SIST ES 202 740 V1.3.2:2016
(Main)Speech and multimedia Transmission Quality (STQ) - Transmission requirements for wideband VoIP loudspeaking and handsfree terminals from a QoS perspective as perceived by the user
Speech and multimedia Transmission Quality (STQ) - Transmission requirements for wideband VoIP loudspeaking and handsfree terminals from a QoS perspective as perceived by the user
The present document provides speech transmission performance requirements for 8 kHz wideband VoIP loudspeaking
and hands-free terminals; it addresses all types of IP based terminals, including wireless, softphones and group audio
terminals.
In contrast to other standards which define minimum performance requirements it is the intention of the present
document to specify terminal equipment requirements which enable manufacturers and service providers to enable good
quality end-to-end speech performance as perceived by the user.
In addition to basic testing procedures, the present document describes advanced testing procedures taking into account
further quality parameters as perceived by the user.
NOTE: The present document does not concern headset terminals.
Kakovost prenosa govora in večpredstavnih vsebin (STQ) - Prenosne zahteve za širokopasovne zvočniške in prostoročne terminale VoIP glede na kakovost storitev (QoS), kot jih dojema uporabnik
V tem dokumentu so podane zahteve glede učinkovitosti prenosa govora za 8 kHz širokopasovne zvočniške in prostoročne terminale VoIP; obravnava vse vrste terminalov na podlagi IP, vključno z brezžičnimi terminali, programskimi telefoni in terminali za skupinske zvočne klice.
V nasprotju z ostalimi standardi, ki opredeljujejo minimalne zahteve glede učinkovitosti, je namen tega dokumenta določiti zahteve za terminalsko opremo, ki proizvajalcem in ponudnikom storitev omogočajo, da zagotavljajo dobro kakovost govora od začetka do konca, kot jo dojema uporabnik.
Poleg osnovnih preskusnih postopkov ta dokument opisuje napredne preskusne postopke, ki upoštevajo še druge parametre kakovosti, kot jih dojema uporabnik.
OPOMBA: Ta dokument ne zadeva naglavnih terminalov.
General Information
Standards Content (Sample)
Final draft ETSI ES 202 740 V1.3.2 (2010-07)
ETSI Standard
Speech and multimedia Transmission Quality (STQ);
Transmission requirements for
wideband VoIP loudspeaking and
handsfree terminals from a
QoS perspective as perceived by the user
2 Final draft ETSI ES 202 740 V1.3.2 (2010-07)
Reference
RES/STQ-00165
Keywords
terminal, handsfree, loudspeaking, VoIP, quality
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3 Final draft ETSI ES 202 740 V1.3.2 (2010-07)
Contents
Intellectual Property Rights . 6
Foreword . 6
Introduction . 6
1 Scope . 7
2 References . 7
2.1 Normative references . 7
2.2 Informative references . 8
3 Definitions and abbreviations . 9
3.1 Definitions . 9
3.2 Abbreviations . 9
4 General considerations . 10
4.1 Coding Algorithm. 10
4.2 End-to-end considerations . 10
4.3 Parameters to be investigated . 11
4.3.1 Basic parameters . 11
4.3.2 Further Parameters with respect to Speech Processing Devices . 11
5 Test equipment . 11
5.1 IP half channel measurement adaptor . 11
5.2 Environmental conditions for tests . 11
5.3 Accuracy of measurements and test signal generation . 12
5.4 Network impairment simulation . 12
5.5 Acoustic environment . 13
5.6 Influence of terminal delay on measurements . 14
6 Test Setup . 14
6.1 Setup for terminals . 15
6.1.1 Hands-free measurements . 15
6.1.2 Measurements in loudspeaking mode . 20
6.2 Test signal levels . 20
6.2.1 Send . 20
6.2.2 Receive . 21
6.3 Setup of background noise simulation. 21
7 Measurements and Requirements for Basic Parameters . 22
7.1 Coding independent parameters . 22
7.1.1 Send sensitivity/frequency response . 22
7.1.1.1 Requirement . 22
7.1.1.2 Measurement method . 23
7.1.2 Send loudness rating . 23
7.1.2.1 Requirement . 23
7.1.2.2 Measurement method . 24
7.1.3 Send distortion . 24
7.1.3.1 Requirement . 24
7.1.3.2 Measurement method . 24
7.1.4 Out-of-band signals in send direction (informative) . 24
7.1.4.1 Requirement . 24
7.1.4.2 Measurement method . 25
7.1.5 Send noise . 25
7.1.5.1 Requirement . 25
7.1.5.2 Measurement method . 25
7.1.6 Receive Frequency Response . 25
7.1.6.1 Requirement . 25
7.1.6.2 Measurement method . 27
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4 Final draft ETSI ES 202 740 V1.3.2 (2010-07)
7.1.7 Receive Loudness Rating . 28
7.1.7.1 Requirement . 28
7.1.7.2 Measurement method . 28
7.1.8 Receive distortion . 29
7.1.8.1 Requirement . 29
7.1.8.2 Measurement method . 29
7.1.9 Out-of-band signals in receive direction (informative) . 30
7.1.9.1 Requirement . 30
7.1.9.2 Measurement Method. 30
7.1.10 Receive noise . 30
7.1.10.1 Requirement . 30
7.1.10.2 Measurement method . 30
7.1.11 Terminal Coupling Loss . 31
7.1.11.1 Requirement . 31
7.1.11.2 Measurement method . 31
7.1.12 Stability Loss . 31
7.1.12.1 Requirement . 31
7.1.12.2 Measurement method . 32
7.2 Codec Specific Requirements. 32
7.2.1 Send Delay . 32
7.2.1.1 Requirement . 33
7.2.1.2 Measurement Method. 33
7.2.2 Receive delay . 34
7.2.2.1 Requirements . 34
7.2.2.2 Measurement Method. 35
8 Measurements and Requirements for Parameters with respect to Speech Processing Devices . 35
8.1 Objective Listening Speech Quality MOS-LQOM in Send direction . 35
8.2 Objective Listening Quality MOS-LQOM in Receive direction . 35
8.3 Minimum activation level and sensitivity in Receive direction . 35
8.4 Automatic Level Control in Receive . 35
8.5 Double Talk Performance. 35
8.5.1 Attenuation Range in Send Direction during Double Talk A . 36
H,S,dt
8.5.1.1 Requirement . 36
8.5.1.2 Measurement Method. 36
8.5.2 Attenuation Range in Receive Direction during Double Talk A . 38
H,R,dt
8.5.2.1 Requirement . 38
8.5.2.2 Measurement Method. 38
8.5.3 Detection of Echo Components during Double Talk . 38
8.5.3.1 Requirement . 38
8.5.3.2 Measurement Method. 39
8.5.4 Minimum activation level and sensitivity of double talk detection . 40
