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Bran 5 Expires: April 27, 2015 Plantronics 6 October 24, 2014 8 WebRTC Audio Codec and Processing Requirements 9 draft-ietf-rtcweb-audio-07 11 Abstract 13 This document outlines the audio codec and processing requirements 14 for WebRTC client application and endpoint devices. 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 April 27, 2015. 33 Copyright Notice 35 Copyright (c) 2014 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 . . . . . . . . . . . . . . . . . . 5 61 10.2. Informative References . . . . . . . . . . . . . . . . . 6 62 Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 6 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 client 70 implementations. 72 2. Terminology 74 The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT", 75 "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this 76 document are to be interpreted as described in RFC 2119 [RFC2119]. 78 3. Codec Requirements 80 To ensure a baseline level of interoperability between WebRTC 81 clients, a minimum set of required codecs are specified below. If 82 other suitable audio codecs are available for the browser to use, it 83 is RECOMMENDED that they are also be included in the offer in order 84 to maximize the possibility to establish the session without the need 85 for audio transcoding. 87 WebRTC clients are REQUIRED to implement the following audio codecs: 89 o Opus [RFC6716] with the payload format specified in 90 [I-D.ietf-payload-rtp-opus]. 92 o G.711 PCMA and PCMU with the payload format specified in section 93 4.5.14 of [RFC3551]. 95 o [RFC3389] comfort noise (CN). Receivers MUST support RFC3389 CN 96 for streams encoded with G.711 or any other supported codec that 97 does not provide its own CN. Since Opus provides its own CN 98 mechanism, the use of RFC3389 CN with Opus is NOT RECOMMENDED. 99 Use of DTX/CN by senders is OPTIONAL. 101 o The audio/telephone-event media format as specified in [RFC4733]. 102 WebRTC clients are REQUIRED to be able to generate and consume the 103 following events: 105 +------------+--------------------------------+-----------+ 106 |Event Code | Event Name | Reference | 107 +------------+--------------------------------+-----------+ 108 | 0 | DTMF digit "0" | RFC4733 | 109 | 1 | DTMF digit "1" | RFC4733 | 110 | 2 | DTMF digit "2" | RFC4733 | 111 | 3 | DTMF digit "3" | RFC4733 | 112 | 4 | DTMF digit "4" | RFC4733 | 113 | 5 | DTMF digit "5" | RFC4733 | 114 | 6 | DTMF digit "6" | RFC4733 | 115 | 7 | DTMF digit "7" | RFC4733 | 116 | 8 | DTMF digit "8" | RFC4733 | 117 | 9 | DTMF digit "9" | RFC4733 | 118 | 10 | DTMF digit "*" | RFC4733 | 119 | 11 | DTMF digit "#" | RFC4733 | 120 +------------+--------------------------------+-----------+ 122 For all cases where the client is able to process audio at a sampling 123 rate higher than 8 kHz, it is RECOMMENDED that Opus be offered before 124 PCMA/PCMU. For Opus, all modes MUST be supported on the decoder 125 side. The choice of encoder-side modes is left to the implementer. 126 Clients MAY use the offer/answer mechanism to signal a preference for 127 a particular mode or ptime. 129 For additional information on implementing codecs other than the 130 mandatory-to-implement codecs listed above, refer to 131 [I-D.ietf-rtcweb-audio-codecs-for-interop]. 133 4. Audio Level 135 It is desirable to standardize the "on the wire" audio level for 136 speech transmission to avoid users having to manually adjust the 137 playback and to facilitate mixing in conferencing applications. It 138 is also desirable to be consistent with ITU-T recommendations G.169 139 and G.115, which recommend an active audio level of -19 dBm0. 140 However, unlike G.169 and G.115, the audio for WebRTC is not 141 constrained to have a passband specified by G.712 and can in fact be 142 sampled at any sampling rate from 8 kHz to 48 kHz and up. For this 143 reason, the level SHOULD be normalized by only considering 144 frequencies above 300 Hz, regardless of the sampling rate used. The 145 level SHOULD also be adapted to avoid clipping, either by lowering 146 the gain to a level below -19 dBm0, or through the use of a 147 compressor. 149 Assuming 16-bit PCM with a value of +/-32767, -19 dBm0 corresponds to 150 a root mean square (RMS) level of 2600. Only active speech should be 151 considered in the RMS calculation. If the client has control over 152 the entire audio capture path, as is typically the case for a regular 153 phone, then it is RECOMMENDED that the gain be adjusted in such a way 154 that active speech have a level of 2600 (-19 dBm0) for an average 155 speaker. If the client does not have control over the entire audio 156 capture, as is typically the case for a software client, then the 157 client SHOULD use automatic gain control (AGC) to dynamically adjust 158 the level to 2600 (-19 dBm0) +/- 6 dB. For music or desktop sharing 159 applications, the level SHOULD NOT be automatically adjusted and the 160 client SHOULD allow the user to set the gain manually. 