.11.2 Activation in send direction . 38
6.4.11.3 Silence suppression and comfort noise generation . 38
6.4.12 Speech and audio quality in presence of noise. 38
6.4.12.1 Performance in send in the presence of background noise . 38
6.4.12.2 Speech quality in the presence of background noise . 39
6.4.12.3 Quality of background noise transmission (with far end speech). 40
6.4.13 Quality of echo cancellation . 41
6.4.13.1 Temporal echo effects . 41
6.4.13.2 Spectral echo attenuation . 41
6.4.13.3 Occurrence of artefacts . 42
6.4.13.4 Variable echo path. 42
6.4.14 Variant impairments; network dependant . 42
6.4.14.1 Clock accuracy send . 42
6.4.14.2 Clock accuracy receive . 43
6.4.14.3 Send packet delay variation. 43
6.4.15 Send and receive delay - round trip delay . 44
6.5 Other parameters . 45
6.5.1 Objective listening quality . 45
6.5.2 Quality of jitter buffer adjustment . 46
Annex A: Void . 48
History . 49

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5 ETSI TS 102 924 V1.2.1 (2018-03)
Intellectual Property Rights
Essential patents
IPRs essential or potentially essential to normative deliverables may have been declared to ETSI. The information
pertaining to these essential IPRs, if any, is publicly available for ETSI members and non-members, and can be found
in ETSI SR 000 314: "Intellectual Property Rights (IPRs); Essential, or potentially Essential, IPRs notified to ETSI in
respect of ETSI standards", which is available from the ETSI Secretariat. Latest updates are available on the ETSI Web
server (https://ipr.etsi.org/).
Pursuant to the ETSI IPR Policy, no investigation, including IPR searches, has been carried out by ETSI. No guarantee
can be given as to the existence of other IPRs not referenced in ETSI SR 000 314 (or the updates on the ETSI Web
server) which are, or may be, or may become, essential to the present document.
Trademarks
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Foreword
This Technical Specification (TS) has been produced by ETSI Technical Committee Speech and multimedia
Transmission Quality (STQ).
Modal verbs terminology
In the present document "shall", "shall not", "should", "should not", "may", "need not", "will", "will not", "can" and
"cannot" are to be interpreted as described in clause 3.2 of the ETSI Drafting Rules (Verbal forms for the expression of
provisions).
"must" and "must not" are NOT allowed in ETSI deliverables except when used in direct citation.
Introduction
Speech terminals are currently implementing narrowband and wideband bandwidth. Terminal equipment may offer
wider bandwidth, due to features already available in these terminals. Such equipment may implement conversational
features that may benefit of the electroacoustic equipment already available in the terminal and may provide wider
quality for the end users.
.
The present document is intended to provide initial requirements and test methods for such type of equipment
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6 ETSI TS 102 924 V1.2.1 (2018-03)
1 Scope
The present document provides speech & audio transmission performance requirements and measurement methods for
handset and headset functions of super-wideband/fullband terminals. The present document provides requirements in
order to optimize the end to end quality perceived by users.
Users become more sensitive to voice and music quality (for music used in conversational services) when using
ICT/terminal equipment and so are more demanding for further enhancement especially further extension of the audio
coded bandwidth.
For instance, this is the case for high quality conferencing services with music on hold, better background environment
rendering and longer duration than normal point to point calls.
Standardized super-wideband and fullband coders are now available, some being also compatible with wideband
coders.
The present document will consider only conversational services (that may be mixed with other services) and does not
cover the streaming-only services.
Such applications include:
• Speech and audio communication including conferencing.
• Bandwidth extension which may allow usage for some mixed content.
• Super-wideband enhancement coupled with stereo/dichotic.
The send path it can be characterized in two ways:
• The signal picked up by microphone may combine speech, music and every type of environmental signal.
• Direct insertion of any type of signal.
For receive path, signal may be combine two types:
• Communication signals such as described for send path.
• Signal coming from distributed applications (e.g. advertisement, music on hold, etc.).
2 References
2.1 Normative references
References are either specific (identified by date of publication and/or edition number or version number) or
non-specific. For specific references, only the cited version applies. For non-specific references, the latest version of the
referenced document (including any amendments) applies.
Referenced documents which are not found to be publicly available in the expected location might be found at
https://docbox.etsi.org/Reference/.
