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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: February 15, 2015 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 August 14, 2014 18 Additional WebRTC audio codecs for interoperability with legacy 19 networks. 20 draft-proust-rtcweb-audio-codecs-for-interop-01 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 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 February 15, 2015. 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. Code Components extracted from this document must 62 include Simplified BSD License text as described in Section 4.e of 63 the Trust Legal Provisions and are provided without warranty as 64 described in the Simplified BSD License. 66 Table of Contents 68 1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 3 69 2. Terminology . . . . . . . . . . . . . . . . . . . . . . . . . 3 70 3. Definitions . . . . . . . . . . . . . . . . . . . . . . . . . 3 71 4. Rationale for additional WebRTC codecs . . . . . . . . . . . 3 72 5. Additional suitable codecs for WebRTC . . . . . . . . . . . . 5 73 5.1. AMR-WB . . . . . . . . . . . . . . . . . . . . . . . . . 5 74 5.1.1. AMR-WB General description . . . . . . . . . . . . . 5 75 5.1.2. WebRTC relevant use case for AMR-WB . . . . . . . . . 5 76 5.1.3. Guidelines for AMR-WB usage and implementation with 77 WebRTC . . . . . . . . . . . . . . . . . . . . . . . 5 78 5.2. AMR . . . . . . . . . . . . . . . . . . . . . . . . . . . 5 79 5.2.1. AMR General description . . . . . . . . . . . . . . . 6 80 5.2.2. WebRTC relevant use case for AMR . . . . . . . . . . 6 81 5.2.3. Guidelines for AMR usage and implementation with 82 WebRTC . . . . . . . . . . . . . . . . . . . . . . . 6 83 5.3. G.722 . . . . . . . . . . . . . . . . . . . . . . . . . . 6 84 5.3.1. G.722 General description . . . . . . . . . . . . . . 6 85 5.3.2. WebRTC relevant use case for G.722 . . . . . . . . . 7 86 5.3.3. Guidelines for G.722 usage and implementation . . . . 7 87 5.4. [Codec x] (tbd) . . . . . . . . . . . . . . . . . . . . . 7 88 5.4.1. [Codec X] General description . . . . . . . . . . . . 7 89 5.4.2. WebRTC relevant use case for [Codec X] . . . . . . . 7 90 5.4.3. Guidelines for [Codec X] usage and implementation 91 with WebRTC . . . . . . . . . . . . . . . . . . . . . 7 92 6. Security Considerations . . . . . . . . . . . . . . . . . . . 7 93 7. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 7 94 8. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . 8 95 9. References . . . . . . . . . . . . . . . . . . . . . . . . . 8 96 9.1. Normative references . . . . . . . . . . . . . . . . . . 8 97 9.2. Informative references . . . . . . . . . . . . . . . . . 8 98 Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 9 100 1. Introduction 102 As indicated in [I-D.ietf-rtcweb-overview], it has been anticipated 103 that WebRTC will not remain an isolated island and that some WebRTC 104 endpoints will need to communicate with devices used in other 105 existing networks with the help of a gateway. Therefore, in order to 106 maximize the possibility to establish the session without the need 107 for audio transcoding, it is recommended in [I-D.ietf-rtcweb-audio] 108 to include in the offer other suitable audio codecs that are 109 available to the browser. This document provides some guidelines on 110 the suitable codecs to be considered for WebRTC clients to address 111 the most relevant interoperability use cases. 113 The purpose of this 01 version is to maintain the draft alive. The 114 authors will submit a 02 version before IETF-91, which will take into 115 account the comments received at IETF-90. 117 2. Terminology 119 In this document, the key words "MUST", "MUST NOT", "REQUIRED", 120 "SHALL", "SHALL NOT", "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", 121 and "OPTIONAL" are to be interpreted as described in RFC 2119 122 [RFC2119]. 