1、 INTERNATIONAL TELECOMMUNICATION UNION ITU-T P.1010TELECOMMUNICATION STANDARDIZATION SECTOR OF ITU (07/2004) SERIES P: TELEPHONE TRANSMISSION QUALITY, TELEPHONE INSTALLATIONS, LOCAL LINE NETWORKS Transmission performance and QoS aspects of IP end-points Fundamental voice transmission objectives for
2、VoIP terminals and gateways ITU-T Recommendation P.1010 ITU-T P-SERIES RECOMMENDATIONS TELEPHONE TRANSMISSION QUALITY, TELEPHONE INSTALLATIONS, LOCAL LINE NETWORKS Vocabulary and effects of transmission parameters on customer opinion of transmission quality Series P.10 Subscribers lines and sets Ser
3、ies P.30 P.300 Transmission standards Series P.40 Objective measuring apparatus Series P.50 P.500 Objective electro-acoustical measurements Series P.60 Measurements related to speech loudness Series P.70 Methods for objective and subjective assessment of quality Series P.80 P.800 Audiovisual quality
4、 in multimedia services Series P.900 Transmission performance and QoS aspects of IP end-points Series P.1000 For further details, please refer to the list of ITU-T Recommendations. ITU-T Rec. P.1010 (07/2004) i ITU-T Recommendation P.1010 Fundamental voice transmission objectives for VoIP terminals
5、and gateways Summary This Recommendation provides 3.1 kHz telephony speech transmission performance requirements for the whole range of packet-based gateways and terminals, including wireless and softphones. Measurement methodologies are not covered by this Recommendation, however, work on this topi
6、c is under way in Study Group 12 and is planned to be incorporated in a future revision or, alternatively, in a separate new Recommendation. Also, requirements for wideband telephony may be added in a future version of this Recommendation. Source ITU-T Recommendation P.1010 was approved on 7 July 20
7、04 by ITU-T Study Group 12 (2001-2004) under the ITU-T Recommendation A.8 procedure. ii ITU-T Rec. P.1010 (07/2004) FOREWORD The International Telecommunication Union (ITU) is the United Nations specialized agency in the field of telecommunications. The ITU Telecommunication Standardization Sector (
8、ITU-T) is a permanent organ of ITU. ITU-T is responsible for studying technical, operating and tariff questions and issuing Recommendations on them with a view to standardizing telecommunications on a worldwide basis. The World Telecommunication Standardization Assembly (WTSA), which meets every fou
9、r years, establishes the topics for study by the ITU-T study groups which, in turn, produce Recommendations on these topics. The approval of ITU-T Recommendations is covered by the procedure laid down in WTSA Resolution 1. In some areas of information technology which fall within ITU-Ts purview, the
10、 necessary standards are prepared on a collaborative basis with ISO and IEC. NOTE In this Recommendation, the expression “Administration“ is used for conciseness to indicate both a telecommunication administration and a recognized operating agency. Compliance with this Recommendation is voluntary. H
11、owever, the Recommendation may contain certain mandatory provisions (to ensure e.g. interoperability or applicability) and compliance with the Recommendation is achieved when all of these mandatory provisions are met. The words “shall“ or some other obligatory language such as “must“ and the negativ
12、e equivalents are used to express requirements. The use of such words does not suggest that compliance with the Recommendation is required of any party. INTELLECTUAL PROPERTY RIGHTS ITU draws attention to the possibility that the practice or implementation of this Recommendation may involve the use
13、of a claimed Intellectual Property Right. ITU takes no position concerning the evidence, validity or applicability of claimed Intellectual Property Rights, whether asserted by ITU members or others outside of the Recommendation development process. As of the date of approval of this Recommendation,
14、ITU had not received notice of intellectual property, protected by patents, which may be required to implement this Recommendation. However, implementors are cautioned that this may not represent the latest information and are therefore strongly urged to consult the TSB patent database. ITU 2004 All
15、 rights reserved. No part of this publication may be reproduced, by any means whatsoever, without the prior written permission of ITU. ITU-T Rec. P.1010 (07/2004) iii CONTENTS Page Scope 1 1 Introduction 1 2 References. 1 3 Terms and definitions . 2 4 Abbreviations and acronyms 3 5 End-to-end consid
16、erations 3 6 Transmission characteristics. 3 6.1 Default coding algorithm 4 6.2 Delay. 4 6.3 Send Loudness Rating 5 6.4 Receive Loudness Rating . 5 6.5 Weighted Terminal Coupling Loss (TCLw) 6 6.6 Residual Echo Level. 6 6.7 Further parameters with respect to speech processing devices 6 7 Measurement
17、 considerations 6 ITU-T Rec. P.1010 (07/2004) 1 ITU-T Recommendation P.1010 Fundamental voice transmission objectives for VoIP terminals and gateways Scope This Recommendation provides fundamental speech transmission performance requirements for 3.1 kHz handset telephony of packet-based terminals an
18、d gateways; it applies to both access gateways and trunking gateways. Requirements are given for handset operation, with the assumption that the corresponding requirements for headset and handsfree are obtained via appropriate conversions to take into account the different acoustic reference points.