8.5.5 Switching characteristics . 40
8.5.5.1 Activation in Send Direction . 41
8.5.5.1.1 Requirements . 41
8.5.5.1.2 Measurement Method . 41
8.5.5.2 Silence Suppression and Comfort Noise Generation . 42
8.5.5.3 Performance in send direction in the presence of background noise . 42
8.5.5.3.1 Requirement . 42
8.5.5.3.2 Measurement Method . 42
8.5.5.4 Speech Quality in the Presence of Background Noise . 43
8.5.5.4.1 Requirement . 43
8.5.5.4.2 Measurement Method . 43
8.5.5.5 Quality of Background Noise Transmission (with Far End Speech) . 43
8.5.5.5.1 Requirements . 43
8.5.5.5.2 Measurement Method . 44
8.5.5.6 Quality of Background Noise Transmission (with Near End Speech) . 44
8.5.5.6.1 Requirements . 44
8.5.5.6.2 Measurement Method . 44
8.5.6 Quality of echo cancellation . 44
8.5.6.1 Temporal echo effects . 44
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8.5.6.1.1 Requirements . 44
8.5.6.1.2 Measurement Method . 45
8.5.6.2 Spectral Echo Attenuation . 45
8.5.6.2.1 Requirements . 45
8.5.6.2.2 Measurement Method . 45
8.5.6.3 Occurrence of Artefacts . 45
8.5.7 Variant Impairments; Network dependant . 46
8.5.7.1 Send and receive delay - Round trip delay . 46
8.5.7.2 Delay versus Time Send . 47
8.5.7.3 Delay versus Time Receive . 47
8.5.7.4 Quality of Jitter buffer adjustment . 47
Annex A (informative): Processing delays in VoIP terminals . 48
Annex B (informative): Bibliography . 51
History . 52
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6 Final draft ETSI ES 202 740 V1.3.2 (2010-07)
Intellectual Property Rights
IPRs essential or potentially essential to the present document 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 (http://webapp.etsi.org/IPR/home.asp).
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.
Foreword
This ETSI Standard (ES) has been produced by ETSI Technical Committee Speech and multimedia Transmission
Quality (STQ), and is now submitted for the ETSI standards Membership Approval Procedure.
Introduction
Traditionally, the analogue and digital telephones were interfacing switched-circuit 64 kbit/s PCM networks. With the
fast growth of IP networks, wideband terminals providing higher audio-bandwidth and directly interfacing
packet-switched networks (VoIP) are being rapidly introduced. Such IP network edge devices may include gateways,
specifically designed IP phones, soft phones or other devices connected to the IP based networks and providing
telephony service. Since the IP networks will be in many cases interworking with the traditional PSTN and private
networks, many of the basic transmission requirements have to be harmonized with specifications for traditional digital
terminals. However, due to the unique characteristics of the IP networks including packet loss, delay, etc. new
performance specification, as well as appropriate measuring methods, will have to be developed. Terminals are getting
increasingly complex, advanced signal processing is used to address the IP specific issues.
NOTE: Requirement limits are given in tables, the associated curve when provided is given for illustration.
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7 Final draft ETSI ES 202 740 V1.3.2 (2010-07)
1 Scope
The present document provides speech transmission performance requirements for 8 kHz wideband VoIP loudspeaking
and hands-free terminals; it addresses all types of IP based terminals, including wireless, softphones and group audio
terminals.
In contrast to other standards which define minimum performance requirements it is the intention of the present
document to specify terminal equipment requirements which enable manufacturers and service providers to enable good
quality end-to-end speech performance as perceived by the user.
In addition to basic testing procedures, the present document describes advanced testing procedures taking into account
further quality parameters as perceived by the user.
NOTE: The present document does not concern headset terminals.
2 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
reference document (including any amendments) applies.
Referenced documents which are not found to be publicly available in the expected location might be found at
http://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.
2.1 Normative references
The following referenced documents are necessary for the application of the present document.
[1] ETSI I-ETS 300 245-6: "Integrated Services Digital Network (ISDN); Technical characteristics of
telephony terminals; Part 6: Wideband (7 kHz), loudspeaking and hands free telephony".
[2] ETSI TS 126 171: "Digital cellular telecommunications system (Phase 2+); Universal Mobile
Telecommunications System (UMTS); AMR speech codec, wideband; General description
(3GPP TS 26.171 version 6.0.0 Release 6)".
[3] ITU-T Recommendation G.108: "Application of the E-model: A planning guide".
[4] ITU-T Recommendation G.109: "Definition of categories of speech transmission quality".
[5] ITU-T Recommendation G.122: "Influence of national systems on stability and talker echo in
international connections".
[6] ITU-T Recommendation G.131: "Talker echo and its control".
[7] ITU-T Recommendation G.711: "Pulse code modulation (PCM) of voice frequencies".
[8] ITU-T Recommendation G.722: "7 kHz audio-coding within 64 kbit/s".
[9] ITU-T Recommendation G.722.1: "Low-complexity coding at 24 and 32 kbit/s for hands-free
operation in systems with low frame loss".
[10] ITU-T Recommendation G.729.1: "G.729 based Embedded Variable bit-rate coder: An 8-32 kbit/s
scalable wideband coder bitstream interoperable with G.729".
[11] ITU-T Recommendation G.1020: "Performance parameter definitions for quality of speech and
other voiceband applications utilizing IP networks".
[12] ITU-T Recommendation P.50: "Artificial voices".
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8 Final draft ETSI ES 202 740 V1.3.2 (2010-07)
[13] ITU-T Recommendation P.56: "Objective measurement of active speech level".
