1、 TSB-116-A (Revision of TSB-116) March 2006 (Reaffirmed May 2011) Telecommunications IP Telephony equipment Voice Quality Recommendations for IP Telephony NOTICE OF COPYRIGHT This document is copyrighted by the TIA. Reproduction of these documents either in hard copy or soft copy (including posting
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14、S CONTAINED HEREIN, INCLUDING WITHOUT LIMITATION ANY AND ALL INDIRECT, SPECIAL, INCIDENTAL OR CONSEQUENTIAL DAMAGES (INCLUDING DAMAGES FOR LOSS OF BUSINESS, LOSS OF PROFITS, LITIGATION, OR THE LIKE), WHETHER BASED UPON BREACH OF CONTRACT, BREACH OF WARRANTY, TORT (INCLUDING NEGLIGENCE), PRODUCT LIAB
15、ILITY OR OTHERWISE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGES. THE FOREGOING NEGATION OF DAMAGES IS A FUNDAMENTAL ELEMENT OF THE USE OF THE CONTENTS HEREOF, AND THESE CONTENTS WOULD NOT BE PUBLISHED BY TIA WITHOUT SUCH LIMITATIONS. TIA/TSB-116-A i TABLE OF CONTENTS 1. INTRODUCTION . 1 1.1.
16、SUMMARY OF VOICE QUALITY RECOMMENDATIONS FOR IP TELEPHONY. 1 2. REFERENCES 2 3. DEFINITIONS, ABBREVIATIONS AND ACRONYMS . 4 3.1. WAVEFORM CODEC AND SPEECH COMPRESSION CODEC . 4 3.2. ABBREVIATIONS AND ACRONYMS 4 4. THE E-MODEL. 5 4.1. TRANSMISSION RATING FACTOR “R” . 6 4.2. IP TELEPHONY IMPAIRMENTS A
17、ND THE E-MODEL. 6 4.2.1. Delay . 7 4.2.2. Echo 8 4.2.3. Speech Compression . 11 4.2.4. Packet Loss . 13 4.3. WHAT DOES R SOUND LIKE? . 16 4.4. E-MODEL SYMMETRY AND PERFORMANCE 17 4.5. E-MODEL ENHANCEMENTS 17 4.6. E-MODEL CONVENTIONS 18 5. WIRELINE PSTN VOICE QUALITY BENCHMARKS 19 5.1. ISDN VOICE QUA
18、LITY. 20 5.2. PSTN VOICE QUALITY 21 5.3. TOLL COMPRESSION VOICE QUALITY 24 5.4. WIRELINE PSTN VOICE QUALITY SUMMARY 26 6. IP TELEPHONY VOICE QUALITY ANALYSIS 27 6.1. VOICE QUALITY ISSUES FOR IP TELEPHONY . 29 6.2. VOICE QUALITY RECOMMENDATIONS FOR IP TELEPHONY 30 6.2.1. Delay .30 6.2.2. Speech Compr
19、ession . 35 6.2.3. Packet Loss . 37 6.2.4. Transcoding. 41 6.2.5. Tandeming 42 6.2.6. New Voice Gateway and Digital End Office Loss Plans 45 ANNEX A (INFORMATIVE) VOIP END-TO-END DELAY BUDGET PLANNING FOR PRIVATE NETWORKS 47 A.1. VOIP END-TO-END DELAY SOURCES OVERVIEW. 47 A.2. VOIP END-TO-END DELAY
20、SOURCE DEFINITIONS 48 A.3. VOIP END-TO-END DELAY BUDGET CASE 1 . 51 A.4. VOIP END-TO-END DELAY BUDGET CASE 2 . 53 A.4. VOIP END-TO-END DELAY BUDGET CASE 2 . 53 TIA/TSB-116-A ii FOREWORD (This foreword is not part of this standard.) This document is a TIA Telecommunications Technical Services Bulleti
21、n (TSB) revised by Subcommittee TR-41.1 of Committee TR-41. This TSB was developed in accordance with TIA procedural guidelines, and represents the consensus position of the Subcommittee TR-41.1, which served as the formulating group. The TR-41.1.2 VoIP Voice Quality Working Group acknowledges the c
22、ontribution made by the following individuals in the development of this standard. Name Representing Roger Britt Nortel Networks Editor Mark Armstrong Nortel Networks Dermot Kavanagh Nortel Networks Peter Melton eOn Communications Kirit Patel Cisco Systems Copyrighted parts of ITU-T Appendix I to Re
23、commendation G.113 and Recommendation G.114 are used with permission of the ITU. The ITU owns the copyright for the ITU Recommendations. Suggestions for improvement of this standard are welcome. They should be sent to: Telecommunications Industry Association Engineering Department Suite 300 250 Wils
24、on Boulevard Arlington, VA 22201 TIA/TSB-116-A 1 1. Introduction The objectives of this TSB are to provide end-to-end voice quality guidelines for North American IP Telephony and to an E-Model tutorial for IP scenarios. IP Telephony introduces several impairments, some of which are familiar and some