19、 This Recommendation addresses the whole range of IP-based gateways and terminals, including wireless and softphones. Measurement methodologies are not covered by this Recommendation; however, work on this topic is under way in Study Group 12 and is planned to be incorporated in a future revision or
20、 alternatively in a separate new Recommendation. 1 Introduction This Recommendation limits its scope to the fundamental speech transmission performance requirements for VoIP terminals and VoIP gateways with respect to the interface with the packet-based network. Thus, a number of traditional paramet
21、ers have not been considered in this Recommendation. This is mainly because it is anticipated that the respective requirements from existing Recommendations, e.g., P.310 11 or in the Q.55x series of Recommendations, can be applied to VoIP terminals and VoIP gateways in an appropriate way. Attention
22、should also be paid to the end-to-end considerations provided in clause 5. 2 References The following ITU-T Recommendations and other references contain provisions which, through reference in this text, constitute provisions of this Recommendation. At the time of publication, the editions indicated
23、were valid. All Recommendations and other references are subject to revision; users of this Recommendation are therefore encouraged to investigate the possibility of applying the most recent edition of the Recommendations and other references listed below. A list of the currently valid ITU-T Recomme
24、ndations is regularly published. The reference to a document within this Recommendation does not give it, as a stand-alone document, the status of a Recommendation. 1 ITU-T Recommendation G.100 (2001), Definitions used in Recommendations on general characteristics of international telephone connecti
25、ons and circuits. 2 ITU-T Recommendation G.107 (2003), The E-model, a computational model for use in transmission planning. 3 ITU-T Recommendation G.108 (1999), Application of the E-model: A planning guide, plus Amendment 1 (2003): New Appendix I The relationship between and interaction of talker ec
26、ho and absolute delay, and Amendment 2 (2004): New Appendix II Planning examples regarding delay in packet-based networks. 4 ITU-T Recommendation G.108.2 (2003), Transmission planning aspects of echo cancellers. 5 ITU-T Recommendation G.109 (1999), Definition of categories of speech transmission qua
27、lity. 2 ITU-T Rec. P.1010 (07/2004) 6 ITU-T Recommendation G.168 (2004), Digital network echo cancellers. 7 ITU-T Recommendation G.711 (1988), Pulse code modulation (PCM) of voice frequencies. 8 ITU-T Recommendation G.799.1/Y.1451.1 (2004), Functionality and interface specifications for GSTN transpo
28、rt network equipment for interconnecting GSTN and IP networks. 9 ITU-T Recommendation G.1020 (2003), Performance parameter definitions for quality of speech and other voiceband applications utilizing IP network, plus Amendment 1. 10 ITU-T Recommendation P.10 (1998), Vocabulary of terms on telephone
29、transmission quality and telephone sets, plus Amendment 1 (2003): New Annex A List of psychoacoustic parameters. 11 ITU-T Recommendation P.310 (2003), Transmission characteristics for telephone band (300-3400 Hz) digital telephones. 12 ITU-T Recommendation P.330 (2003), Speech processing devices for
30、 acoustic enhancement. 13 ITU-T Recommendation P.340 (2000), Transmission characteristics and speech quality parameters of hands-free terminals. 14 ITU-T Recommendation P.380 (2003), Electro-acoustic measurements on headsets. 15 ITU-T Recommendation P.501 (2000), Test signals for use in telephonomet
31、ry. 16 ITU-T Recommendation P.502 (2000), Objective test methods for speech communication systems using complex test signals. 17 ITU-T Recommendation Q.115.1 (2002), Logic for the control of echo control devices and functions. 18 ITU-T Recommendation Y.1541 (2002), Network performance objectives for
32、 IP-based services. 19 ANSI/TIA/EIA-810-A (2000), Transmission Requirements for Narrowband Voice over IP and Voice over PCM Digital Wireline Telephones. 3 Terms and definitions Beside those in ITU-T Recs P.10 10 and G.100 1 this Recommendation defines the following terms: 3.1 send loudness rating (S
33、LR): SLR = SLR(set) + =nii1CLR 3.2 receive loudness rating (RLR): RLR = RLR(set) + +=Nnii1CLR 3.3 overall loudness rating (OLR): OLR = SLR + RLRmmmmmmmm 3.4 circuit loudness rating (CLR): The loudness loss between two electrical interfaces in a connection or circuit. 3.5 softphone: A softphone is an
34、 implementation of a VoIP terminal which is realized by a software program enabling a personal computer (PC) or personal digital assistant (PDA) equipped with an acoustic interface (microphone and earpiece/loudspeaker) to be used for two-way voice communications. ITU-T Rec. P.1010 (07/2004) 3 4 Abbr
