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Run idnits with the --verbose option for more detailed information about the items above. -------------------------------------------------------------------------------- 2 Network Working Group S. Proust 3 Internet-Draft Orange 4 Intended status: Informational E. Berger 5 Expires: August 18, 2014 Cisco 6 B. Feiten 7 Deutsche Telekom 8 B. Burman 9 Ericsson 10 K. Bogineni 11 Verizon Wireless 12 M. Lei 13 Huawei 14 E. Marocco 15 Telecom Italia 16 February 14, 2014 18 Additional WebRTC audio codecs for interoperability with legacy 19 networks. 20 draft-proust-rtcweb-audio-codecs-for-interop-00 22 Abstract 24 To ensure a baseline level of interoperability between WebRTC 25 clients, [I-D.ietf-rtcweb-audio] requires a minimum set of codecs. 26 However, to maximize the possibility to establish the session without 27 the need for audio transcoding, it is also recommended to include in 28 the offer other suitable audio codecs that are available to the 29 browser. 31 This document provides some guidelines on the suitable codecs to be 32 considered for WebRTC clients to address the most relevant 33 interoperability use cases. 35 Status of This Memo 37 This Internet-Draft is submitted to IETF in full conformance with the 38 provisions of BCP 78 and BCP 79. 40 Internet-Drafts are working documents of the Internet Engineering 41 Task Force (IETF). Note that other groups may also distribute 42 working documents as Internet-Drafts. The list of current Internet- 43 Drafts is at http://datatracker.ietf.org/drafts/current/. 45 Internet-Drafts are draft documents valid for a maximum of six months 46 and may be updated, replaced, or obsoleted by other documents at any 47 time. It is inappropriate to use Internet-Drafts as reference 48 material or to cite them other than as "work in progress." 49 This Internet-Draft will expire on August 18, 2014. 51 Copyright Notice 53 Copyright (c) 2014 IETF Trust and the persons identified as the 54 document authors. All rights reserved. 56 This document is subject to BCP 78 and the IETF Trust's Legal 57 Provisions Relating to IETF Documents 58 (http://trustee.ietf.org/license-info) in effect on the date of 59 publication of this document. Please review these documents 60 carefully, as they describe your rights and restrictions with respect 61 to this document. 63 Table of Contents 65 1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 3 66 2. Terminology . . . . . . . . . . . . . . . . . . . . . . . . . 3 67 3. Definitions . . . . . . . . . . . . . . . . . . . . . . . . . 3 68 4. Rationale for additional WebRTC codecs . . . . . . . . . . . 3 69 5. Additional suitable codecs for WebRTC . . . . . . . . . . . . 4 70 5.1. AMR-WB . . . . . . . . . . . . . . . . . . . . . . . . . 5 71 5.1.1. AMR-WB General description . . . . . . . . . . . . . 5 72 5.1.2. WebRTC relevant use case for AMR-WB . . . . . . . . . 5 73 5.1.3. Guidelines for AMR-WB usage and implementation with 74 WebRTC . . . . . . . . . . . . . . . . . . . . . . . 5 75 5.2. AMR . . . . . . . . . . . . . . . . . . . . . . . . . . . 5 76 5.2.1. AMR General description . . . . . . . . . . . . . . . 5 77 5.2.2. WebRTC relevant use case for AMR . . . . . . . . . . 6 78 5.2.3. Guidelines for AMR usage and implementation with 79 WebRTC . . . . . . . . . . . . . . . . . . . . . . . 6 80 5.3. G.722 . . . . . . . . . . . . . . . . . . . . . . . . . . 6 81 5.3.1. G.722 General description . . . . . . . . . . . . . . 6 82 5.3.2. WebRTC relevant use case for G.722 . . . . . . . . . 6 83 5.3.3. Guidelines for G.722 usage and implementation . . . . 7 84 5.4. [Codec x] (tbd) . . . . . . . . . . . . . . . . . . . . . 7 85 5.4.1. [Codec X] General description . . . . . . . . . . . . 7 86 5.4.2. WebRTC relevant use case for [Codec X] . . . . . . . 7 87 5.4.3. Guidelines for [Codec X] usage and implementation 88 with WebRTC . . . . . . . . . . . . . . . . . . . . . 7 89 6. Security Considerations . . . . . . . . . . . . . . . . . . . 7 90 7. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 7 91 8. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . 7 92 9. References . . . . . . . . . . . . . . . . . . . . . . . . . 7 93 9.1. Normative references . . . . . . . . . . . . . . . . . . 7 94 9.2. Informative references . . . . . . . . . . . . . . . . . 