SIST-TP TR 101 329-7 V2.1.1:2004
(Main)Telecommunications and Internet Protocol Harmonization Over Networks (TIPHON) Release 3; End-to-end Quality of Service in TIPHON systems; Part 7: Design guide for elements of a TIPHON connection from an end-to-end speech transmission performance point of view
Telecommunications and Internet Protocol Harmonization Over Networks (TIPHON) Release 3; End-to-end Quality of Service in TIPHON systems; Part 7: Design guide for elements of a TIPHON connection from an end-to-end speech transmission performance point of view
Enhancement for Design Guide for Elements of TIPHON connection from end-to-end speech transmission performance point of view.
Harmonizacija telekomunikacij in internetnega protokola prek omrežij (TIPHON), 3. izdaja - Kakovost storitve od konca do konca v sistemih TIPHON - 7. del: Vodilo za načrtovanje elementov povezav TIPHON z vidika zmogljivosti prenosa govora od konca do konca
General Information
Standards Content (Sample)
SLOVENSKI STANDARD
SIST-TP TR 101 329-7 V2.1.1:2004
01-april-2004
+DUPRQL]DFLMDWHOHNRPXQLNDFLMLQLQWHUQHWQHJDSURWRNRODSUHNRPUHåLM 7,3+21
L]GDMD.DNRYRVWVWRULWYHRGNRQFDGRNRQFDYVLVWHPLK7,3+21GHO 9RGLOR]D
QDþUWRYDQMHHOHPHQWRYSRYH]DY7,3+21]YLGLND]PRJOMLYRVWLSUHQRVDJRYRUDRG
NRQFDGRNRQFD
Telecommunications and Internet Protocol Harmonization Over Networks (TIPHON)
Release 3; End-to-end Quality of Service in TIPHON systems; Part 7: Design guide for
elements of a TIPHON connection from an end-to-end speech transmission performance
point of view
Ta slovenski standard je istoveten z: TR 101 329-7 Version 2.1.1
ICS:
33.020 Telekomunikacije na splošno Telecommunications in
general
SIST-TP TR 101 329-7 V2.1.1:2004 en
2003-01.Slovenski inštitut za standardizacijo. Razmnoževanje celote ali delov tega standarda ni dovoljeno.
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SIST-TP TR 101 329-7 V2.1.1:2004
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SIST-TP TR 101 329-7 V2.1.1:2004
ETSI TR 101 329-7 V2.1.1 (2002-02)
Technical Report
Telecommunications and Internet Protocol
Harmonization Over Networks (TIPHON) Release 3;
End-to-end Quality of Service in TIPHON systems;
Part 7: Design guide for elements of a TIPHON
connection from an end-to-end speech
transmission performance point of view
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SIST-TP TR 101 329-7 V2.1.1:2004
2 ETSI TR 101 329-7 V2.1.1 (2002-02)
Reference
RTR/TIPHON-05014
Keywords
coding, E-model, internet, IP, network,
performance, planning, protocol, QoS, quality,
speech, transmission, voice
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© European Telecommunications Standards Institute 2002.
All rights reserved.
ETSI
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3 ETSI TR 101 329-7 V2.1.1 (2002-02)
Contents
Intellectual Property Rights.5
Foreword.5
Introduction .6
1 Scope.7
2 References.7
3 Definitions and abbreviations.8
3.1 Definitions.8
3.2 Abbreviations.9
4 General considerations.10
4.1 Transmission planning.11
4.2 User interaction.11
4.3 Maintenance.11
4.4 Monitoring & verification.11
4.5 Interconnection of TIPHON systems with other IP networks .11
5 Guidance on main transmission parameters.14
5.1 Loudness ratings.14
5.1.1 General considerations.14
5.1.2 IP terminals.14
5.1.3 IP gateways.14
5.2 Mean one-way delay .14
5.2.1 Absolute delay.14
5.3 Delay jitter.15
5.3.1 Jitter buffer implementations .15
5.3.1.1 Static jitter buffers.15
5.3.1.2 Dynamic Jitter Buffers .16
5.3.2 Jitter buffer monitoring capabilities.16
5.3.3 Impact.17
5.4 Echo loss, echo cancellation.18
5.4.1 General considerations.18
5.4.2 IP terminals.18
5.4.3 IP gateways.18
5.5 Coding distortion.18
5.5.1 General considerations.18
5.6 Speech processing other than coding.19
5.6.1 General considerations.19
5.6.2 IP terminals.20
5.6.3 IP gateways.20
5.7 Transcoding in network elements.20
5.8 Packet loss.22
5.9 Example of TIPHON QoS parameter allocation .22
6 Calculation and planning examples.23
6.1 Examples with respect to loudness ratings .25
6.2 Examples with respect to mean one-way delay.25
6.2.1 Delay due to speech processing and packetization .25
6.2.2 Planning examples regarding the occurrence of long delay.28
6.2.2.1 Introduction.28
6.2.2.1.1 Application of the advantage factor A with respect to the following examples .28