[14] ITU-T Recommendation P.58: "Head and torso simulator for telephonometry".
[15] ITU-T Recommendation P.79: "Calculation of loudness ratings for telephone sets".
[16] ITU-T Recommendation P.310: "Transmission characteristics for telephone band (300-3400 Hz)
digital telephones".
[17] ITU-T Recommendation P.340: "Transmission characteristics and speech quality parameters of
hands-free terminals".
[18] ITU-T Recommendation P.341: "Transmission characteristics for wideband (150-7000 Hz) digital
hands-free telephony terminals".
[19] ITU-T Recommendation P.501: "Test signals for use in telephonometry".
[20] ITU-T Recommendation P.502: "Objective test methods for speech communication systems using
complex test signals".
[21] ITU-T Recommendation P.581: "Use of head and torso simulator (HATS) for hands-free terminal
testing".
[22] ITU-T Recommendation P.862: "Perceptual evaluation of speech quality (PESQ): An objective
method for end-to-end speech quality assessment of narrow-band telephone networks and speech
codecs".
[23] ISO 3 (1973): "Preferred numbers - Series of preferred numbers".
[24] ITU-T Recommendation P.800.1: "Mean Opinion Score (MOS) terminology".
2.2 Informative references
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] ETSI EG 202 396-1: "Speech Processing, Transmission and Quality Aspects (STQ); Speech
quality performance in the presence of background noise; Part 1: Background noise simulation
technique and background noise database".
[i.2] ETSI EG 202 425: "Speech Processing, Transmission and Quality Aspects (STQ); Definition and
implementation of VoIP reference point".
[i.3] ETSI EG 202 396-3: "Speech Processing, Transmission and Quality Aspects (STQ); Speech
quality performance in the presence of background noise; Part 3: Background noise transmission -
objective model".
[i.4] ETSI TR 102 648-1: "Speech Processing, Transmission and Quality Aspects (STQ); Test
Methodologies for ETSI Test Events and Results; Part 1: VoIP Speech Quality Testing".
[i.5] NIST net.
NOTE: Available at http://snad.ncsl.nist.gov/itg/nistnet/.
[i.6] Netem.
NOTE: Available at http://www.linuxfoundation.org/en/Net:Netem.
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9 Final draft ETSI ES 202 740 V1.3.2 (2010-07)
3 Definitions and abbreviations
3.1 Definitions
For the purposes of the present document, the following terms and definitions apply:
artificial ear: device for the calibration of earphones incorporating an acoustic coupler and a calibrated microphone for
the measurement of the sound pressure and having an overall acoustic impedance similar to that of the median adult
human ear over a given frequency band
codec: combination of an analogue-to-digital encoder and a digital-to-analogue decoder operating in opposite directions
of transmission in the same equipment
ear-Drum Reference Point (DRP): point located at the end of the ear canal, corresponding to the ear-drum position
freefield equalization: artificial head is equalized in such a way that for frontal sound incidence in anechoic conditions
the frequency response of the artificial head is flat
freefield reference point: point located in the free sound field, at least in 1,5 m distance from a sound source radiating
in free air
NOTE: In case of a head and torso simulator (HATS) in the centre of the artificial head with no artificial head
present.
group-audio terminal: handsfree terminal primarily designed for use by several users which will not be equipped with
a handset
handsfree telephony terminal: telephony terminal using a loudspeaker associated with an amplifier as a telephone
receiver and which can be used without a handset
HATS Hands-Free Reference Point (HATS HFRP): corresponds to a reference point "n" from ITU-T
Recommendation P.58 [14] "n" is one of the points numbered from 11 to 17 and defined in table 6a of ITU-T
Recommendation P.58 [14] (coordinates of far field front point)
NOTE: The HATS HFRP depends on the location(s) of the microphones of the terminal under test: the
appropriate axis lip-ring/HATS HFRP is to be as close as possible to the axis lip-ring/HFT microphone
under test.
Head And Torso Simulator (HATS) for telephonometry: manikin extending downward from the top of the head to
the waist, designed to simulate the sound pick-up characteristics and the acoustic diffraction produced by a median
human adult and to reproduce the acoustic field generated by the human mouth
loudspeaking function: function of a handset telephone using a loudspeaker associated with an amplifier as a
telephone receiver
Mouth Reference Point (MRP): is located on axis and 25 mm in front of the lip plane of a mouth simulator
nominal setting of the volume control: setting which is closest to the nominal RLR
softphone: speech communication system based upon a computer
3.2 Abbreviations
For the purposes of the present document, the following abbreviations apply:
CSS Composite Source Signal
DRP ear Drum Reference Point
EL Echo Loss
ERP Ear Reference Point
HATS Head And Torso Simulator
HFRP Hands Free Reference Point
L Earphone coupling Loss
E
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10 Final draft ETSI ES 202 740 V1.3.2 (2010-07)
MOS-LQOy Mean Opinion Score - Listening Quality Objective, y being n for narrow-band, w for wideband,
and M for mixed
NOTE: See ITU-T Recommendation P.800.1 [24].
MRP Mouth Reference Point
NLP Non Linear Processor
PCM Pulse Code Modulation
PLC Packet Loss Concealment
POI Point Of Interconnection
PSTN Public Switched Telephone Network
QoS Quality of Service
RLR Receive Loudness Rating
RLRmax Receive Loudness Rating corresponding to the maximum setting of the volume control
RLRmin Receive Loudness Rating corresponding to the minimum setting of the volume control
SLR Send Loudness Rating
TCLw Terminal Coupling Loss (weighted)
TCN Trace Control for Netem
TELR Talker Echo Loudness Rating
VoIP Voice over Internet Protocol
4 General considerations
4.1 Coding Algorithm
The assumed coding algorithm is according to ITU-T Recommendation G.722 [8]. VoIP terminals may support other
coding algorithms.
NOTE: Associated Packet Loss Concealment, e.g. as defined in ITU-T Recommendation G.722 [8], appendixes 3
and 4, should be used.