25、 new. The E-Model (ITU-T Recommendation G.107) is a tool that can estimate the end-to-end voice quality, taking the IP Telephony parameters and impairments into account. This TSB first describes how the E-Model handles IP Telephony impairments and then it provides general design recommendations for
26、the best possible voice quality performance irrespective of cost, available technology or customer requirements. These recommendations are illustrated with specific IP scenarios to provide an E-Model tutorial for analyzing real networks. Since initially, IP telephony is a replacement technology for
27、the existing wireline PSTN, the focus of this document is on wireline scenarios. The impairments introduced by wireline IP packet technology can be significant. The reader should be aware that wireless and satellite technologies also introduce significant impairments and that only a few of the combi
28、ned effects are illustrated here. This TSB builds on similar work done for North American PBX private networks that was published in TIA/EIA/TSB32-A and the focus of this document remains on providing guidelines for engineered private networks as opposed to the Internet. The E-Model scenarios detail
29、ed in this TSB, are available for download as a spreadsheet tool at: http:/www.e-model.org/ 1.1. Summary of Voice Quality Recommendations for IP Telephony Section 6 uses the E-Model to develop the following IP Telephony voice quality recommendations: Delay Rec. #1: Use G.711 end-to-end because it ha
30、s the lowest Ie-value and therefore it allows more delay for a given voice quality level. Delay Rec. #2: Minimize the speech frame size and the number of speech frames per packet. Delay Rec. #3: Actively minimize jitter buffer delay. Delay Rec. #4: Actively minimize one-way delay. Delay Rec. #5: Acc
31、ept the E-Model results, which permit longer delays for low Ie-value codecs, like G.711, for a given R-value; see Figure 22 and Figure 27. Delay Rec. #6: Use priority scheduling for voice-class traffic, as well as RTP header compression and data packet fragmentation on slow-speed links to minimize t
32、he contribution of this variable delay source. Delay Rec. #7: Avoid using slow serial links. Speech Compression Rec. #1: Use G.711 unless the link speed demands compression. Speech Compression Rec. #2: Speech compression codecs for wireless networks and packet networks should be rationalized to mini
33、mize transcoding issues. Packet Loss Rec. #1: Keep (random) packet loss well below 1%. Packet Loss Rec. #2: Use packet loss concealment with G.711. Packet Loss Rec. #3: If other codecs are used, then use codecs that have built-in or add-on PLCs. Packet Loss Rec. #4: New PLCs should be optimized for
34、less than 1% of (random) packet loss. Transcoding Rec. #1: Avoid transcoding where possible. Adds Ie and delay impairment. Transcoding Rec. #2: For interoperability, IP gateways should support wireless codecs or IP should implement unified Transcoder Free Operation with wireless. Tandeming Rec. #1:
35、Avoid asynchronous tandeming if possible. Adds Ie and delay impairment. Tandeming Rec. #2: Synchronous tandeming of G.726 is generally permissible. Impairment is delay dependent, so long delay DCME equipment should be avoided. Loss Plan Rec. #1: Use ANSI/TIA-912-A, Voice Gateway Transmission Require
36、ments. Loss Plan Rec. #2: Use ANSI T1.508-2003, Loss Plan for Digital Networks. TIA/TSB-116-A 2 2. References The following documents contain provisions that are referenced in this TSB. At the time of publication, the editions indicated were valid. All standards are subject to revision, and parties
37、to agreements based on this Standard are encouraged to investigate the possibility of applying the most recent editions of the standards indicated below. ANSI and TIA maintain registers of currently valid national standards published by them. 1 ANSI/TIA-464-C-2003, Requirements for PBX Switching Equ