124 3. Definitions 126 Legacy networks: In this draft, legacy networks encompass the 127 conversational networks that are already deployed like the PSTN, the 128 PLMN, the IMS, H.323 networks. 130 4. Rationale for additional WebRTC codecs 132 The mandatory implementation of OPUS [RFC6716] in WebRTC clients can 133 guarantee the codec interoperability (without transcoding) at the 134 state of the art voice quality (better than narrow band "PSTN" 135 quality) only between WebRTC clients. The WebRTC technology is 136 however expected to have more extended usage to communicate with 137 other types of clients. It can be used for instance as an access 138 technology to 3GPP IMS services or to interoperate with fixed or 139 mobile VoIP legacy HD voice service. Consequently, a significant 140 number of calls are likely to occur between terminals supporting 141 WebRTC clients and other terminals like mobile handsets, fixed VoIP 142 terminals, DECT terminals that do not support WebRTC clients nor 143 implement OPUS. As a consequence, these calls are likely to be 144 either of low narrow band PSTN quality using G.711 at both ends or 145 affected by transcoding operations. The drawbacks of such 146 transcoding operations are recalled below: 148 o Degraded user experience with respect to voice quality: voice 149 quality is significantly degraded by transcoding. For instance, 150 the degradation is around 0.2 to 0.3 MOS for most of transcoding 151 use cases with AMR-WB at 12.65 kbit/s and in the same range for 152 other wideband transcoding cases. It should be stressed that if 153 G.711 is used as a fall back codec for interoperation, wideband 154 voice quality will be lost. Such bandwidth reduction effect down 155 to narrow band clearly degrades the user perceived quality of 156 service leading to shorter and less frequent calls. Such a switch 157 to G.711 is less than desirable or acceptable choice for 158 customers. If transcoding is performed between OPUS and any other 159 wideband codec, wideband communication could be maintained but 160 with degraded quality (MOS scores of transcoding between AMR-WB 161 12.65kbit/s and OPUS at 16 kbit/s in both directions are 162 significantly lower than those of AMR-WB at 12.65kbit/s or OPUS at 163 16 kbit/s). Furthermore, in degraded conditions, the addition of 164 defects, like audio artifacts due to packet losses, and the audio 165 effects resulting from the cascading of different packet loss 166 recovery algorithms may result in a quality below the acceptable 167 limit for the customers. 169 o Degraded user experience with respect to conversational 170 interactivity: the degradation of conversational interactivity is 171 due to the increase of end to end latency for both directions that 172 is introduced by the transcoding operations. Transcoding requires 173 full de-packetization for decoding of the media stream (including 174 mechanisms of de-jitter buffering and packet loss recovery) then 175 re-encoding, re-packetization and re-sending. The delays produced 176 by all these operations are additive and may increase the end to 177 end delay beyond acceptable limits like with more than 1s end to 178 end latency. 180 o Additional costs in networks: transcoding places important 181 additional costs on network gateways mainly related to codec 182 implementation, codecs license, deployments, testing and 183 validation costs. It must be noted that transcoding of wideband 184 to wideband would require more CPU and be more costly than between 185 narrowband codecs. 187 5. Additional suitable codecs for WebRTC 189 The following codecs are considered as relevant suitable codecs with 190 respect to the general purpose described in section 4. This list 191 reflects the current status of WebRTC foreseen use cases. It is not 192 limitative and opened to further inclusion of other codecs for which 193 relevant use cases can be identified. 