4.2 End-to-end considerations
In order to achieve a desired end-to-end speech transmission performance (mouth-to-ear) it is recommended that
general rules of transmission planning tasks are carried out with the E-model taking into account that E-model does not
directly address handsfree or loudspeaking terminals; this includes the a-priori determination of the desired category of
speech transmission quality as defined in ITU-T Recommendation G.109 [4].
While, in general, the transmission characteristics of single circuit-oriented network elements, such as switches or
terminals can be assumed to have a single input value for the planning tasks of ITU-T Recommendation G.108 [3], this
approach is not applicable in packet based systems and thus there is a need for the transmission planner's specific
attention.
In particular the decision as to which delay measured according to the present document should is acceptable or
representative for the specific configuration is the responsibility of the individual transmission planner.
ITU-T Recommendation G.108 [3] with its amendments provides further guidance on this important issue.
The following optimum terminal parameters from a users' perspective need to be considered:
• Minimized delay in send and receive direction.
• Optimum loudness Rating (RLR, SLR).
• Compensation for network delay variation.
• Packet loss recovery performance.
• Maximized terminal coupling loss.
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11 Final draft ETSI ES 202 740 V1.3.2 (2010-07)
• Some more basic (I-ETS 300 245-6 [1]) parameters are applicable, if ITU-T Recommendation G.722 [8] is
used.
4.3 Parameters to be investigated
4.3.1 Basic parameters
The basic parameters are given in I-ETS 300 245-6 [1], ITU-T Recommendation P.340 [17] and ITU-T
Recommendation P.341 [18].
4.3.2 Further Parameters with respect to Speech Processing Devices
For VoIP terminals that contain non-linear speech processing devices, the following parameters require additional
attention in the context of the present document.
The measurements for further parameters with respect to speech processing devices which are novelties to terminal
requirement standards, have been successfully used in the ETSI Speech Quality Test Events (see TR 102 648-1 [i.4]):
• Objective evaluation of speech quality for VoIP terminals.
• Minimum activation level and sensitivity in Receive direction.
• Automatic Level Control in Receive.
• Double Talk Performance.
• Minimum activation level and sensitivity of double talk detection.
• Switching characteristics.
• Quality of echo cancellation.
• Variant Impairments; Network dependant.
• Etc.
5 Test equipment
5.1 IP half channel measurement adaptor
The IP half channel measurement adaptor is described in EG 202 425 [i.2].
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 %;
c) air pressure: 86 kPa to 106 kPa (860 mbar to 1 060 mbar).
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12 Final draft ETSI ES 202 740 V1.3.2 (2010-07)
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 Newton
Measured maximum frequency 10 kHz
NOTE: The measured maximum frequency is due to P. 58 limitations [14].
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 at ±3 dB for frequencies from 100 Hz to 200 Hz
Mouth Reference Point (MRP) ±1 dB for frequencies from 200 Hz to 4 000 Hz
±3 dB for frequencies from 4 000 Hz to 8 000 Hz
Electrical excitation levels ±0,4 dB across the whole frequency range
Frequency generation ±2 % (see note)
Time ±0,2 %
Specified component values ±1 %
NOTE: This tolerance may be used to avoid measurements at critical frequencies, e.g. those
due to sampling operations within the terminal under test.
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
alternate current, 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 loss of performance of the packet network.
An appropriate network simulator has to be used, for example NISTnet [i.5] (http://snad.ncsl.nist.gov/itg/nistnet/) or
Netem [i.6].
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.
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13 Final draft ETSI ES 202 740 V1.3.2 (2010-07)
Here is a brief blurb about NIST Net:
• 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).
• 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: tuneable packet delay distributions, congestion and
background loss, bandwidth limitation, and packet reordering/duplication. The X interface allows the user to
select and monitor specific traffic streams passing through the router and to apply selected performance
"effects" to the IP packets of the stream. In addition to the interactive interface, NIST Net can be driven by
traces produced from measurements of actual network conditions. NIST Net also provides support for user
defined packet handlers to be added to the system. Examples of the use of such packet handlers include: time
stamping/data collection, interception and diversion of selected flows, generation of protocol responses from
emulated clients.
The key points of Netem can be summarized as follows:
• Netem is nowadays part of most Linux distributions, it only has to be switched on, when compiling a kernel.
With Netem, there are the same possibilities as with nistnet, there can be generated loss, duplication, delay and
jitter (and the distribution can be chosen during runtime). Netem can be run on a Linux-PC running as a bridge
or a router (Nistnet only runs on routers).
• With an amendment of Netem, TCN (Trace Control for Netem) which was developed by ETH Zurich, it is
even possible, to control the behaviour of single packets via a trace file. So it is for example possible to
generate a single packet loss, or a specific delay pattern. This amendment is planned to be included in new
Linux kernels, nowadays it is available as a patch to a specific kernel and to the iproute2 tool (iproute2
contains Netem).
• It is not advised to define specific distortion patterns for testing in standards, because it will be easy to adapt
devices to these patterns (as it is already done for test signals). But if a pattern is unknown to a manufacturer,
the same pattern can be used by a test lab for different devices and gives comparable results. It is also possible
to take a trace of Nistnet distortions, generate a file out of this and playback exactly the same distortions with
Netem.
5.5 Acoustic environment
In general two possible approaches need to be taken into account: either room noise and background noise are an
inherent part of the test environment or room noise and background noise shall be eliminated to such an extent that their
influence on the test results can be neglected.
Unless stated otherwise measurements shall be conducted under quiet and "anechoic" conditions. Depending on the
distance of the transducers from mouth and ear a quiet office room may be sufficient e.g. for handsets where artificial
mouth and artificial ear are located close to the acoustical transducers. But this is not applicable for handsfree and
loudspeaking terminals.
In cases where real or simulated background noise is used as part of the testing environment, the original background
noise must not be noticeably influenced by the acoustical properties of the room.
In all cases where the performance of acoustic echo cancellers shall be tested, a realistic room, which represents the
typical user environment for the terminal shall be used.
In case where an anechoic room is not available the test room has to be an acoustically treated room with few
reflections and a low noise level.
Considering this, test laboratory, in the case where its test room does not conform to anechoic conditions as given in
ITU-T Recommendation P.341 [18], has to present difference in results for measurements due to its test room.