162 The RECOMMENDED filter for normalizing the signal energy is a second- 163 order Butterworth filter with a 300 Hz cutoff frequency. 165 It is common for the audio output on some devices to be "calibrated" 166 for playing back pre-recorded "commercial" music, which is typically 167 around 12 dB louder than the level recommended in this section. 168 Because of this, clients MAY increase the gain before playback. 170 5. Acoustic Echo Cancellation (AEC) 172 It is plausible that the dominant near to mid-term WebRTC usage model 173 will be people using the interactive audio and video capabilities to 174 communicate with each other via web browsers running on a notebook 175 computer that has built-in microphone and speakers. The notebook-as- 176 communication-device paradigm presents challenging echo cancellation 177 problems, the specific remedy of which will not be mandated here. 178 However, while no specific algorithm or standard will be required by 179 WebRTC compatible clients, echo cancellation will improve the user 180 experience and should be implemented by the endpoint device. 182 WebRTC clients SHOULD include an AEC or some other form of echo 183 control and if that is not possible, the clients SHOULD ensure that 184 the speaker-to-microphone gain is below unity at all frequencies to 185 avoid instability when none of the client has echo control. For 186 clients that do not control the audio capture and playback hardware, 187 it is RECOMMENDED to support echo cancellation between devices 188 running at slightly different sampling rates, such as when a webcam 189 is used for microphone. 191 Clients SHOULD allow the entire AEC and/or the non-linear processing 192 (NLP) to be turned off for applications, such as music, that do not 193 behave well with the spectral attenuation methods typically used in 194 NLPs. Similarly, clients SHOULD have the ability to detect the 195 presence of a headset and disable echo cancellation. 197 For some applications where the remote client may not have an echo 198 canceller, the local client MAY include a far-end echo canceller, but 199 if that is the case, it SHOULD be disabled by default. 201 6. Legacy VoIP Interoperability 203 The codec requirements above will ensure, at a minimum, voice 204 interoperability capabilities between WebRTC client applications and 205 legacy phone systems that support G.711. 207 7. IANA Considerations 209 This document makes no request of IANA. 211 Note to RFC Editor: this section may be removed on publication as an 212 RFC. 214 8. Security Considerations 216 Implementers should consider whether the use of VBR is appropriate 217 for their application based on [RFC6562]. Encryption and 218 authentication issues are beyond the scope of this document. 220 9. Acknowledgements 222 This draft incorporates ideas and text from various other drafts. In 223 particularly we would like to acknowledge, and say thanks for, work 224 we incorporated from Harald Alvestrand and Cullen Jennings. 226 10. References 228 10.1. Normative References 230 [RFC2119] Bradner, S., "Key words for use in RFCs to Indicate 231 Requirement Levels", BCP 14, RFC 2119, March 1997. 233 [RFC3551] Schulzrinne, H. and S. Casner, "RTP Profile for Audio and 234 Video Conferences with Minimal Control", STD 65, RFC 3551, 235 July 2003. 237 [RFC3389] Zopf, R., "Real-time Transport Protocol (RTP) Payload for 238 Comfort Noise (CN)", RFC 3389, September 2002. 240 [RFC4733] Schulzrinne, H. and T. Taylor, "RTP Payload for DTMF 241 Digits, Telephony Tones, and Telephony Signals", RFC 4733, 242 December 2006. 244 [RFC6716] Valin, JM., Vos, K., and T. Terriberry, "Definition of the 245 Opus Audio Codec", RFC 6716, September 2012. 247 [RFC6562] Perkins, C. and JM. Valin, "Guidelines for the Use of 248 Variable Bit Rate Audio with Secure RTP", RFC 6562, March 249 2012. 251 [I-D.ietf-payload-rtp-opus] 252 Spittka, J., Vos, K., and J. Valin, "RTP Payload Format 253 for Opus Speech and Audio Codec", draft-ietf-payload-rtp- 254 opus-03 (work in progress), July 2014. 256 10.2. Informative References 258 [I-D.ietf-rtcweb-audio-codecs-for-interop] 259 Proust, S., Berger, E., Feiten, B., Bogineni, K., Lei, M., 260 and E. Marocco, "Additional WebRTC audio codecs for 261 interoperability with legacy networks.", draft-ietf- 262 rtcweb-audio-codecs-for-interop-00 (work in progress), 263 September 2014. 265 Authors' Addresses 267 Jean-Marc Valin 268 Mozilla 269 331 E. Evelyn Avenue 270 Mountain View, CA 94041 271 USA 273 Email: jmvalin@jmvalin.ca 275 Cary Bran 276 Plantronics 277 345 Encinial Street 278 Santa Cruz, CA 95060 279 USA 281 Phone: +1 206 661-2398 282 Email: cary.bran@plantronics.com