NOTE: While any hyperlinks included in this clause were valid at the time of publication, ETSI cannot guarantee
their long term validity.
The following referenced documents are necessary for the application of the present document.
[1] Recommendation ITU-T P.501: "Test signals for use in telephonometry".
[2] Recommendation ITU-T P.10/G.100: "Vocabulary for performance and quality of service".
[3] Recommendation ITU-T P.58: "Head and torso simulator for telephonometry".
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7 ETSI TS 102 924 V1.2.1 (2018-03)
[4] Recommendation ITU-T P.581: "Use of head and torso simulator for hands-free and handset
terminal testing".
[5] Recommendation ITU-T P.79: "Calculation of loudness ratings for telephone sets".
[6] Recommendation G.711.1 (2008) Amendment 4 (11/10): "Wideband embedded extension for
G.711 pulse code modulation".
[7] Recommendation ITU-T G.722.1 (annex C): "Low-complexity coding at 24 and 32 kbit/s for
hands-free operation in systems with low frame loss".
[8] Recommendation G.729.1 (05/06): "G.729-based embedded variable bit-rate coder: An 8-32 kbit/s
scalable wideband coder bitstream interoperable with G.729".
[9] Recommendation ITU-T G.718 (06/08)": "Frame error robust narrow-band and wideband
embedded variable bit-rate coding of speech and audio from 8-32 kbit/s".
[10] Recommendation ITU-T G.719: "Low-complexity, full-band audio coding for high-quality,
conversational applications".
[11] ETSI TS 103 224: "Speech and multimedia Transmission Quality (STQ); A sound field
reproduction method for terminal testing including a background noise database".
[12] ETSI ES 202 739: "Speech and multimedia Transmission Quality (STQ);Transmission
requirements for wideband VoIP terminals (handset and headset) from a QoS perspective as
perceived by the user".
[13] ETSI TS 103 739: "Speech and multimedia Transmission Quality (STQ); Transmission
requirements for wideband wireless terminals (handset and headset) from a QoS perspective as
perceived by the user".
[14] Recommendation ITU-T P.863: "Perceptual objective listening quality assessment".
[15] Recommendation ITU-T P.380: "Electro-acoustic measurements on headsets".
[16] IEC 61260-1: "Electroacoustics - Octave-band and fractional-octave-band filters - Part 1:
Specifications".
[17] Void.
[18] Void.
[19] Recommendation ITU-T G.722: "7 kHz audio-coding within 64 kbit/s".
[20] Void.
[21] Recommendation ITU-T G.711.1 (annex F): "Wideband embedded extension for G.711 pulse code
modulation".
[22] Recommendation ITU-T P.57: "Artificial ears".
[23] Recommendation ITU-T P.64: "Determination of sensitivity/frequency characteristics of local
telephone systems".
[24] ISO 3745: "Acoustics -- Determination of sound power levels and sound energy levels of noise
sources using sound pressure -- Precision methods for anechoic rooms and hemi-anechoic rooms".
[25] ETSI TR 126 952: "Universal Mobile Telecommunications System (UMTS); LTE; Codec for
Enhanced Voice Services (EVS); Performance characterization (3GPP TR 26.952 version 12.2.0
Release 12)".
[26] ETSI TS 126 441: "Universal Mobile Telecommunications System (UMTS); LTE; Codec for
Enhanced Voice Services (EVS); General overview (3GPP TS 26.441)".
[27] Recommendation ITU-T P.56: "Objective measurement of active speech level".
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8 ETSI TS 102 924 V1.2.1 (2018-03)
[28] ETSI TS 103 281: "Speech and multimedia Transmission Quality (STQ); Speech quality in the
presence of background noise: Objective test methods for super-wideband and fullband terminals".
[29] Recommendation ITU-T G.122: "Influence of national systems on stability and talker echo in
international connections".
[30] Recommendation ITU-T P.340: "Transmission characteristics and speech quality parameters of
hands-free terminals".
[31] Recommendation ITU-T P.502: "Objective test methods for speech communication systems using
complex test signals".
[32] Recommendation ITU-T P.863.1: "Application Guide for Recommendation ITU-T P.863".
[33] Recommendation ITU-T P.1010: "Fundamental voice transmission objectives for VoIP terminals
and gateways".
[34] IETF RFC 3550: "RTP: A Transport Protocol for Real-Time Applications".