ETSI
14 Final draft ETSI ES 202 740 V1.3.2 (2010-07)
5.6 Influence of terminal delay on measurements
As delay is introduced by the terminal, care shall be taken for all measurements where exact position of the analysis
window is required. It shall be checked that the test is performed on the test signal and not on any other signal.
6 Test Setup
In order to use a compatible test system for all types of speech terminals a HATS (Head And Torso Simulator) will be
used instead of free field microphone (for receive measurement) and artificial mouth (for send measurement). HATS is
described in ITU-T Recommendation P.58 [14].
The preferred way of testing a terminal is to connect it to a network simulator with exact defined settings and access
points. The test sequences are fed in either electricall
...
ETSI Standard
Speech and multimedia Transmission Quality (STQ);
Transmission requirements for wideband
VoIP loudspeaking and handsfree terminals
from a QoS perspective as perceived by the user
2 ETSI ES 202 740 V1.3.2 (2010-09)
Reference
RES/STQ-00165
Keywords
terminal, handsfree, loudspeaking, VoIP, quality
ETSI
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ETSI
3 ETSI ES 202 740 V1.3.2 (2010-09)
Contents
Intellectual Property Rights . 6
Foreword . 6
Introduction . 6
1 Scope . 7
2 References . 7
2.1 Normative references . 7
2.2 Informative references . 8
3 Definitions and abbreviations . 9
3.1 Definitions . 9
3.2 Abbreviations . 9
4 General considerations . 10
4.1 Coding Algorithm. 10
4.2 End-to-end considerations . 10
4.3 Parameters to be investigated . 11
4.3.1 Basic parameters . 11
4.3.2 Further Parameters with respect to Speech Processing Devices . 11
5 Test equipment . 11
5.1 IP half channel measurement adaptor . 11
5.2 Environmental conditions for tests . 11
5.3 Accuracy of measurements and test signal generation . 12
5.4 Network impairment simulation . 12
5.5 Acoustic environment . 13
5.6 Influence of terminal delay on measurements . 13
6 Test Setup . 14
6.1 Setup for terminals . 14
6.1.1 Hands-free measurements . 14
6.1.2 Measurements in loudspeaking mode . 20
6.2 Test signal levels . 20
6.2.1 Send . 20
6.2.2 Receive . 21
6.3 Setup of background noise simulation. 21
7 Measurements and Requirements for Basic Parameters . 22
7.1 Coding independent parameters . 22
7.1.1 Send sensitivity/frequency response . 22
7.1.1.1 Requirement . 22
7.1.1.2 Measurement method . 23
7.1.2 Send loudness rating . 23
7.1.2.1 Requirement . 23
7.1.2.2 Measurement method . 24
7.1.3 Send distortion . 24
7.1.3.1 Requirement . 24
7.1.3.2 Measurement method . 24
7.1.4 Out-of-band signals in send direction (informative) . 24
7.1.4.1 Requirement . 24
7.1.4.2 Measurement method . 25
7.1.5 Send noise . 25
7.1.5.1 Requirement . 25
7.1.5.2 Measurement method . 25
7.1.6 Receive Frequency Response . 25
7.1.6.1 Requirement . 25
7.1.6.2 Measurement method . 27
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4 ETSI ES 202 740 V1.3.2 (2010-09)
7.1.7 Receive Loudness Rating . 28
7.1.7.1 Requirement . 28
7.1.7.2 Measurement method . 28
7.1.8 Receive distortion . 29
7.1.8.1 Requirement . 29
7.1.8.2 Measurement method . 29
7.1.9 Out-of-band signals in receive direction (informative) . 30
7.1.9.1 Requirement . 30
7.1.9.2 Measurement Method. 30
7.1.10 Receive noise . 30
7.1.10.1 Requirement . 30
7.1.10.2 Measurement method . 30
7.1.11 Terminal Coupling Loss . 31
7.1.11.1 Requirement . 31
7.1.11.2 Measurement method . 31
7.1.12 Stability Loss . 31
7.1.12.1 Requirement . 31
7.1.12.2 Measurement method . 32
7.2 Codec Specific Requirements. 32
7.2.1 Send Delay . 32
7.2.1.1 Requirement . 33
7.2.1.2 Measurement Method. 33
7.2.2 Receive delay . 33
7.2.2.1 Requirements . 34
7.2.2.2 Measurement Method. 34
8 Measurements and Requirements for Parameters with respect to Speech Processing Devices . 35
8.1 Objective Listening Speech Quality MOS-LQOM in Send direction . 35
8.2 Objective Listening Quality MOS-LQOM in Receive direction . 35
8.3 Minimum activation level and sensitivity in Receive direction . 35
8.4 Automatic Level Control in Receive . 35
8.5 Double Talk Performance. 35
8.5.1 Attenuation Range in Send Direction during Double Talk A . 36
H,S,dt
8.5.1.1 Requirement . 36
8.5.1.2 Measurement Method. 36
8.5.2 Attenuation Range in Receive Direction during Double Talk A . 37
H,R,dt
8.5.2.1 Requirement . 37
8.5.2.2 Measurement Method. 37
8.5.3 Detection of Echo Components during Double Talk . 38
8.5.3.1 Requirement . 38
8.5.3.2 Measurement Method. 38
8.5.4 Minimum activation level and sensitivity of double talk detection . 40
8.5.5 Switching characteristics . 40
8.5.5.1 Activation in Send Direction . 41
8.5.5.1.1 Requirements . 41
8.5.5.1.2 Measurement Method . 41
8.5.5.2 Silence Suppression and Comfort Noise Generation . 42
8.5.5.3 Performance in send direction in the presence of background noise . 42
8.5.5.3.1 Requirement . 42
8.5.5.3.2 Measurement Method . 42
8.5.5.4 Speech Quality in the Presence of Background Noise . 43
8.5.5.4.1 Requirement . 43
8.5.5.4.2 Measurement Method . 43
8.5.5.5 Quality of Background Noise Transmission (with Far End Speech) . 43
8.5.5.5.1 Requirements . 43
8.5.5.5.2 Measurement Method . 44
8.5.5.6 Quality of Background Noise Transmission (with Near End Speech) . 44
8.5.5.6.1 Requirements . 44
8.5.5.6.2 Measurement Method . 44
8.5.6 Quality of echo cancellation . 44
8.5.6.1 Temporal echo effects . 44
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5 ETSI ES 202 740 V1.3.2 (2010-09)
8.5.6.1.1 Requirements . 44
8.5.6.1.2 Measurement Method . 45
8.5.6.2 Spectral Echo Attenuation . 45
8.5.6.2.1 Requirements . 45
8.5.6.2.2 Measurement Method . 45
8.5.6.3 Occurrence of Artefacts . 45
8.5.7 Variant Impairments; Network dependant . 46
8.5.7.1 Send and receive delay - Round trip delay . 46
8.5.7.2 Delay versus Time Send . 47
8.5.7.3 Delay versus Time Receive . 47
8.5.7.4 Quality of Jitter buffer adjustment . 47
Annex A (informative): Processing delays in VoIP terminals . 48
Annex B (informative): Bibliography . 51
History . 52
ETSI
6 ETSI ES 202 740 V1.3.2 (2010-09)
Intellectual Property Rights
IPRs essential or potentially essential to the present document 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 (http://webapp.etsi.org/IPR/home.asp).