6.2.2.1.2 Distinction between different communication situations for the following examples with
regard to the grade of interactivity between the two parties.29
6.2.2.1.3 Introduction of an additional equipment impairment factor with respect to double-talk
situations for the following examples.29
6.2.2.1.4 Purpose and general structure of the following examples .30
ETSI
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4 ETSI TR 101 329-7 V2.1.1 (2002-02)
6.2.2.2 Connections to regions to which significantly shorter delay is available ("Competition").31
6.2.2.2.1 Speech transmission performance as perceived in listening-only communication situations.31
6.2.2.2.2 Speech transmission performance as perceived in typical communication situations.32
6.2.2.2.3 Speech transmission performance as perceived in highly interactive communication
situations.33
6.2.2.3 Connections to regions to which no shorter delay is available ("Hard-to-reach") .34
6.2.2.3.1 Speech transmission performance as perceived in listening-only communication situations.34
6.2.2.3.2 Speech transmission performance as perceived in typical communication situations.35
6.2.2.3.3 Speech transmission performance as perceived in highly interactive communication
situations.36
6.2.2.4 Summary on planning results for long delay.36
6.2.3 VoIP end-to-end delay budget planning for private networks .37
6.2.3.1 VoIP end-to-end delay sources overview.37
6.2.3.2 VoIP end-to-end delay sources definitions .38
6.2.3.3 VoIP End-to-end Delay Budget Case 1.41
6.2.3.4 VoIP End-to-end Delay Budget Case 2.43
6.2.4 E-Model analysis of the VoIP over MPLS reference model.44
6.2.4.1 Introduction.44
6.2.4.2 MPLS in core networks only.45
6.2.4.2.1 Scenario 1 - Multiple MPLS Networks .45
6.2.4.2.2 Scenario 2 - PSTN and typical DCME practice .46
6.2.4.2.3 Scenario 3 - GSM and Typical DCME practice .47
6.2.4.2.4 VoMPLS core network summary .48
6.2.4.3 Extending VoMPLS into the access network.49
6.2.4.3.1 Scenario 4 - VoMPLS access on USA to Japan .49
6.2.4.3.2 Scenario 5 deployment of GSM and VoMPLS access .50
6.2.4.4 Effects of Voice Codecs in the access network.51
6.2.4.4.1 Scenario 6 - Deployment of Codecs in one Access Leg (USA – Japan) .52
6.2.4.4.2 Scenario 7 - Codec Deployment in both Access Legs (USA - Japan).53
6.2.4.4.3 Scenario 8 codec deployment and mobile access (USA - Australia).54
6.2.4.4.4 Voice codec summary .54
6.2.4.5 Overall conclusions.55
6.3 Examples with respect to the provision of proper echo control .55
6.3.1 TIPHON QoS class "narrowband high (2H)" .55
6.3.2 TIPHON QoS class "narrowband medium (2M)".57
6.3.3 TIPHON QoS class "narrowband acceptable (2A)" .59
6.4 Examples with respect to coding distortion.61
6.5 Examples with respect to speech processing other than coding .61
6.6 Examples with respect to packet loss .61
6.7 Interpretation of the results.61
6.8 Guidance on the relation and Interdependency between Auditory MOS, Objective MOS and Predicted
MOS .62
Annex A: Bibliography.64
History .72
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5 ETSI TR 101 329-7 V2.1.1 (2002-02)
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 Technical Report (TR) has been produced by ETSI Project Telecommunications and Internet Protocol
Harmonization Over Networks (TIPHON).
The present document is part 7 of a multi-part deliverable covering End-to-end Quality of Service in TIPHON systems,
as identified below:
TR 101 329-1: "General aspects of Quality of Service (QoS)";
TS 101 329-2: "Definition of speech Quality of Service (QoS) classes";
TS 101 329-3: "Signalling and control of end-to-end Quality of Service (QoS)";
TS 101 329-5: "Quality of Service (QoS) measurement methodologies";
TR 101 329-6: "Actual measurements of network and terminal characteristics and performance parameters in
TIPHON networks and their influence on voice quality";
TR 101 329-7: "Design guide for elements of a TIPHON connection from an end-to-end speech
transmission performance point of view".
Quality of Service aspects of TIPHON Release 4 and 5 Systems will be covered in TS 102 024 and TS 102 025
respectively, and more comprehensive versions of the Release 3 documents listed above will be published as part of
Releases 4 and 5 as work progresses.