8 95 Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 8 97 1. Introduction 99 As indicated in [I-D.ietf-rtcweb-overview], it has been anticipated 100 that WebRTC will not remain an isolated island and that some WebRTC 101 endpoints will need to communicate with devices used in other 102 existing networks with the help of a gateway. Therefore, in order to 103 maximize the possibility to establish the session without the need 104 for audio transcoding, it is recommended in [I-D.ietf-rtcweb-audio] 105 to include in the offer other suitable audio codecs that are 106 available to the browser. This document provides some guidelines on 107 the suitable codecs to be considered for WebRTC clients to address 108 the most relevant interoperability use cases. 110 2. Terminology 112 In this document, the key words "MUST", "MUST NOT", "REQUIRED", 113 "SHALL", "SHALL NOT", "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", 114 and "OPTIONAL" are to be interpreted as described in RFC 2119 115 [RFC2119]. 117 3. Definitions 119 Legacy networks: In this draft, legacy networks encompass the 120 conversational networks that are already deployed like the PSTN, the 121 PLMN, the IMS, H.323 networks. 123 4. Rationale for additional WebRTC codecs 125 The mandatory implementation of OPUS [RFC6716] in WebRTC clients can 126 guarantee the codec interoperability (without transcoding) at the 127 state of the art voice quality (better than narrow band "PSTN" 128 quality) only between WebRTC clients. The WebRTC technology is 129 however expected to have more extended usage to communicate with 130 other types of clients. It can be used for instance as an access 131 technology to 3GPP IMS services or to interoperate with fixed or 132 mobile VoIP legacy HD voice service. Consequently, a significant 133 number of calls are likely to occur between terminals supporting 134 WebRTC clients and other terminals like mobile handsets, fixed VoIP 135 terminals, DECT terminals that do not support WebRTC clients nor 136 implement OPUS. As a consequence, these calls are likely to be 137 either of low narrow band PSTN quality using G.711 at both ends or 138 affected by transcoding operations. The drawbacks of such 139 transcoding operations are recalled below: 141 o Degraded user experience with respect to voice quality: voice 142 quality is significantly degraded by transcoding. For instance, 143 the degradation is around 0.2 to 0.3 MOS for most of transcoding 144 use cases with AMR-WB at 12.65 kbit/s and in the same range for 145 other wideband transcoding cases. It should be stressed that if 146 G.711 is used as a fall back codec for interoperation, wideband 147 voice quality will be lost. Such bandwidth reduction effect down 148 to narrow band clearly degrades the user perceived quality of 149 service leading to shorter and less frequent calls. Such a switch 150 to G.711 is less than desirable or acceptable choice for 151 customers. If transcoding is performed between OPUS and any other 152 wideband codec, wideband communication could be maintained but 153 with degraded quality (MOS scores of transcoding between AMR-WB 154 12.65kbit/s and OPUS at 16 kbit/s in both directions are 155 significantly lower than those of AMR-WB at 12.65kbit/s or OPUS at 156 16 kbit/s). Furthermore, in degraded conditions, the addition of 157 defects, like audio artifacts due to packet losses, and the audio 158 effects resulting from the cascading of different packet loss 159 recovery algorithms may result in a quality below the acceptable 160 limit for the customers. 162 o Degraded user experience with respect to conversational 163 interactivity: the degradation of conversational interactivity is 164 due to the increase of end to end latency for both directions that 165 is introduced by the transcoding operations. Transcoding requires 166 full de-packetization for decoding of the media stream (including 167 mechanisms of de-jitter buffering and packet loss recovery) then 168 re-encoding, re-packetization and re-sending. The delays produced 169 by all these operations are additive and may increase the end to 170 end delay beyond acceptable limits like with more than 1s end to 171 end latency. 