35、eviations and acronyms This Recommendation uses the following abbreviations: CLR Circuit Loudness Rating IP Internet Protocol IPDV IP Packet Delay Variation OLR Overall Loudness Rating RLR Receive Loudness Rating RLR(set) RLR of a telephone set SCN Switched-Circuit Network SLR Send Loudness Rating S
36、LR(set) SLR of a telephone set T Mean One-Way Delay TCLw Weighted Terminal Coupling Loss UNI User-Network Interface VAD Voice Activity Detection VoIP Voice over Internet Protocol 5 End-to-end considerations In order to achieve a desired end-to-end speech transmission performance (mouth-to-ear) it is
37、 recommended that transmission planning tasks be carried out with the E-model of ITU-T Rec. G.107 2 as described in ITU-T Rec. G.108 3; this includes the a priori determination of the desired category of speech transmission quality as defined in ITU-T Rec. G.109 5. While, in general, the transmissio
38、n 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 Rec. G.108 3, this approach is not applicable in packet-based systems and, thus, there is a need for the transmission planners spe
39、cific attention. In particular, the decision as to which delay category given in this Recommendation should be chosen to represent the specific configuration is the responsibility of the individual transmission planner. ITU-T Rec. G.108 with its Amendments 3 provides further guidance on this importa
40、nt issue. 6 Transmission characteristics The requirements specified in this clause apply to both VoIP terminals and VoIP gateways with respect to the interface with the packet-based network as illustrated in Figure 1. 4 ITU-T Rec. P.1010 (07/2004) P1010_F1Figure 1/P.1010 Area of applicability of thi
41、s Recommendation 6.1 Default coding algorithm VoIP terminals and VoIP gateways shall support the coding algorithm according to ITU-T Rec. G.711 7 (both -law and A-law). VoIP terminals and VoIP gateways may support other coding algorithms. 6.2 Delay While delay incurred from traditional terminals was
42、 significantly lower than the delay possibly incurred from the network, this is no longer true for VoIP terminals and for VoIP gateways. Hence, for traditional terminals, the appropriate Recommendations (e.g., ITU-T Rec. P.310) were able to specify one single requirement for delay. With the introduc
43、tion of packet-based transmission, however, the increased delays introduced by the terminals and the gateways can have a tremendous impact on the end-to-end delay and, thus, need proper consideration in transmission planning. For example, depending on the delay variation accumulated by packets trave
44、rsing an IP network, the design of the de-jitter buffer of an IP gateway or terminal is extremely critical and has to consider a trade-off between the permissible number of dropped packets and the maximum delay to be incurred (see Figure 3/G.1020 9). Accordingly, for the parameter of delay, IP termi
45、nals and gateways may be designed to operate in a number of modes, with the mode selection being based on end-to-end considerations, especially desired speech quality level and likely network configurations. For IP networks which comply with or which perform better than Class 0 of ITU-T Rec. Y.1541
46、18, terminals or gateways that comply with any of the three delay categories given in this Recommendation will be a starting point to achieve acceptable speech quality. For other kinds of networks, for example, networks with larger delay variations than any class of Y.1541 limits, only delay categor
47、ies A or B may be a valid choice. Furthermore, there are cases such as the interconnection of two local IP islands via an intercontinental path through the PSTN, where only delay category A will be a valid basis for achieving a reasonable end-to-end speech transmission performance (due to the long d
48、elay of the PSTN path). Irrespective of network configurations encountered, the deployment of some low bit rate codecs may further decrease the perceived speech quality and narrow the valid choice of delay categories. The following subclauses specify three delay categories for either send or receive
49、 directions. It is the common understanding that, in order to comply with a specific delay category, a VoIP terminal or a VoIP gateway has to comply with both send and receive requirements of this category. NOTE For some VoIP terminals, e.g., softphones, it is well recognized that, under certain circumstances, technical and practical considerations may not permit any of the above categories to be met at this time. Therefore, this issue, as well as the introduction of a fourth delay category, are for further study. I
