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Bran 5 Expires: May 8, 2016 Plantronics 6 November 5, 2015 8 WebRTC Audio Codec and Processing Requirements 9 draft-ietf-rtcweb-audio-09 11 Abstract 13 This document outlines the audio codec and processing requirements 14 for WebRTC endpoints. 16 Status of This Memo 18 This Internet-Draft is submitted in full conformance with the 19 provisions of BCP 78 and BCP 79. 21 Internet-Drafts are working documents of the Internet Engineering 22 Task Force (IETF). Note that other groups may also distribute 23 working documents as Internet-Drafts. The list of current Internet- 24 Drafts is at http://datatracker.ietf.org/drafts/current/. 26 Internet-Drafts are draft documents valid for a maximum of six months 27 and may be updated, replaced, or obsoleted by other documents at any 28 time. It is inappropriate to use Internet-Drafts as reference 29 material or to cite them other than as "work in progress." 31 This Internet-Draft will expire on May 8, 2016. 33 Copyright Notice 35 Copyright (c) 2015 IETF Trust and the persons identified as the 36 document authors. All rights reserved. 38 This document is subject to BCP 78 and the IETF Trust's Legal 39 Provisions Relating to IETF Documents 40 (http://trustee.ietf.org/license-info) in effect on the date of 41 publication of this document. Please review these documents 42 carefully, as they describe your rights and restrictions with respect 43 to this document. Code Components extracted from this document must 44 include Simplified BSD License text as described in Section 4.e of 45 the Trust Legal Provisions and are provided without warranty as 46 described in the Simplified BSD License. 48 Table of Contents 50 1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 2 51 2. Terminology . . . . . . . . . . . . . . . . . . . . . . . . . 2 52 3. Codec Requirements . . . . . . . . . . . . . . . . . . . . . 2 53 4. Audio Level . . . . . . . . . . . . . . . . . . . . . . . . . 3 54 5. Acoustic Echo Cancellation (AEC) . . . . . . . . . . . . . . 4 55 6. Legacy VoIP Interoperability . . . . . . . . . . . . . . . . 5 56 7. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 5 57 8. Security Considerations . . . . . . . . . . . . . . . . . . . 5 58 9. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . 5 59 10. References . . . . . . . . . . . . . . . . . . . . . . . . . 5 60 10.1. Normative References . . . . . . . . . . . . . . . . . . 6 61 10.2. Informative References . . . . . . . . . . . . . . . . . 6 62 Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 7 64 1. Introduction 66 An integral part of the success and adoption of the Web Real Time 67 Communications (WebRTC) will be the voice and video interoperability 68 between WebRTC applications. This specification will outline the 69 audio processing and codec requirements for WebRTC endpoint 70 implementations. 72 2. Terminology 74 The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT", 75 "SHOULD", "SHOULD NOT", "RECOMMENDED", "NOT RECOMMENDED", "MAY", and 76 "OPTIONAL" in this document are to be interpreted as described in RFC 77 2119 [RFC2119]. 79 3. Codec Requirements 81 To ensure a baseline level of interoperability between WebRTC 82 endpoints, a minimum set of required codecs are specified below. If 83 other suitable audio codecs are available for the browser to use, it 84 is RECOMMENDED that they are also be included in the offer in order 85 to maximize the possibility to establish the session without the need 86 for audio transcoding. 88 WebRTC endpoints are REQUIRED to implement the following audio 89 codecs: 91 o Opus [RFC6716] with the payload format specified in 92 [I-D.ietf-payload-rtp-opus]. 94 o G.711 PCMA and PCMU with the payload format specified in section 95 4.5.14 of [RFC3551]. 97 o [RFC3389] comfort noise (CN). Receivers MUST support RFC3389 CN 98 for streams encoded with G.711 or any other supported codec that 99 does not provide its own CN. Since Opus provides its own CN 100 mechanism, the use of RFC3389 CN with Opus is NOT RECOMMENDED. 101 Use of DTX/CN by senders is OPTIONAL. 103 o The audio/telephone-event media format as specified in [RFC4733]. 