[35] ETSI ES 202 737: "Speech and multimedia Transmission Quality (STQ); Transmission
requirements for narrowband VoIP terminals (handset and headset) from a QoS perspective as
perceived by the user".
[36] IETF RFC 6716: "Definition of the Opus Audio Codec".
2.2 Informative references
References are either specific (identified by date of publication and/or edition number or version number) or
non-specific. For specific references, only the cited version applies. For non-specific references, the latest version of the
referenced document (including any amendments) applies.
NOTE: While any hyperlinks included in this clause were valid at the time of publication, ETSI cannot guarantee
their long term validity.
The following referenced documents are not necessary for the application of the present document but they assist the
user with regard to a particular subject area.
[i.1] ISO 532: "Acoustics -- Method for calculating loudness level".
TM
[i.2] NIST Net .
NOTE: Available at https://www-x.antd.nist.gov/itg/nistnet/.
TM
[i.3] Netem .
NOTE: Available at http://www.linuxfoundation.org/en/Net:Netem.
[i.4] Trace Control for Netem (TCN) (2006): "Trace Control for Netem, Semester Thesis SA-2006-15",
ETH Zürich, A. Keller.
[i.5] ETSI EG 202 425: "Speech Processing, Transmission and Quality Aspects (STQ); Definition and
implementation of VoIP reference point".
[i.6] STQ(15)48-0309: "Objective Codec Evaluation of EVS. HEAD acoustics GmbH".
3 Definitions and abbreviations
3.1 Definitions
For the purposes of the present document, the following terms and definitions apply:
binaural listening: both ears are involved for the perception of sound
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9 ETSI TS 102 924 V1.2.1 (2018-03)
dichotic: relating to or involving the presentation of a stimulus to one ear that differs in some respect (as pitch,
loudness, frequency, or energy) from a stimulus presented to the other ear
diotic: pertaining to or affecting both ears (same signal in both ears)
dual channel mode: audio mode, in which two audio channels with independent programme contents (e.g. bilingual)
are encoded within one audio bit stream
fullband bandwidth: transmission of speech with a nominal bandwidth of 20 Hz - 20 kHz
stereo mode: audio mode in which two channels forming a stereo pair (left and right) are encoded within one bit stream
and for which the coding process is the same as for the Dual channel mode
super-wideband: transmission with supre-wideband bandwith which may cover at least mono capabilities. Stereo
capabilities may be possible
super-wideband bandwidth: transmission of speech with a nominal pass-band wider than 100 Hz to 7 000 Hz, usually
understood to be 50 Hz - 14 000 Hz (definition from Recommendation ITU-T P.10 /G.100 [2])
3.2 Abbreviations
For the purposes of the present document, the following abbreviations apply:
ACR Absolute Category Rating
DRP ear Drum Reference Point
ERP Ear reference Point
EVS Enhanced Voice Services
FB FullBand
GAT Group Audio Terminal
G-MOS-LQO Overall Quality Mean Opinion Score, Listening Quality Objective, fullband
F
HATS Head and Torso Simulator
MCU Multiplexing Control Unit
MRP Mouth Reference Point
MS Mid-sized Stereo
N-MOS-LQO Noise Quality Mean Opinion Score, Listening Quality Objective, fullband
F
POI Point Of Interconnection
SLR Send Loudness Rating
S-MOS-LQO Speech Quality Mean Opinion Score, Listening Quality Objective, fullband
F
SWB Super-WideBand
TCL Terminal Echo Loss
4 Applications and coder considerations
4.1 Applications
The following applications are within the scope of the present document:
• Speech and audio communication including conferencing using high quality hands free systems, for which
super-wideband/fullband coding can better reproduce the audio environment and provide improved quality and
audio immersion. These applications cover also GATs (Group Audio Terminals) and teleconference systems
such as "Telepresence".
• Bandwidth extension which may allow usage for some mixed content applications where wider bandwidth
could bring a significant added value for the customer (support of 14 kHz and 20 kHz bandwidth and
stereo/multichannel capability).
• Super-wideband enhancement coupled with stereo/multichannel to maximize the quality enhancement for the
customer when the terminal device can support this capability.
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10 ETSI TS 102 924 V1.2.1 (2018-03)
The send path can be characterized in two ways:
• The signal picked up by microphone(s) may combine speech, music and every type of environmental signal.