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.
Foreword
This ETSI Standard (ES) has been produced by ETSI Technical Committee Speech and multimedia Transmission
Quality (STQ).
Introduction
Traditionally, the analogue and digital telephones were interfacing switched-circuit 64 kbit/s PCM networks. With the
fast growth of IP networks, wideband terminals providing higher audio-bandwidth and directly interfacing
packet-switched networks (VoIP) are being rapidly introduced. Such IP network edge devices may include gateways,
specifically designed IP phones, soft phones or other devices connected to the IP based networks and providing
telephony service. Since the IP networks will be in many cases interworking with the traditional PSTN and private
networks, many of the basic transmission requirements have to be harmonized with specifications for traditional digital
terminals. However, due to the unique characteristics of the IP networks including packet loss, delay, etc. new
performance specification, as well as appropriate measuring methods, will have to be developed. Terminals are getting
increasingly complex, advanced signal processing is used to address the IP specific issues.
NOTE: Requirement limits are given in tables, the associated curve when provided is given for illustration.
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7 ETSI ES 202 740 V1.3.2 (2010-09)
1 Scope
The present document provides speech transmission performance requirements for 8 kHz wideband VoIP loudspeaking
and hands-free terminals; it addresses all types of IP based terminals, including wireless, softphones and group audio
terminals.
In contrast to other standards which define minimum performance requirements it is the intention of the present
document to specify terminal equipment requirements which enable manufacturers and service providers to enable good
quality end-to-end speech performance as perceived by the user.
In addition to basic testing procedures, the present document describes advanced testing procedures taking into account
further quality parameters as perceived by the user.
NOTE: The present document does not concern headset terminals.
2 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
reference document (including any amendments) applies.
Referenced documents which are not found to be publicly available in the expected location might be found at
http://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.
2.1 Normative references
The following referenced documents are necessary for the application of the present document.
[1] ETSI I-ETS 300 245-6: "Integrated Services Digital Network (ISDN); Technical characteristics of
telephony terminals; Part 6: Wideband (7 kHz), loudspeaking and hands free telephony".
[2] ETSI TS 126 171: "Digital cellular telecommunications system (Phase 2+); Universal Mobile
Telecommunications System (UMTS); AMR speech codec, wideband; General description
(3GPP TS 26.171 version 6.0.0 Release 6)".
[3] ITU-T Recommendation G.108: "Application of the E-model: A planning guide".
[4] ITU-T Recommendation G.109: "Definition of categories of speech transmission quality".
[5] ITU-T Recommendation G.122: "Influence of national systems on stability and talker echo in
international connections".
[6] ITU-T Recommendation G.131: "Talker echo and its control".
[7] ITU-T Recommendation G.711: "Pulse code modulation (PCM) of voice frequencies".
[8] ITU-T Recommendation G.722: "7 kHz audio-coding within 64 kbit/s".
[9] ITU-T Recommendation G.722.1: "Low-complexity coding at 24 and 32 kbit/s for hands-free
operation in systems with low frame loss".
[10] ITU-T Recommendation G.729.1: "G.729 based Embedded Variable bit-rate coder: An 8-32 kbit/s
scalable wideband coder bitstream interoperable with G.729".
[11] ITU-T Recommendation G.1020: "Performance parameter definitions for quality of speech and
other voiceband applications utilizing IP networks".
[12] ITU-T Recommendation P.50: "Artificial voices".
ETSI
8 ETSI ES 202 740 V1.3.2 (2010-09)
[13] ITU-T Recommendation P.56: "Objective measurement of active speech level".
[14] ITU-T Recommendation P.58: "Head and torso simulator for telephonometry".
[15] ITU-T Recommendation P.79: "Calculation of loudness ratings for telephone sets".
[16] ITU-T Recommendation P.310: "Transmission characteristics for telephone band (300-3400 Hz)
digital telephones".
[17] ITU-T Recommendation P.340: "Transmission characteristics and speech quality parameters of
hands-free terminals".
[18] ITU-T Recommendation P.341: "Transmission characteristics for wideband (150-7000 Hz) digital
hands-free telephony terminals".
[19] ITU-T Recommendation P.501: "Test signals for use in telephonometry".
[20] ITU-T Recommendation P.502: "Objective test methods for speech communication systems using
complex test signals".
[21] ITU-T Recommendation P.581: "Use of head and torso simulator (HATS) for hands-free terminal
testing".
[22] ITU-T Recommendation P.862: "Perceptual evaluation of speech quality (PESQ): An objective
method for end-to-end speech quality assessment of narrow-band telephone networks and speech
codecs".
[23] ISO 3 (1973): "Preferred numbers - Series of preferred numbers".
[24] ITU-T Recommendation P.800.1: "Mean Opinion Score (MOS) terminology".
2.2 Informative references
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] ETSI EG 202 396-1: "Speech Processing, Transmission and Quality Aspects (STQ); Speech
quality performance in the presence of background noise; Part 1: Background noise simulation
technique and background noise database".