104 WebRTC endpoints are REQUIRED to be able to generate and consume 105 the following events: 107 +------------+--------------------------------+-----------+ 108 |Event Code | Event Name | Reference | 109 +------------+--------------------------------+-----------+ 110 | 0 | DTMF digit "0" | RFC4733 | 111 | 1 | DTMF digit "1" | RFC4733 | 112 | 2 | DTMF digit "2" | RFC4733 | 113 | 3 | DTMF digit "3" | RFC4733 | 114 | 4 | DTMF digit "4" | RFC4733 | 115 | 5 | DTMF digit "5" | RFC4733 | 116 | 6 | DTMF digit "6" | RFC4733 | 117 | 7 | DTMF digit "7" | RFC4733 | 118 | 8 | DTMF digit "8" | RFC4733 | 119 | 9 | DTMF digit "9" | RFC4733 | 120 | 10 | DTMF digit "*" | RFC4733 | 121 | 11 | DTMF digit "#" | RFC4733 | 122 +------------+--------------------------------+-----------+ 124 For all cases where the endpoint is able to process audio at a 125 sampling rate higher than 8 kHz, it is RECOMMENDED that Opus be 126 offered before PCMA/PCMU. For Opus, all modes MUST be supported on 127 the decoder side. The choice of encoder-side modes is left to the 128 implementer. Endpoints MAY use the offer/answer mechanism to signal 129 a preference for a particular mode or ptime. 131 For additional information on implementing codecs other than the 132 mandatory-to-implement codecs listed above, refer to 133 [I-D.ietf-rtcweb-audio-codecs-for-interop]. 135 4. Audio Level 137 It is desirable to standardize the "on the wire" audio level for 138 speech transmission to avoid users having to manually adjust the 139 playback and to facilitate mixing in conferencing applications. It 140 is also desirable to be consistent with ITU-T recommendations G.169 141 and G.115, which recommend an active audio level of -19 dBm0. 142 However, unlike G.169 and G.115, the audio for WebRTC is not 143 constrained to have a passband specified by G.712 and can in fact be 144 sampled at any sampling rate from 8 kHz to 48 kHz and up. For this 145 reason, the level SHOULD be normalized by only considering 146 frequencies above 300 Hz, regardless of the sampling rate used. The 147 level SHOULD also be adapted to avoid clipping, either by lowering 148 the gain to a level below -19 dBm0, or through the use of a 149 compressor. 151 Assuming 16-bit PCM with a value of +/-32767, -19 dBm0 corresponds to 152 a root mean square (RMS) level of 2600. Only active speech should be 153 considered in the RMS calculation. If the endpoint has control over 154 the entire audio capture path, as is typically the case for a regular 155 phone, then it is RECOMMENDED that the gain be adjusted in such a way 156 that active speech have a level of 2600 (-19 dBm0) for an average 157 speaker. If the endpoint does not have control over the entire audio 158 capture, as is typically the case for a software endpoint, then the 159 endpoint SHOULD use automatic gain control (AGC) to dynamically 160 adjust the level to 2600 (-19 dBm0) +/- 6 dB. For music or desktop 161 sharing applications, the level SHOULD NOT be automatically adjusted 162 and the endpoint SHOULD allow the user to set the gain manually. 164 The RECOMMENDED filter for normalizing the signal energy is a second- 165 order Butterworth filter with a 300 Hz cutoff frequency. 167 It is common for the audio output on some devices to be "calibrated" 168 for playing back pre-recorded "commercial" music, which is typically 169 around 12 dB louder than the level recommended in this section. 170 Because of this, endpoints MAY increase the gain before playback. 172 5. Acoustic Echo Cancellation (AEC) 174 It is plausible that the dominant near to mid-term WebRTC usage model 175 will be people using the interactive audio and video capabilities to 176 communicate with each other via web browsers running on a notebook 177 computer that has built-in microphone and speakers. The notebook-as- 178 communication-device paradigm presents challenging echo cancellation 179 problems, the specific remedy of which will not be mandated here. 180 However, while no specific algorithm or standard will be required by 181 WebRTC compatible endpoints, echo cancellation will improve the user 182 experience and should be implemented by the endpoint device. 184 WebRTC endpoints SHOULD include an AEC or some other form of echo 185 control. On general purpose platforms (e.g. PC), it is common for 186 the audio capture ADC and the audio playback DAC to use different 187 clocks. In these cases, such as when a webcam is used for capture 188 and a separate soundcard is used for playback, the sampling rates are 189 likely to differ slightly. Endpoint AECs SHOULD be robust to such 190 conditions, unless they are shipped along with hardware that 191 guarantees capture and playback to be sampled from the same clock. 