NOTE: For some applications (e.g. journalist reporting) the user should have the possibility to cancel the noise
environment or to transmit it without degradation.
• Direct insertion of any type of signal.
For receive path, signal may combine the two following types:
• Communication signal such as described for send path.
• Signal coming from distributed applications (e.g. advertisement, music on hold, etc.).
4.2 Coder considerations
4.2.0 Premise
As indicated in the scope only coders supporting conversational SWB and FB services are applicable to the present
document.
4.2.1 Super-wideband (SWB)
Table 0: Use cases for coders
Coder Reference Speech Other signals Stereo Remark
VoLTE (IMS) ETSI TS 126 441 [26] X X Music X
Recommendation ITU-T G.722.1 [7] X X Music For low frame loss
annex C
Recommendation ITU-T G.729.1 [8] X X background
annex E (extension SWB) noise
(X) Music
Recommendation ITU-T G.718 [9] X X Music
annex B
Recommendation ITU-T G.711.1 X X X (annex F)
annexes D [6] and F [21]
Recommendation ITU-T G.722 [19] X X X (annex D)
annexes B and D
OPUS [36] X X X
NOTE: G 722.1 [7] is intended to be used for hand-free application. It is referenced here considering
that a terminal using this coder may implement a handset or headset function.

When X is in brackets, it means that the coder is not optimized for this application.
The following coders are recommended for Super-wideband:
• Recommendation ITU-T G.722.1 [7] Low-complexity coding at 24 and 32 kbit/s for hands-free operation in
systems with low frame loss. Annex C 14 kHz mode at 24, 32 and 48 kbit/s.
The algorithm is recommended for use in hands-free applications such as conferencing where there is a low
probability of frame loss. It may be used with speech or music inputs. The bit rate may be changed at any
20 ms frame boundary. New annex C contains the description of a low-complexity extension mode to G.722.1,
which doubles the algorithm to permit 14-kHz audio bandwidth using a 32-kHz audio sample rate, at 24, 32,
and 48 kbit/s.
Annex C of [7]: this annex provides a description of the 14-kHz mode at 24, 32 and 48 kbit/s for this
Recommendation.
• Recommendation ITU-T G.729.1 [8] annex E (extension SWB for G.729.1 [8]).
This annex provides the high-level description of the higher bit-rate extension of G.729 designed to
accommodate a wide range of input signals, such as speech, with background noise and even music.
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11 ETSI TS 102 924 V1.2.1 (2018-03)
• Recommendation ITU-T G.718 [9] annex B Super-wideband scalable (extension for
Recommendation ITU-T G.718 [9]). This annex describes a scalable super-wideband (SWB, 50 to 14 000 Hz)
speech and audio coding algorithm operating from 36 to 48 kbit/s and interoperable with
Recommendation ITU-T G.718 [9].
• Recommendation ITU-T G.711.1 [6] annex D defines the super-wideband extension
Annex F defines the Stereo embedded extension for Recommendation ITU-T G.711.1 [6].
"Annex F is intended as a stereo extension to the G.711.1 [6] wideband coding algorithm and its super-
wideband annex D. Compared to discrete two-channel (dual-mono) audio transmission, this stereo extension
G.711.1 [6] annex F saves valuable bandwidth for stereo transmission. It is specified to offer the stereo
capability while providing backward compatibility with the monaural core in an embedded scalable way. The
annex provides very good quality for stereo speech contents (clean speech and noisy speech with various
stereo sound pickup systems: binaural, MS, etc.), and for most of the conditions it provides significantly higher
quality than low bitrate dual-mono. For some music contents, e.g. highly reverberated and/or with diffuse
sound, the algorithm may have some performance limitations and may not perform as good as dual-mono
codecs, however it achieves the quality of state-of-the-art parametric stereo codecs".
• Recommendation ITU-T G.722 [19] annex B defines the super-wideband extension
and annex D defines the Stereo embedded extension for Recommendation ITU-T G.722 [19].
"Annex B describes a scalable super-wideband (SWB, 50 to 14 000 Hz) speech and audio coding algorithm
operating at 64, 80 and 96 kbit/s. The Recommendation ITU-T G.722 [19] super-wideband extension codec is
interoperable with Recommendation ITU-T G.722 [19]. The output of the Recommendation ITU-T G.722 [19]
SWB coder has a bandwidth of 50 to 14 000 Hz".