ETSI
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Introduction
The present document forms one of a series of technical specifications and technical reports produced by TIPHON
Working Group 5 addressing Quality of Service (QoS) in TIPHON Systems. The structure of this work is illustrated in
figure 1.
Introduction Definition of 5
Speech Classes
TR 101 329-1 TS 101 329-2
General Speech
QoS
Aspects
of QoS Classes
SPEC
REPORT
Generic QoS
Specific Aspects of QoS
TS 101 329-5 TR 101 329-6 TR 101 329-7
TS 101 329-3
QoS Measure- Actual Design
Control ment Test Guidelines
Methods Results
SPEC REPORT REPORT
SPEC
QoS signalling Measurement Repository of Useful info for
requirements methodologies real test results designers
Figure 1: Structure of TIPHON QoS Documentation for Release 3
For a concise understanding of the guidance provided in the present document it is strongly recommended that the
reader be aware of the content of the most recent version of TS 101 329-6 [3] which is a repository of real results.
Figure 2: Void
ETSI
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7 ETSI TR 101 329-7 V2.1.1 (2002-02)
1 Scope
The present document provides a collection of informative background information and guidance to supplement
parts 1 to part 6 of TS 101 329. The issues covered concern the practical design phases for both equipment and
networks with respect to speech performance, and therefore is relevant to TIPHON equipment manufacturers, service
providers and network designers.
2 References
For the purposes of this Technical Report (TR) the following references apply:
[1] ETSI TS 101 329-2: "Telecommunications and Internet Protocol Harmonization Over Networks
(TIPHON) Release 3; End-to-end Quality of Service in TIPHON Systems; Part 2: Definition of
Speech Quality of Service (QoS) Classes".
[2] ETSI TS 101 329-5: "Telecommunications and Internet protocol Harmonization Over Networks
(TIPHON) Release 3; End-to-end Quality of Service in TIPHON Systems; Part 5: Quality of
Service (QoS) measurement methodologies".
[3] ETSI TS 101 329-6: "Telecommunications and Internet Protocol Harmonization Over Networks
(TIPHON) Release 3; End-to-end Quality of Service in TIPHON systems; Part 6: Actual
measurements of network and terminal characteristics and performance parameters in TIPHON
networks and their influence on voice quality".
[4] ITU-T Recommendation G.100: "Definitions used in Recommendations on general characteristics
of international telephone connections and circuits".
[5] ITU-T Recommendation G.122: "Influence of national systems on stability and talker echo in
international connections".
[6] ITU-T Recommendation G.131: "Control of talker echo".
[7] ITU-T Recommendation G.111: "Loudness ratings (LRs) in an international connection".
[8] ITU-T Recommendations P.64: "Determination of sensitivity/frequency characteristics of local
telephone systems".
[9] ITU-T Recommendation G.109: "Definition of categories of speech transmission quality".
[10] ITU-T Recommendation G.726: "40, 32, 24, 16 kbit/s adaptive differential pulse code modulation
(ADPCM)".
[11] ITU-T Recommendation G.114: "One-way transmission time".
[12] ETSI I-ETS 300 245-2: "Integrated Services Digital Network (ISDN); Technical characteristics of
telephony terminals; Part 2: PCM A-law handset telephony".
[13] ITU-T Recommendation P.800: "Methods for subjective determination of transmission quality".
[14] ITU-T Recommendation P.310: "Transmission characteristics for telephone-band (300-3400 Hz)
digital telephones".
[15] ANSI/TIA/EIA-810-A-2000: "Telecommunications-Telephone Terminal Equipment-Transmission
Requirements for Narrowband".
[16] ITU-T Recommendation G.168: "Digital network echo cancellers".
[17] ITU-T Recommendation G.107: "The E-Model, a computational model for use in transmission
planning".
[18] ITU-T Recommendation G.113: "Transmission impairments due to speech processing".
ETSI
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[19] ETSI EG 201 050: "Speech Processing, Transmission and Quality Aspects (STQ); Overall
Transmission Plan Aspects for Telephony in a Private Network".
[20] ITU-T Recommendation G.108: "Application of the E-model: A planning guide".
[21] ETSI ETR 250: "Transmission and Multiplexing (TM); Speech communication quality from
mouth to ear for 3,1 kHz handset telephony across networks".
[22] ITU-T Recommendation P.833: "Methodology for derivation of equipment impairment factors
from subjective listening?only tests".