193 Endpoints SHOULD allow the entire AEC and/or the non-linear 194 processing (NLP) to be turned off for applications, such as music, 195 that do not behave well with the spectral attenuation methods 196 typically used in NLPs. Similarly, endpoints SHOULD have the ability 197 to detect the presence of a headset and disable echo cancellation. 199 For some applications where the remote endpoint may not have an echo 200 canceller, the local endpoint MAY include a far-end echo canceller, 201 but if that is the case, it SHOULD be disabled by default. 203 6. Legacy VoIP Interoperability 205 The codec requirements above will ensure, at a minimum, voice 206 interoperability capabilities between WebRTC endpoints applications 207 and legacy phone systems that support G.711. 209 7. IANA Considerations 211 This document makes no request of IANA. 213 Note to RFC Editor: this section may be removed on publication as an 214 RFC. 216 8. Security Considerations 218 For security considerations regarding the codecs themselves please 219 refer their specifications, including [RFC6716], 220 [I-D.ietf-payload-rtp-opus], [RFC3551], [RFC3389], and [RFC4733]. 221 Likewise, consult the RTP base specification for security RTP-based 222 security considerations. WebRTC security is further discussed in 223 [I-D.ietf-rtcweb-security] and [I-D.ietf-rtcweb-security-arch] and 224 [I-D.ietf-rtcweb-rtp-usage]. 226 Implementers should consider whether the use of VBR is appropriate 227 for their application based on [RFC6562]. Encryption and 228 authentication issues are beyond the scope of this document. 230 9. Acknowledgements 232 This draft incorporates ideas and text from various other drafts. In 233 particularly we would like to acknowledge, and say thanks for, work 234 we incorporated from Harald Alvestrand and Cullen Jennings. 236 10. References 237 10.1. Normative References 239 [RFC2119] Bradner, S., "Key words for use in RFCs to Indicate 240 Requirement Levels", BCP 14, RFC 2119, 241 DOI 10.17487/RFC2119, March 1997, 242 . 244 [RFC3551] Schulzrinne, H. and S. Casner, "RTP Profile for Audio and 245 Video Conferences with Minimal Control", STD 65, RFC 3551, 246 DOI 10.17487/RFC3551, July 2003, 247 . 249 [RFC3389] Zopf, R., "Real-time Transport Protocol (RTP) Payload for 250 Comfort Noise (CN)", RFC 3389, DOI 10.17487/RFC3389, 251 September 2002, . 253 [RFC4733] Schulzrinne, H. and T. Taylor, "RTP Payload for DTMF 254 Digits, Telephony Tones, and Telephony Signals", RFC 4733, 255 DOI 10.17487/RFC4733, December 2006, 256 . 258 [RFC6716] Valin, JM., Vos, K., and T. Terriberry, "Definition of the 259 Opus Audio Codec", RFC 6716, DOI 10.17487/RFC6716, 260 September 2012, . 262 [RFC6562] Perkins, C. and JM. Valin, "Guidelines for the Use of 263 Variable Bit Rate Audio with Secure RTP", RFC 6562, 264 DOI 10.17487/RFC6562, March 2012, 265 . 267 [I-D.ietf-payload-rtp-opus] 268 Spittka, J., Vos, K., and J. Valin, "RTP Payload Format 269 for the Opus Speech and Audio Codec", draft-ietf-payload- 270 rtp-opus-11 (work in progress), April 2015. 272 10.2. Informative References 274 [I-D.ietf-rtcweb-security] 275 Rescorla, E., "Security Considerations for WebRTC", draft- 276 ietf-rtcweb-security-08 (work in progress), February 2015. 278 [I-D.ietf-rtcweb-security-arch] 279 Rescorla, E., "WebRTC Security Architecture", draft-ietf- 280 rtcweb-security-arch-11 (work in progress), March 2015. 282 [I-D.ietf-rtcweb-rtp-usage] 283 Perkins, C., Westerlund, M., and J. Ott, "Web Real-Time 284 Communication (WebRTC): Media Transport and Use of RTP", 285 draft-ietf-rtcweb-rtp-usage-23 (work in progress), March 286 2015. 288 [I-D.ietf-rtcweb-audio-codecs-for-interop] 289 Proust, S., Berger, E., Feiten, B., Burman, B., Bogineni, 290 K., Lei, M., and E. Marocco, "Additional WebRTC audio 291 codecs for interoperability.", draft-ietf-rtcweb-audio- 292 codecs-for-interop-01 (work in progress), January 2015. 294 Authors' Addresses 296 Jean-Marc Valin 297 Mozilla 298 331 E. Evelyn Avenue 299 Mountain View, CA 94041 300 USA 302 Email: jmvalin@jmvalin.ca 304 Cary Bran 305 Plantronics 306 345 Encinial Street 307 Santa Cruz, CA 95060 308 USA 310 Phone: +1 206 661-2398 311 Email: cary.bran@plantronics.com