[i.2] ETSI EG 202 425: "Speech Processing, Transmission and Quality Aspects (STQ); Definition and
implementation of VoIP reference point".
[i.3] ETSI EG 202 396-3: "Speech Processing, Transmission and Quality Aspects (STQ); Speech
quality performance in the presence of background noise; Part 3: Background noise transmission -
objective model".
[i.4] ETSI TR 102 648-1: "Speech Processing, Transmission and Quality Aspects (STQ); Test
Methodologies for ETSI Test Events and Results; Part 1: VoIP Speech Quality Testing".
[i.5] NIST net.
NOTE: Available at http://snad.ncsl.nist.gov/itg/nistnet/.
[i.6] Netem.
NOTE: Available at http://www.linuxfoundation.org/en/Net:Netem.
ETSI
9 ETSI ES 202 740 V1.3.2 (2010-09)
3 Definitions and abbreviations
3.1 Definitions
For the purposes of the present document, the following terms and definitions apply:
artificial ear: device for the calibration of earphones incorporating an acoustic coupler and a calibrated microphone for
the measurement of the sound pressure and having an overall acoustic impedance similar to that of the median adult
human ear over a given frequency band
codec: combination of an analogue-to-digital encoder and a digital-to-analogue decoder operating in opposite directions
of transmission in the same equipment
ear-Drum Reference Point (DRP): point located at the end of the ear canal, corresponding to the ear-drum position
freefield equalization: artificial head is equalized in such a way that for frontal sound incidence in anechoic conditions
the frequency response of the artificial head is flat
freefield reference point: point located in the free sound field, at least in 1,5 m distance from a sound source radiating
in free air
NOTE: In case of a head and torso simulator (HATS) in the centre of the artificial head with no artificial head
present.
group-audio terminal: handsfree terminal primarily designed for use by several users which will not be equipped with
a handset
handsfree telephony terminal: telephony terminal using a loudspeaker associated with an amplifier as a telephone
receiver and which can be used without a handset
HATS Hands-Free Reference Point (HATS HFRP): corresponds to a reference point "n" from ITU-T
Recommendation P.58 [14] "n" is one of the points numbered from 11 to 17 and defined in table 6a of ITU-T
Recommendation P.58 [14] (coordinates of far field front point)
NOTE: The HATS HFRP depends on the location(s) of the microphones of the terminal under test: the
appropriate axis lip-ring/HATS HFRP is to be as close as possible to the axis lip-ring/HFT microphone
under test.
Head And Torso Simulator (HATS) for telephonometry: manikin extending downward from the top of the head to
the waist, designed to simulate the sound pick-up characteristics and the acoustic diffraction produced by a median
human adult and to reproduce the acoustic field generated by the human mouth
loudspeaking function: function of a handset telephone using a loudspeaker associated with an amplifier as a
telephone receiver
Mouth Reference Point (MRP): is located on axis and 25 mm in front of the lip plane of a mouth simulator
nominal setting of the volume control: setting which is closest to the nominal RLR
softphone: speech communication system based upon a computer
3.2 Abbreviations
For the purposes of the present document, the following abbreviations apply:
CSS Composite Source Signal
DRP ear Drum Reference Point
EL Echo Loss
ERP Ear Reference Point
HATS Head And Torso Simulator
HFRP Hands Free Reference Point
L Earphone coupling Loss
E
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10 ETSI ES 202 740 V1.3.2 (2010-09)
MOS-LQOy Mean Opinion Score - Listening Quality Objective, y being n for narrow-band, w for wideband,
and M for mixed
NOTE: See ITU-T Recommendation P.800.1 [24].
MRP Mouth Reference Point
NLP Non Linear Processor
PCM Pulse Code Modulation
PLC Packet Loss Concealment
POI Point Of Interconnection
PSTN Public Switched Telephone Network
QoS Quality of Service
RLR Receive Loudness Rating
RLRmax Receive Loudness Rating corresponding to the maximum setting of the volume control
RLRmin Receive Loudness Rating corresponding to the minimum setting of the volume control
SLR Send Loudness Rating
TCLw Terminal Coupling Loss (weighted)
TCN Trace Control for Netem
TELR Talker Echo Loudness Rating
VoIP Voice over Internet Protocol
4 General considerations
4.1 Coding Algorithm
The assumed coding algorithm is according to ITU-T Recommendation G.722 [8]. VoIP terminals may support other
coding algorithms.
NOTE: Associated Packet Loss Concealment, e.g. as defined in ITU-T Recommendation G.722 [8], appendixes 3
and 4, should be used.
4.2 End-to-end considerations
In order to achieve a desired end-to-end speech transmission performance (mouth-to-ear) it is recommended that
general rules of transmission planning tasks are carried out with the E-model taking into account that E-model does not
directly address handsfree or loudspeaking terminals; this includes the a-priori determination of the desired category of
speech transmission quality as defined in ITU-T Recommendation G.109 [4].
While, in general, the transmission characteristics of single circuit-oriented network elements, such as switches or
terminals can be assumed to have a single input value for the planning tasks of ITU-T Recommendation G.108 [3], this
approach is not applicable in packet based systems and thus there is a need for the transmission planner's specific
attention.
In particular the decision as to which delay measured according to the present document should is acceptable or
representative for the specific configuration is the responsibility of the individual transmission planner.
ITU-T Recommendation G.108 [3] with its amendments provides further guidance on this important issue.
The following optimum terminal parameters from a users' perspective need to be considered:
• Minimized delay in send and receive direction.
• Optimum loudness Rating (RLR, SLR).
• Compensation for network delay variation.
• Packet loss recovery performance.
• Maximized terminal coupling loss.
ETSI
11 ETSI ES 202 740 V1.3.2 (2010-09)
• Some more basic (I-ETS 300 245-6 [1]) parameters are applicable, if ITU-T Recommendation G.722 [8] is
used.
4.3 Parameters to be investigated
4.3.1 Basic parameters
The basic parameters are given in I-ETS 300 245-6 [1], ITU-T Recommendation P.340 [17] and ITU-T
Recommendation P.341 [18].