195 5.1. AMR-WB 197 5.1.1. AMR-WB General description 199 The Adaptive Multi-Rate WideBand (AMR-WB) is a 3GPP defined speech 200 codec that is mandatory to implement in any 3GPP terminal that 201 supports wideband speech communication. It is being used in circuit 202 switched mobile telephony services and new multimedia telephony 203 services over IP/IMS and 4G/VoLTE, specified by GSMA as voice IMS 204 profile for VoLTE in [IR.92]. More detailed information on AMR-WB 205 can be found in [IR.36]. [IR.36] includes references for all 3GPP 206 AMR-WB related specifications including detailed codec description 207 and Source code. 209 5.1.2. WebRTC relevant use case for AMR-WB 211 The market of voice personal communication is driven by mobile 212 terminals. AMR-WB is now implemented in more than 200 devices models 213 and 85 HD mobile networks in 60 countries with a customer base of 214 more than 100 million. A high number of calls are consequently 215 likely to occur between WebRTC clients and mobile 3GPP terminals. 216 The use of AMR-WB by WebRTC clients would consequently allow 217 transcoding free interoperation with all mobile 3GPP wideband 218 terminal. Besides, WebRTC clients running on mobile terminals 219 (smartphones) may reuse the AMR-WB codec already implemented on these 220 devices. 222 5.1.3. Guidelines for AMR-WB usage and implementation with WebRTC 224 Guidelines for implementing and using AMR-WB and ensuring 225 interoperability with 3GPP mobile services can be found in 226 [TS26.114]. In order to ensure interoperability with 4G/VoLTE as 227 specified by GSMA, the more specific IMS profile for voice derived 228 from [TS26.114] should be considered in [IR.92]. 230 5.2. AMR 231 5.2.1. AMR General description 233 Adaptive Multi-Rate (AMR) is a 3GPP defined speech codec that is 234 mandatory to implement in any 3GPP terminal that supports voice 235 communication, i.e. several hundred millions of terminals. This 236 include both mobile phone calls using GSM and 3G cellular systems as 237 well as multimedia telephony services over IP/IMS and 4G/VoLTE, such 238 as GSMA voice IMS profile for VoLTE in [IR.92]. In addition to 239 impacts listed above, support of AMR can avoid degrading the high 240 efficiency over mobile radio access. 242 5.2.2. WebRTC relevant use case for AMR 244 A user of a WebRTC endpoint on a device integrating an AMR module 245 wants to communicate with another user that can only be reached on a 246 mobile device that only supports AMR. Although more and more 247 terminal devices are now "HD voice" and support AMR-WB; there is 248 still a high number of legacy terminals supporting only AMR 249 (terminals with no wideband / HD Voice capabilities) are still used. 250 The use of AMR by WebRTC client would consequently allow transcoding 251 free interoperation with all mobile 3GPP terminals. Besides, WebRTC 252 client running on mobile terminals (smartphones) may reuse the AMR 253 codec already implemented on these devices. 255 5.2.3. Guidelines for AMR usage and implementation with WebRTC 257 Guidelines for implementing and using AMR with purpose to ensure 258 interoperability with 3GPP mobile services can be found in 259 [TS26.114]. In order to ensure interoperability with 4G/VoLTE as 260 specified by GSMA, the more specific IMS profile for voice derived 261 from [TS26.114] should be considered in [IR.92]. 263 5.3. G.722 265 5.3.1. G.722 General description 267 G.722 is an ITU-T defined wideband speech codec. [G.722] was 268 approved by ITU-T in 1988. It is a royalty free codec that is common 269 in a wide range of terminals and end-points supporting wideband 270 speech and requiring low complexity. The complexity of G.722 is 271 estimated to 10 MIPS [EN300175-8] which is 2.5 to 3 times lower than 272 AMR-WB. Especially, G.722 has been chosen by ETSI DECT as the 273 mandatory wideband codec for New Generation DECT with purpose to 274 greatly increase the voice quality by extending the bandwidth from 275 narrow band to wideband. G.722 is the wideband codec required for 276 CAT-iq DECT certified terminal and the V2.0 of CAT-iq specifications 277 have been approved by GSMA as minimum requirements for HD voice logo 278 usage on "fixed" devices; i.e., broadband connections using the G.722 279 codec. 