"Annex D describes a stereo extension of the wideband codec G.722 and its super-wideband extension, G.722
annex B. It is optimized for the transmission of stereo signals with limited additional bitrate, while keeping full
compatibility with both codecs. Annex D operates from 64 to 128 kbit/s with four super-wideband stereo
bitrates at 80, 96, 112 and 128 kbit/s and two wideband stereo bitrates at 64 and 80 kbit/s".
• 3GPP VoLTE (IMS) ETSI TS 126 441 [26]. The Enhanced Voice Services coder consists of the multi-rate
audio coder optimized for operation with voice and music/mixed content signals, a source controlled rate
scheme including a voice/sound activity detector and a comfort noise generation system, and an error
concealment mechanism to combat the effects of transmission errors and lost packets.
Coder EVS (Enhanced Voice Services) is defined in ETSI TS 126 441 [26] and ETSI TR 126 952 [25]. The
tests conducted on codec implementations, e.g. [i.6] show that the requirements and test methods for SWB
terminals as defined in the present document apply.
EVS is designed for packet-switched and circuit-switched networks/Mobile VoIP and VoLTE is a key target
application.
The key features of EVS are Super-wideband speech (32 kHz sampling) with improved speech quality and
improved music performance.
4.2.2 Fullband (FB)
The following codecs are recommended for fullband:
• Recommendation ITU-T G.719 [10] Low-complexity, fullband audio coding for high-quality,
conversational applications.
"Recommendation ITU-T G.719 [10] describes the G.719 [10] coding algorithm for low-complexity fullband
conversational speech and audio, operating from 32 kbit/s up to 128 kbit/s".
The encoder input and decoder output are sampled at 48 kHz. The codec enables full bandwidth, from 20 Hz
to 20 kHz, encoding of speech, music and general audio content. The codec operates on 20-ms frames and has
an algorithmic delay of 40 ms.
NOTE: Amendment 1 adds new annex A that specifies the use of the ISO base media file format as container for
the G.719 [10] bitstream addresses non-conversational use cases of the codec (e.g. call waiting music
playback and recording of teleconferencing sessions, voice mail messages, online "jam"-sessions).
• 3GPP VoLTE (IMS) ETSI TS 126 441 [26]. The Enhanced Voice Services coder consists of the multi-rate
audio coder optimized for operation with voice and music/mixed content signals, a source controlled rate
scheme including a voice/sound activity detector and a comfort noise generation system, and an error
concealment mechanism to combat the effects of transmission errors and lost packets.
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12 ETSI TS 102 924 V1.2.1 (2018-03)
Coder EVS (Enhanced Voice Services) is defined in ETSI TS 126 441 [26] and ETSI TR 126 952 [25]. The
tests conducted on codec implementations, e.g. [i.6] show that the requirements and test methods for FB
terminals as defined in the present document apply.
EVS is designed for packet-switched and circuit-switched networks/Mobile VoIP and VoLTE is a key target
application. The key features of EVS are Fullband speech with improved speech quality and improved music
performance.
5 Test considerations and test equipment
5.0 Introduction
The terminals within the scope of the present document are not only dedicated to speech communication but are also
mixing speech and audio contents and may implement stereo and multichannel transmissions. As a consequence there is
a need to define new parameters, such as:
• Loudness: Loudness Rating is determined only for speech or speech-like signals. Loudness may be calculated
over any types of signals (audio sequences, speech sequences and mix of these sequences). Moreover it is not
intended to define Loudness Rating algorithms for Super-wideband and fullband speech. To be consistent with
transmission planning, the loudness rating shall be determined using wideband calculation and loudness shall
be measurement for all the bandwidths. Clause 5.4.1.2 details the measurement principles.
• Binaural listening: The most of the test assessment methods and requirements for speech terminals are based
on monaural listening, Even if some of them (e.g. for Handsfree Loudness rating) are intended to take into
account binaural listening, the basic methods and requirements are only taking into account correction factors.
The plan is to adapt test methods to effective binaural listening.
As a consequence, the present document takes into account test arrangements that are defined for speech terminals or
for audio equipment.
Recommendation ITU-T P.58 [3] give information about use of HATS only from 100 Hz to 10 kHz, but new designs
offer wider bandwidths.
For send the HATS can be used between 50 Hz and 16 kHz. Until the development of new systems with larger
bandwidth, send measurement will be limited to those frequencies.