173 o Additional costs in networks: transcoding places important 174 additional costs on network gateways mainly related to codec 175 implementation, codecs license, deployments, testing and 176 validation costs. It must be noted that transcoding of wideband 177 to wideband would require more CPU and be more costly than between 178 narrowband codecs. 180 5. Additional suitable codecs for WebRTC 182 The following codecs are considered as relevant suitable codecs with 183 respect to the general purpose described in section 4. This list 184 reflects the current status of WebRTC foreseen use cases. It is not 185 limitative and opened to further inclusion of other codecs for which 186 relevant use cases can be identified. 188 5.1. AMR-WB 190 5.1.1. AMR-WB General description 192 The Adaptive Multi-Rate WideBand (AMR-WB) is a 3GPP defined speech 193 codec that is mandatory to implement in any 3GPP terminal that 194 supports wideband speech communication. It is being used in circuit 195 switched mobile telephony services and new multimedia telephony 196 services over IP/IMS and 4G/VoLTE, specified by GSMA as voice IMS 197 profile for VoLTE in [IR.92]. More detailed information on AMR-WB 198 can be found in [IR.36]. [IR.36] includes references for all 3GPP 199 AMR-WB related specifications including detailed codec description 200 and Source code. 202 5.1.2. WebRTC relevant use case for AMR-WB 204 The market of voice personal communication is driven by mobile 205 terminals. AMR-WB is now implemented in more than 200 devices models 206 and 85 HD mobile networks in 60 countries with a customer base of 207 more than 100 million. A high number of calls are consequently 208 likely to occur between WebRTC clients and mobile 3GPP terminals. 209 The use of AMR-WB by WebRTC clients would consequently allow 210 transcoding free interoperation with all mobile 3GPP wideband 211 terminal. Besides, WebRTC clients running on mobile terminals 212 (smartphones) may reuse the AMR-WB codec already implemented on these 213 devices. 215 5.1.3. Guidelines for AMR-WB usage and implementation with WebRTC 217 Guidelines for implementing and using AMR-WB and ensuring 218 interoperability with 3GPP mobile services can be found in 219 [TS26.114]. In order to ensure interoperability with 4G/VoLTE as 220 specified by GSMA, the more specific IMS profile for voice derived 221 from [TS26.114] should be considered in [IR.92]. 223 5.2. AMR 225 5.2.1. AMR General description 227 Adaptive Multi-Rate (AMR) is a 3GPP defined speech codec that is 228 mandatory to implement in any 3GPP terminal that supports voice 229 communication, i.e. several hundred millions of terminals. This 230 include both mobile phone calls using GSM and 3G cellular systems as 231 well as multimedia telephony services over IP/IMS and 4G/VoLTE, such 232 as GSMA voice IMS profile for VoLTE in [IR.92]. In addition to 233 impacts listed above, support of AMR can avoid degrading the high 234 efficiency over mobile radio access. 236 5.2.2. WebRTC relevant use case for AMR 238 A user of a WebRTC endpoint on a device integrating an AMR module 239 wants to communicate with another user that can only be reached on a 240 mobile device that only supports AMR. Although more and more 241 terminal devices are now "HD voice" and support AMR-WB; there is 242 still a high number of legacy terminals supporting only AMR 243 (terminals with no wideband / HD Voice capabilities) are still used. 244 The use of AMR by WebRTC client would consequently allow transcoding 245 free interoperation with all mobile 3GPP terminals. Besides, WebRTC 246 client running on mobile terminals (smartphones) may reuse the AMR 247 codec already implemented on these devices. 249 5.2.3. Guidelines for AMR usage and implementation with WebRTC 251 Guidelines for implementing and using AMR with purpose to ensure 252 interoperability with 3GPP mobile services can be found in 253 [TS26.114]. In order to ensure interoperability with 4G/VoLTE as 254 specified by GSMA, the more specific IMS profile for voice derived 255 from [TS26.114] should be considered in [IR.92]. 