281 5.3.2. WebRTC relevant use case for G.722 283 G.722 is the wideband codec required for DECT CAT-iq terminals. The 284 market for DECT cordeless phones including DECT chipset is more than 285 150 Millions per year and CAT-IQ is a registered trade make in 47 286 countries worldwide. G.722 has also been specified by ETSI in 287 [TS181005] as mandatory wideband codec for IMS multimedia telephony 288 communication service and supplementary services using fixed 289 broadband access. The support of G.722 would consequently allow 290 transcoding free IP interoperation between WebRTC client and fixed 291 VoIP terminals including DECT / CAT-IQ terminals supporting G.722. 292 Besides, WebRTC client running on fixed terminals implementing G.722 293 may reuse the G.722 codec already implemented on these devices. 295 5.3.3. Guidelines for G.722 usage and implementation 297 Guidelines for implementing and using G.722 with purpose to ensure 298 interoperability with Multimedia Telephony services overs IMS can be 299 found in section 7 of [TS26.114]. Additional information of G.722 300 implementation in DECT can be found in [EN300175-8] and full codec 301 description and C source code in [G.722]. 303 5.4. [Codec x] (tbd) 305 5.4.1. [Codec X] General description 307 tbd 309 5.4.2. WebRTC relevant use case for [Codec X] 311 tbd 313 5.4.3. Guidelines for [Codec X] usage and implementation with WebRTC 315 tbd 317 6. Security Considerations 319 7. IANA Considerations 321 None. 323 8. Acknowledgements 325 Thanks to Milan Patel for his review. 327 9. References 329 9.1. Normative references 331 [RFC2119] Bradner, S., "Key words for use in RFCs to Indicate 332 Requirement Levels", BCP 14, RFC 2119, March 1997. 334 9.2. Informative references 336 [EN300175-8] 337 ETSI, "ETSI EN 300 175-8, v2.5.1: "Digital Enhanced 338 Cordless Telecommunications (DECT); Common Interface (CI); 339 Part 8: Speech and audio coding and transmission".", 2009. 341 [G.722] ITU, "Recommendation ITU-T G.722 (2012): "7 kHz audio- 342 coding within 64 kbit/s".", 2012. 344 [I-D.ietf-rtcweb-audio] 345 Valin, J. and C. Bran, "WebRTC Audio Codec and Processing 346 Requirements", draft-ietf-rtcweb-audio-05 (work in 347 progress), February 2014. 349 [I-D.ietf-rtcweb-overview] 350 Alvestrand, H., "Overview: Real Time Protocols for 351 Browser-based Applications", draft-ietf-rtcweb-overview-10 352 (work in progress), June 2014. 354 [I-D.ietf-rtcweb-use-cases-and-requirements] 355 Holmberg, C., Hakansson, S., and G. Eriksson, "Web Real- 356 Time Communication Use-cases and Requirements", draft- 357 ietf-rtcweb-use-cases-and-requirements-14 (work in 358 progress), February 2014. 360 [IR.36] GSMA, "Adaptive Multirate Wide Band", 2013. 362 [IR.92] GSMA, "IMS Profile for Voice and SMS", 2013. 364 [RFC6716] Valin, JM., Vos, K., and T. Terriberry, "Definition of the 365 Opus Audio Codec", RFC 6716, September 2012. 367 [TS181005] 368 ETSI, "Telecommunications and Internet converged Services 369 and Protocols for Advanced Networking (TISPAN); Service 370 and Capability Requirements V3.3.1 (2009-12)", 2009. 372 [TS26.114] 373 3GPP, "IP Multimedia Subsystem (IMS); Multimedia 374 telephony; Media handling and interaction", 2011. 376 Authors' Addresses 378 Stephane Proust 379 Orange 380 2, avenue Pierre Marzin 381 Lannion 22307 382 France 384 Email: stephane.proust@orange.com 386 Espen Berger 387 Cisco 389 Email: espeberg@cisco.com 391 Bernhard Feiten 392 Deutsche Telekom 394 Email: Bernhard.Feiten@telekom.de 396 Bo Burman 397 Ericsson 399 Email: bo.burman@ericsson.com 401 Kalyani Bogineni 402 Verizon Wireless 404 Email: Kalyani.Bogineni@VerizonWireless.com 406 Miao Lei 407 Huawei 409 Email: lei.miao@huawei.com 410 Enrico Marocco 411 Telecom Italia 413 Email: enrico.marocco@telecomitalia.it