NOTE 1: With some measurement equipment the use of such of bandwidth is not possible and should be limited to
100 Hz to 14 kHz.
For receive, a correction factor (given, in annex B) allows measurement at DRP until 16 kHz.
NOTE 2: It is not the intention of the present document to define new requirements to adapt HATS for super-
wideband and fullband. However when terminals implement Super-wideband or Fullband within
terminals support also WideBand and/or NarrowBand speech, it is intended to use as far as possible test
methods defined for wideband terminals and consequently to use HATS for parameters measured in
wideband bandwidth.
5.1 IP half channel measurement adaptor
The IP half channel measurement adaptor is described in ETSI EG 202 425 [i.5].
5.2 Environmental conditions for tests
The following conditions shall apply for the testing environment:
a) ambient temperature: 15 °C to 35 °C (inclusive);
b) relative humidity: 5 % to 85 %;
ETSI
13 ETSI TS 102 924 V1.2.1 (2018-03)
c) air pressure: 86 kPa to 106 kPa (860 mbar to 1 060 mbar).
5.3 Accuracy of measurements and test signal generation
Unless specified otherwise, the accuracy of measurements made by test equipment shall be equal to or better than:
Table 1: Measurement accuracy
Item Accuracy
Electrical signal level ±0,2 dB for levels ≥ -50 dBV
±0,4 dB for levels < -50 dBV
Sound pressure ±0,7 dB
Frequency ±0,2 %
Time ±0,2 %
Application force ±2 N
Measured maximum frequency 20 kHz

NOTE: The measured maximum frequency is due to Recommendation ITU-T P.58 limitations [3].
Unless specified otherwise, the accuracy of the signals generated by the test equipment shall be better than:
Table 2: Accuracy of test signal generation
Quantity Accuracy
Sound pressure level +2/-6 dB for frequencies from 50 Hz to100 Hz±2 dB for 100 Hz
(see note 2)
±1 dB for frequencies from 200 Hz to 8 000 Hz
±3 dB for frequencies from 8 000 Hz to 16 000 Hz
Electrical excitation levels ±0,4 dB across the whole frequency range
Frequency generation ±2 %
Time ±0,2 %
Specified component values ±1 %
NOTE 1: This tolerance may be used to avoid measurements at critical frequencies, e.g. those
due to sampling operations within the terminal under test.
NOTE 2: The limits for intermediate frequencies lie on a straight line drawn between the given
values on a linear (dB) - logarithmic (Hz) scale.

NOTE: With some measurement equipment the use of such of bandwidth is not possible and should be limited to
100 Hz to 14 kHz.
For terminal equipment which is directly powered from the mains supply, all tests shall be carried out within ±5 % of
the rated voltage of that supply. If the equipment is powered by other means and those means are not supplied as part of
the apparatus, all tests shall be carried out within the power supply limit declared by the supplier. If the power supply is
a.c., the test shall be conducted within ±4 % of the rated frequency.
5.4 Network impairment simulation
At least one set of requirements is based on the assumption of an error free packet network, and at least one other set of
requirements is based on a defined simulated malperformance of the packet network.
TM
An appropriate network simulator has to be used, for example NIST Net [i.2] (https://www-
x.antd.nist.gov/itg/nistnet/) or Netem [i.3].
TM
Based on the positive experience STQ have made during the ETSI Speech Quality Test Events with "NIST Net " this
will be taken as a basis to express and describe the variations of packet network parameters for the appropriate tests.
ETSI
14 ETSI TS 102 924 V1.2.1 (2018-03)
TM
Here is a brief blurb about NIST Net :
TM
• The NIST Net network emulator is a general-purpose tool for emulating performance dynamics in IP
networks. The tool is designed to allow controlled, reproducible experiments with network performance
sensitive/adaptive applications and control protocols in a simple laboratory setting. By operating at the IP
level, NIST Net can emulate the critical end-to-end performance characteristics imposed by various wide area
network situations (e.g. congestion loss) or by various underlying subnetwork technologies (e.g. asymmetric
bandwidth situations of xDSL and cable modems).
TM
• NIST Net is implemented as a kernel module extension to the Linux™ operating system and an X Window
System-based user interface application. In use, the tool allows an inexpensive PC-based router to emulate
numerous complex performance scenarios, including:
...