257 5.3. G.722 259 5.3.1. G.722 General description 261 G.722 is an ITU-T defined wideband speech codec. [G.722] was 262 approved by ITU-T in 1988. It is a royalty free codec that is common 263 in a wide range of terminals and end-points supporting wideband 264 speech and requiring low complexity. The complexity of G.722 is 265 estimated to 10 MIPS [EN300175-8] which is 2.5 to 3 times lower than 266 AMR-WB. Especially, G.722 has been chosen by ETSI DECT as the 267 mandatory wideband codec for New Generation DECT with purpose to 268 greatly increase the voice quality by extending the bandwidth from 269 narrow band to wideband. G.722 is the wideband codec required for 270 CAT-iq DECT certified terminal and the V2.0 of CAT-iq specifications 271 have been approved by GSMA as minimum requirements for HD voice logo 272 usage on "fixed" devices; i.e., broadband connections using the G.722 273 codec. 275 5.3.2. WebRTC relevant use case for G.722 277 G.722 is the wideband codec required for DECT CAT-iq terminals. The 278 market for DECT cordeless phones including DECT chipset is more than 279 150 Millions per year and CAT-IQ is a registered trade make in 47 280 countries worldwide. G.722 has also been specified by ETSI in 281 [TS181005] as mandatory wideband codec for IMS multimedia telephony 282 communication service and supplementary services using fixed 283 broadband access. The support of G.722 would consequently allow 284 transcoding free IP interoperation between WebRTC client and fixed 285 VoIP terminals including DECT / CAT-IQ terminals supporting G.722. 286 Besides, WebRTC client running on fixed terminals implementing G.722 287 may reuse the G.722 codec already implemented on these devices. 289 5.3.3. Guidelines for G.722 usage and implementation 291 Guidelines for implementing and using G.722 with purpose to ensure 292 interoperability with Multimedia Telephony services overs IMS can be 293 found in section 7 of [TS26.114]. Additional information of G.722 294 implementation in DECT can be found in [EN300175-8] and full codec 295 description and C source code in [G.722]. 297 5.4. [Codec x] (tbd) 299 5.4.1. [Codec X] General description 301 tbd 303 5.4.2. WebRTC relevant use case for [Codec X] 305 tbd 307 5.4.3. Guidelines for [Codec X] usage and implementation with WebRTC 309 tbd 311 6. Security Considerations 313 7. IANA Considerations 315 None. 317 8. Acknowledgements 319 Thanks to Milan Patel for his review. 321 9. References 323 9.1. Normative references 325 [RFC2119] Bradner, S., "Key words for use in RFCs to Indicate 326 Requirement Levels", BCP 14, RFC 2119, March 1997. 328 9.2. Informative references 330 [EN300175-8] 331 ETSI, "ETSI EN 300 175-8, v2.5.1: "Digital Enhanced 332 Cordless Telecommunications (DECT); Common Interface (CI); 333 Part 8: Speech and audio coding and transmission".", 2009. 335 [G.722] ITU, "Recommendation ITU-T G.722 (2012): "7 kHz audio- 336 coding within 64 kbit/s".", 2012. 338 [I-D.ietf-rtcweb-audio] 339 Valin, J. and C. Bran, "WebRTC Audio Codec and Processing 340 Requirements", draft-ietf-rtcweb-audio-04 (work in 341 progress), January 2014. 343 [I-D.ietf-rtcweb-overview] 344 Alvestrand, H., "Overview: Real Time Protocols for Brower- 345 based Applications", draft-ietf-rtcweb-overview-08 (work 346 in progress), September 2013. 348 [I-D.ietf-rtcweb-use-cases-and-requirements] 349 Holmberg, C., Hakansson, S., and G. Eriksson, "Web Real- 350 Time Communication Use-cases and Requirements", draft- 351 ietf-rtcweb-use-cases-and-requirements-14 (work in 352 progress), February 2014. 354 [IR.36] GSMA, "Adaptive Multirate Wide Band", 2013. 356 [IR.92] GSMA, "IMS Profile for Voice and SMS", 2013. 358 [RFC6716] Valin, JM., Vos, K., and T. Terriberry, "Definition of the 359 Opus Audio Codec", RFC 6716, September 2012. 361 [TS181005] 362 ETSI, "Telecommunications and Internet converged Services 363 and Protocols for Advanced Networking (TISPAN); Service 364 and Capability Requirements V3.3.1 (2009-12)", 2009. 366 [TS26.114] 367 3GPP, "IP Multimedia Subsystem (IMS); Multimedia 368 telephony; Media handling and interaction", 2011. 370 Authors' Addresses 371 Stephane Proust 372 Orange 373 2, avenue Pierre Marzin 374 Lannion 22307 375 France 377 Email: stephane.proust@orange.com 379 Espen Berger 380 Cisco 382 Email: espeberg@cisco.com 384 Bernhard Feiten 385 Deutsche Telekom 387 Email: Bernhard.Feiten@telekom.de 389 Bo Burman 390 Ericsson 392 Email: bo.burman@ericsson.com 394 Kalyani Bogineni 395 Verizon Wireless 397 Email: Kalyani.Bogineni@VerizonWireless.com 399 Miao Lei 400 Huawei 402 Email: lei.miao@huawei.com 404 Enrico Marocco 405 Telecom Italia 407 Email: enrico.marocco@telecomitalia.it