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Bran 5 Expires: July 31, 2014 Plantronics 6 January 27, 2014 8 WebRTC Audio Codec and Processing Requirements 9 draft-ietf-rtcweb-audio-04 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 July 31, 2014. 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 49 1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 2 50 2. Terminology . . . . . . . . . . . . . . . . . . . . . . . . . 2 51 3. Codec Requirements . . . . . . . . . . . . . . . . . . . . . 2 52 4. Audio Level . . . . . . . . . . . . . . . . . . . . . . . . . 3 53 5. Acoustic Echo Cancellation (AEC) . . . . . . . . . . . . . . 4 54 6. Legacy VoIP Interoperability . . . . . . . . . . . . . . . . 4 55 7. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 5 56 8. Security Considerations . . . . . . . . . . . . . . . . . . . 5 57 9. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . 5 58 10. Normative References . . . . . . . . . . . . . . . . . . . . 5 59 Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 6 61 1. Introduction 63 An integral part of the success and adoption of the Web Real Time 64 Communications (WebRTC) will be the voice and video interoperability 65 between WebRTC applications. This specification will outline the 66 audio processing and codec requirements for WebRTC client 67 implementations. 69 2. Terminology 71 The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT", 72 "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this 73 document are to be interpreted as described in RFC 2119 [RFC2119]. 75 3. Codec Requirements 77 To ensure a baseline level of interoperability between WebRTC 78 clients, a minimum set of required codecs are specified below. If 79 other suitable audio codecs are available for the browser to use, it 80 is RECOMMENDED that they are also be included in the offer in order 81 to maximize the possibility to establish the session without the need 82 for audio transcoding. 84 WebRTC clients are REQUIRED to implement the following audio codecs. 86 o Opus [RFC6716], with the payload format specified in [Opus-RTP] 87 and any ptime value up to 120 ms 89 o G.711 PCMA and PCMU with one channel, a rate of 8000 Hz and any 90 ptime value up to 120 ms - see section 4.5.14 of [RFC3551] 92 o The audio/telephone-event media format as specified in [RFC4733]. 93 WebRTC clients are REQUIRED to be able to generate and consume the 94 following events: 96 +------------+--------------------------------+-----------+ 97 |Event Code | Event Name | Reference | 98 +------------+--------------------------------+-----------+ 99 | 0 | DTMF digit "0" | RFC4733 | 100 | 1 | DTMF digit "1" | RFC4733 | 101 | 2 | DTMF digit "2" | RFC4733 | 102 | 3 | DTMF digit "3" | RFC4733 | 103 | 4 | DTMF digit "4" | RFC4733 | 104 | 5 | DTMF digit "5" | RFC4733 | 105 | 6 | DTMF digit "6" | RFC4733 | 106 | 7 | DTMF digit "7" | RFC4733 | 107 | 8 | DTMF digit "8" | RFC4733 | 108 | 9 | DTMF digit "9" | RFC4733 | 109 | 10 | DTMF digit "*" | RFC4733 | 110 | 11 | DTMF digit "#" | RFC4733 | 111 +------------+--------------------------------+-----------+ 113 For all cases where the client is able to process audio at a sampling 114 rate higher than 8 kHz, it is RECOMMENDED that Opus be offered before 115 PCMA/PCMU. For Opus, all modes MUST be supported on the decoder 116 side. The choice of encoder-side modes is left to the implementer. 117 Clients MAY use the offer/answer mechanism to signal a preference for 118 a particular mode or ptime. 120 4. Audio Level 122 It is desirable to standardize the "on the wire" audio level for 123 speech transmission to avoid users having to manually adjust the 124 playback and to facilitate mixing in conferencing applications. It 125 is also desirable to be consistent with ITU-T recommendations G.169 126 and G.115, which recommend an active audio level of -19 dBm0. 127 However, unlike G.169 and G.115, the audio for WebRTC is not 128 constrained to have a passband specified by G.712 and can in fact be 129 sampled at any sampling rate from 8 kHz to 48 kHz and up. For this 130 reason, the level SHOULD be normalized by only considering 131 frequencies above 300 Hz, regardless of the sampling rate used. The 132 level SHOULD also be adapted to avoid clipping, either by lowering 133 the gain to a level below -19 dBm0, or through the use of a 134 compressor. 