38、ipment. 2 ANSI/TIA-464-C-1-2004, Requirements for PBX Switching Equipment Addendum 1. 3 ANSI/TIA-470.110-C-2004, Handset Acoustic Performance Requirements. 4 ANSI/TIA/EIA-810-A-2000, Transmission Requirements for Narrowband Voice over IP and Voice over PCM Digital Wireline Telephones. 5 ANSI/TIA-912
39、-A-2004, Voice Gateway Transmission Requirements. 6 ANSI T1.508-2003, Loss Plan for Digital Networks. 7 ANSI T1.521-1999, Packet Loss Concealment with ITU-T Recommendation G.711. 8 TIA/EIA/TSB32-A-1998, Overall Transmission Plan Aspects for Telephony in a Private Network. 9 TIA-920-2002, Transmissio
40、n Requirements for Wideband Digital Wireline Telephones. 10 ITU-T Recommendation G.101 (11/03), The transmission plan. 11 ITU-T Recommendation G.107 (12/98), (07/02), (03/03) and (02/05), The E-Model, A Computational Model for use in Transmission Planning. 12 ITU-T Recommendation G.108 (09/99), Appl
41、ication of the E-model: A planning guide. 13 ITU-T Recommendation G.109 (09/99), Definition of categories of speech transmission quality. 14 ITU-T Recommendation G.113 (02/01), Transmission impairments due to speech processing. 15 ITU-T Appendix I to Recommendation G.113 (10/01), Provisional plannin
42、g values for the equipment impairment factor Ie. Note: Although this revision has been superseded, it is the revision referred to by this TSB. 16 ITU-T Recommendation G.114 (05/03), One-way transmission time. 17 CCITT Recommendation G.131 (11/03), Talker echo and its control. 18 ITU-T Recommendation
43、 G.177 (09/99), Transmission planning for voiceband services over hybrid Internet/PSTN connections. 19 CCITT Recommendation G.711 (11/88), Pulse code Modulation (PCM) of voice frequencies. 20 ITU-T Recommendation G.712 (11/01), Transmission performance characteristics of pulse code modulation channe
44、ls. TIA/TSB-116-A 3 21 ITU-T Recommendation G.723.1 (03/96), Speech coders: Dual rate speech coder for multimedia communications transmitting at 5.3 and 6.3 kbit/s. 22 ITU-T Recommendation G.726 (12/90), 40, 32, 24, 16 kbit/s Adaptive Differential Pulse Code M (ADPCM). 23 ITU-T Recommendation G.729
45、(03/96), Coding of speech at 8 kbit/s using conjugate-structure algebraic-code-excited linear-prediction (CS-ACELP). 24 ITU-T Recommendation G.729 Annex A (11/96), Reduced complexity 8 kbit/s CS-ACELP speech codec. TIA/TSB-116-A 4 3. Definitions, Abbreviations and Acronyms 3.1. Waveform Codec and Sp
46、eech Compression Codec A codec is a combination of an analog-to-digital encoder and a digital-to-analog decoder operating in opposite directions of transmission in the same equipment. A “waveform codec” preserves the waveform of the incoming signal, and operates on a sample-by-sample basis. G.711 an
47、d G.726 are examples of waveform codecs. Modern low bit-rate coders use work on a different principle than waveform codecs. They use a uses a model of the human speech mechanism to encode and compress speech signals, based on the analysis of a frame of input samples. The speech model parameters are
48、sent instead of the speech waveform. This document generally uses the term “speech compression codec” for this type of codec, but it also uses the term “vocoder”. Table 1 separates codecs into the waveform and speech compression categories. 3.2. Abbreviations and Acronyms Abbreviations and acronyms,
49、 not in common usage, which appear in this TSB, are defined below. ACELP Algebraic-Code-Excited Linear-Prediction CPE Customer Premise Equipment DCME Digital Circuit Multiplication Equipment DSL Digital Subscriber Line ECAN Echo Canceller EFR Enhanced Full Rate ERL Echo Return Loss ERLE Echo Return Loss Enhancement GoB Good or Better (a quality rating derived from MOS) GSM Global System for Mobile Telecommunications Ie Equipment Impairment factor (an E-Model input parameter) IP Internet Protocol ISDN Integrated Services Digital Network