4.3.2 Further Parameters with respect to Speech Processing Devices
For VoIP terminals that contain non-linear speech processing devices, the following parameters require additional
attention in the context of the present document.
The measurements for further parameters with respect to speech processing devices which are novelties to terminal
requirement standards, have been successfully used in the ETSI Speech Quality Test Events (see TR 102 648-1 [i.4]):
• Objective evaluation of speech quality for VoIP terminals.
• Minimum activation level and sensitivity in Receive direction.
• Automatic Level Control in Receive.
• Double Talk Performance.
• Minimum activation level and sensitivity of double talk detection.
• Switching characteristics.
• Quality of echo cancellation.
• Variant Impairments; Network dependant.
• Etc.
5 Test equipment
5.1 IP half channel measurement adaptor
The IP half channel measurement adaptor is described in EG 202 425 [i.2].
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 %;
c) air pressure: 86 kPa to 106 kPa (860 mbar to 1 060 mbar).
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12 ETSI ES 202 740 V1.3.2 (2010-09)
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 Newton
Measured maximum frequency 10 kHz
NOTE: The measured maximum frequency is due to P. 58 limitations [14].
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 at ±3 dB for frequencies from 100 Hz to 200 Hz
Mouth Reference Point (MRP) ±1 dB for frequencies from 200 Hz to 4 000 Hz
±3 dB for frequencies from 4 000 Hz to 8 000 Hz
Electrical excitation levels ±0,4 dB across the whole frequency range
Frequency generation ±2 % (see note)
Time ±0,2 %
Specified component values ±1 %
NOTE: This tolerance may be used to avoid measurements at critical frequencies, e.g. those
due to sampling operations within the terminal under test.
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
alternate current, 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 loss of performance of the packet network.
An appropriate network simulator has to be used, for example NISTnet [i.5] (http://snad.ncsl.nist.gov/itg/nistnet/) or
Netem [i.6].
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.
Here is a brief blurb about NIST Net:
• 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).
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13 ETSI ES 202 740 V1.3.2 (2010-09)
• 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: tuneable packet delay distributions, congestion and
background loss, bandwidth limitation, and packet reordering/duplication. The X interface allows the user to
select and monitor specific traffic streams passing through the router and to apply selected performance
"effects" to the IP packets of the stream. In addition to the interactive interface, NIST Net can be driven by
traces produced from measurements of actual network conditions. NIST Net also provides support for user
defined packet handlers to be added to the system. Examples of the use of such packet handlers include: time
stamping/data collection, interception and diversion of selected flows, generation of protocol responses from
emulated clients.
The key points of Netem can be summarized as follows:
• Netem is nowadays part of most Linux distributions, it only has to be switched on, when compiling a kernel.
With Netem, there are the same possibilities as with nistnet, there can be generated loss, duplication, delay and
jitter (and the distribution can be chosen during runtime). Netem can be run on a Linux-PC running as a bridge
or a router (Nistnet only runs on routers).
• With an amendment of Netem, TCN (Trace Control for Netem) which was developed by ETH Zurich, it is
even possible, to control the behaviour of single packets via a trace file. So it is for example possible to
generate a single packet loss, or a specific delay pattern. This amendment is planned to be included in new
Linux kernels, nowadays it is available as a patch to a specific kernel and to the iproute2 tool (iproute2
contains Netem).
• It is not advised to define specific distortion patterns for testing in standards, because it will be easy to adapt
devices to these patterns (as it is already done for test signals). But if a pattern is unknown to a manufacturer,
the same pattern can be used by a test lab for different devices and gives comparable results. It is also possible
to take a trace of Nistnet distortions, generate a file out of this and playback exactly the same distortions with
Netem.
5.5 Acoustic environment
In general two possible approaches need to be taken into account: either room noise and background noise are an
inherent part of the test environment or room noise and background noise shall be eliminated to such an extent that their
influence on the test results can be neglected.
Unless stated otherwise measurements shall be conducted under quiet and "anechoic" conditions. Depending on the
distance of the transducers from mouth and ear a quiet office room may be sufficient e.g. for handsets where artificial
mouth and artificial ear are located close to the acoustical transducers. But this is not applicable for handsfree and
loudspeaking terminals.
In cases where real or simulated background noise is used as part of the testing environment, the original background
noise must not be noticeably influenced by the acoustical properties of the room.
In all cases where the performance of acoustic echo cancellers shall be tested, a realistic room, which represents the
typical user environment for the terminal shall be used.
In case where an anechoic room is not available the test room has to be an acoustically treated room with few
reflections and a low noise level.
Considering this, test laboratory, in the case where its test room does not conform to anechoic conditions as given in
ITU-T Recommendation P.341 [18], has to present difference in results for measurements due to its test room.
5.6 Influence of terminal delay on measurements
As delay is introduced by the terminal, care shall be taken for all measurements where exact position of the analysis
window is required. It shall be checked that the test is performed on the test signal and not on any other signal.
ETSI
14 ETSI ES 202 740 V1.3.2 (2010-09)
6 Test Setup
In order to use a compatible test system for all types of speech terminals a HATS (Head And Torso Simulator) will be
used instead of free field microphone (for receive measurement) and artificial mouth (for send measurement). HATS is
described in ITU-T Recommendation P.58 [14].
The preferred way of testing a terminal is to connect it to a network simulator with exact defined settings and access
points. The test sequences are fed in either electrically, using a reference codec or using the direct signal processing
approach or acoustically using ITU-T specified devices.
When, a coder with variable bite rate is used, we should adopt, for testing terminal electro acoustical para
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SLOVENSKI STANDARD
01-julij-2016
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VWRULWHY 4R6 NRWMLKGRMHPDXSRUDEQLN
Speech and multimedia Transmission Quality (STQ) - Transmission requirements for
wideband VoIP loudspeaking and handsfree terminals from a QoS perspective as
perceived by the user
Ta slovenski standard je istoveten z: ETSI ES 202 740 V1.3.2 (2010-09)
ICS:
33.050.01 Telekomunikacijska Telecommunication terminal
terminalska oprema na equipment in general
splošno
2003-01.Slovenski inštitut za standardizacijo. Razmnoževanje celote ali delov tega standarda ni dovoljeno.
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