136 Assuming 16-bit PCM with a value of +/-32767, -19 dBm0 corresponds to 137 a root mean square (RMS) level of 2600. Only active speech should be 138 considered in the RMS calculation. If the client has control over 139 the entire audio capture path, as is typically the case for a regular 140 phone, then it is RECOMMENDED that the gain be adjusted in such a way 141 that active speech have a level of 2600 (-19 dBm0) for an average 142 speaker. If the client does not have control over the entire audio 143 capture, as is typically the case for a software client, then the 144 client SHOULD use automatic gain control (AGC) to dynamically adjust 145 the level to 2600 (-19 dBm0) +/- 6 dB. For music or desktop sharing 146 applications, the level SHOULD NOT be automatically adjusted and the 147 client SHOULD allow the user to set the gain manually. 149 The RECOMMENDED filter for normalizing the signal energy is a second- 150 order Butterworth filter with a 300 Hz cutoff frequency. 152 It is common for the audio output on some devices to be "calibrated" 153 for playing back pre-recorded "commercial" music, which is typically 154 around 12 dB louder than the level recommended in this section. 155 Because of this, clients MAY increase the gain before playback. 157 5. Acoustic Echo Cancellation (AEC) 159 It is plausible that the dominant near to mid-term WebRTC usage model 160 will be people using the interactive audio and video capabilities to 161 communicate with each other via web browsers running on a notebook 162 computer that has built-in microphone and speakers. The notebook-as- 163 communication-device paradigm presents challenging echo cancellation 164 problems, the specific remedy of which will not be mandated here. 165 However, while no specific algorithm or standard will be required by 166 WebRTC compatible clients, echo cancellation will improve the user 167 experience and should be implemented by the endpoint device. 169 WebRTC clients SHOULD include an AEC or some other form of echo 170 control and if that is not possible, the clients SHOULD ensure that 171 the speaker-to-microphone gain is below unity at all frequencies to 172 avoid instability when none of the client has echo control. For 173 clients that do not control the audio capture and playback hardware, 174 it is RECOMMENDED to support echo cancellation between devices 175 running at slightly different sampling rates, such as when a webcam 176 is used for microphone. 178 Clients SHOULD allow the entire AEC and/or the non-linear processing 179 (NLP) to be turned off for applications, such as music, that do not 180 behave well with the spectral attenuation methods typically used in 181 NLPs. Similarly, clients SHOULD have the ability to detect the 182 presence of a headset and disable echo cancellation. 184 For some applications where the remote client may not have an echo 185 canceller, the local client MAY include a far-end echo canceller, but 186 if that is the case, it SHOULD be disabled by default. 188 6. Legacy VoIP Interoperability 189 The codec requirements above will ensure, at a minimum, voice 190 interoperability capabilities between WebRTC client applications and 191 legacy phone systems. 193 7. IANA Considerations 195 This document makes no request of IANA. 197 Note to RFC Editor: this section may be removed on publication as an 198 RFC. 200 8. Security Considerations 202 Implementers should consider whether the use of VBR is appropriate 203 for their application based on [RFC6562]. Encryption and 204 authentication issues are beyond the scope of this document. 206 9. Acknowledgements 208 This draft incorporates ideas and text from various other drafts. In 209 particularly we would like to acknowledge, and say thanks for, work 210 we incorporated from Harald Alvestrand and Cullen Jennings. 212 10. Normative References 214 [RFC2119] Bradner, S., "Key words for use in RFCs to Indicate 215 Requirement Levels", BCP 14, RFC 2119, March 1997. 217 [RFC3551] Schulzrinne, H. and S. Casner, "RTP Profile for Audio and 218 Video Conferences with Minimal Control", STD 65, RFC 3551, 219 July 2003. 221 [RFC4733] Schulzrinne, H. and T. Taylor, "RTP Payload for DTMF 222 Digits, Telephony Tones, and Telephony Signals", RFC 4733, 223 December 2006. 225 [RFC6716] Valin, JM., Vos, K., and T. Terriberry, "Definition of the 226 Opus Audio Codec", RFC 6716, September 2012. 228 [RFC6562] Perkins, C. and JM. Valin, "Guidelines for the Use of 229 Variable Bit Rate Audio with Secure RTP", RFC 6562, March 230 2012. 232 [Opus-RTP] 233 Spittka, J., Vos, K., and JM. Valin, "RTP Payload Format 234 for Opus Codec", August 2013. 236 Authors' Addresses 238 Jean-Marc Valin 239 Mozilla 240 650 Castro Street 241 Mountain View, CA 94041 242 USA 244 Email: jmvalin@jmvalin.ca 246 Cary Bran 247 Plantronics 248 345 Encinial Street 249 Santa Cruz, CA 95060 250 USA 252 Phone: +1 206 661-2398 253 Email: cary.bran@plantronics.com