3 Definitions and abbreviations
3.1 Definitions
For the purposes of the present document, the following terms and definitions apply:
dBm: power level with reference to 1 mW
dBm0: at the reference frequency (1 020 Hz), L dBm0 represents an absolute power level of L dBm measured at the
transmission reference point (0 dBr point), and a level of L + x dBm measured at a point having a relative level of x dBr
NOTE: See ITU-T Recommendation G.100 [4], annex A.4.
echo: unwanted signal delayed to such a degree that it is perceived as distinct from the wanted signal
talker echo: echo produced by reflection near the listener's end of a connection, and disturbing the talker
listener echo: echo produced by double reflected signals and disturbing the listener
Loudness Rating (LR): as used in the G-Series Recommendations for planning; loudness rating is an objective
measure of the loudness loss, i.e. a weighted, electro-acoustic loss between certain interfaces in the telephone network
NOTE: If the circuit between the interfaces is subdivided into sections, the sum of the individual section LRs is
equal to the total LR. In loudness rating contexts, the subscribers are represented from a measuring point
of view by an artificial mouth and an artificial ear respectively, both being accurately specified.
Overall Loudness Rating (OLR): loudness loss between the speaking subscriber's mouth and the listening subscriber's
ear via a connection
Talker Echo Loudness Rating (TELR): loudness loss of the speaker's voice sound reaching his ear as a delayed echo
NOTE: See ITU-T Recommendation G.122 [5], clause 4.2 and ITU-T Recommendation G.131 [6], figure I.1.
Terminal Coupling Loss weighted (TCLw): weighted coupling loss between the receiving port and the sending port
of a terminal due to acoustical coupling at the user interface, electrical coupling due to crosstalk in the handset cord or
within the electrical circuits, seismic coupling through the mechanical parts of the terminal
NOTE: For a digital handset it is commonly in the order of 40 db to 46 dB.
weighted Terminal Coupling Loss-single talk (TCLwst): weighted loss between Rin and Sout network interfaces
when AEC is in normal operation, and when there is no signal coming from the user
weighted Terminal Coupling Loss-double talk (TCLwdt): weighted loss between Rin and Sout network interfaces
when AEC is in normal operation, and when the local user and the far-end user talk simultaneously
Send Loudness Rating (SLR) (from ITU-T Recommendation G.111): loudness loss between the speaking
subscriber's mouth and an electric interface in the network
NOTE: The loudness loss is defined here as the weighted (dB) average of driving sound pressure to measured
voltage. The weighted mean value for ITU-T Recommendations G.111 [7] and G.121 (see Bibliography)
is 7 to 15 in the short term, 7 to 9 in the long term. The rating methodology is described in ITU-T
Recommendations P.64 [8], P.76 (see Bibliography) and P.79 (see Bibliography).
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Receive Loudness Rating (RLR) (from ITU-T Recommendation G.111): loudness loss between an electric interface
in the network and the listening subscriber's ear
NOTE: The loudness loss is here defined as the weighted (dB) average of driving e.m.f. to measured sound
pressure. The weighted mean value for ITU-T Recommendations G.111 [7] and G.121 (see Bibliography)
is 1 to 6 in the short term, 1 to 3 in the long term. The rating methodology is described in ITU-T
Recommendations P.64 [8], P.76 (see Bibliography), P.79 (see Bibliography).
Circuit Loudness Rating (CLR): loudness loss between two electrical interfaces in a connection or circuit, each
interface terminated by its nominal impedance which may be complex
toll quality: In general "toll quality" is a term which is not well defined. Currently, there are two different views:
• ITU-T Recommendation G.109 [9] provides the following guidance:
"Finally, to relate the definitions provided by this Recommendation to concepts and terminology used in the past,
a comment about "toll quality" is in order. "Toll quality" has been used by many different people to mean
different things, but to network planners it really meant that technology being introduced into the network was
robust to the effects of transmission impairments from other sources, and could thus be used in many
configurations where inter-working with other systems would be necessary. In this context, the term "toll
quality" does not have any absolute relation to speech transmission quality today, because, for example, the
impairments of systems such as wireless access or packet-based transport will have the same impact regardless
of whether on a local or a long-distance connection. Instead, the terminology provided here is recommended (i.e.
"best" for R in the range from 90 to 100, "high" in the range from 80 to 90 and "medium" in the range from 70 to
80)."
• Experts on low bit-rate coding (members of ITU-T Study Group 16 and SQEG) use the following explanation:
"In summary, we define toll quality as equivalent to wire-line telephone quality. Basically the 32 kb/s ADPCM
(ITU-T Recommendation G.726 [10]) is considered to be a toll quality coder, and when some low rate coders get
standardized in ITU-T, 32 kb/s ADPCM is used as reference, and if a low rate coder produce equivalent
performance to the 32 kb/s ADPCM, then this is considered to be toll quality."
Consequently, at this time the term "toll quality" should be considered as an internal term of speech coder experts only
which is obsolete and which should
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