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Bran 5 Expires: April 18, 2014 Plantronics 6 October 15, 2013 8 WebRTC Audio Codec and Processing Requirements 9 draft-ietf-rtcweb-audio-03 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 18, 2014. 33 Copyright Notice 35 Copyright (c) 2013 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 . . . . . . . . . . . . . . . . . . . . . 4 56 8. Security Considerations . . . . . . . . . . . . . . . . . . . 5 57 9. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . 5 58 10. Normative References . . . . . . . . . . . . . . . . . . . . 5 59 Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 5 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 a 90 ptime of 20 - see section 4.5.14 of [RFC3551] 92 o Telephone Event - [RFC4733] 94 For all cases where the client is able to process audio at a sampling 95 rate higher than 8 kHz, it is RECOMMENDED that Opus be offered before 96 PCMA/PCMU. For Opus, all modes MUST be supported on the decoder 97 side. The choice of encoder-side modes is left to the implementer. 98 Clients MAY use the offer/answer mechanism to signal a preference for 99 a particular mode or ptime. 101 4. Audio Level 103 It is desirable to standardize the "on the wire" audio level for 104 speech transmission to avoid users having to manually adjust the 105 playback and to facilitate mixing in conferencing applications. It 106 is also desirable to be consistent with ITU-T recommendations G.169 107 and G.115, which recommend an active audio level of -19 dBm0. 108 However, unlike G.169 and G.115, the audio for WebRTC is not 109 constrained to have a passband specified by G.712 and can in fact be 110 sampled at any sampling rate from 8 kHz to 48 kHz and up. For this 111 reason, the level SHOULD be normalized by only considering 112 frequencies above 300 Hz, regardless of the sampling rate used. The 113 level SHOULD also be adapted to avoid clipping, either by lowering 114 the gain to a level below -19 dBm0, or through the use of a 115 compressor. 117 AUTHORS' NOTE: The idea of using the same level as what the ITU-T 118 recommends is that it should improve inter-operability while at the 119 same time maintaining sufficient dynamic range and reducing the risk 120 of clipping. The main drawbacks are that the resulting level is 121 about 12 dB lower than typical "commercial music" levels and it 122 leaves room for ill-behaved clients to be much louder than a normal 123 client. While using music-type levels is not really an option (it 124 would require using the same compressor-limitors that studios use), 125 it would be possible to have a level slightly higher (e.g. 3 dB) 126 than what is recommended above without causing interoperability 127 problems. 129 Assuming 16-bit PCM with a value of +/-32767, -19 dBm0 corresponds to 130 a root mean square (RMS) level of 2600. Only active speech should be 131 considered in the RMS calculation. If the client has control over 132 the entire audio capture path, as is typically the case for a regular 133 phone, then it is RECOMMENDED that the gain be adjusted in such a way 134 that active speech have a level of 2600 (-19 dBm0) for an average 135 speaker. If the client does not have control over the entire audio 136 capture, as is typically the case for a software client, then the 137 client SHOULD use automatic gain control (AGC) to dynamically adjust 138 the level to 2600 (-19 dBm0) +/- 6 dB. For music or desktop sharing 139 applications, the level SHOULD NOT be automatically adjusted and the 140 client SHOULD allow the user to set the gain manually. 142 The RECOMMENDED filter for normalizing the signal energy is a second- 143 order Butterworth filter with a 300 Hz cutoff frequency. 145 It is common for the audio output on some devices to be "calibrated" 146 for playing back pre-recorded "commercial" music, which is typically 147 around 12 dB louder than the level recommended in this section. 148 Because of this, clients MAY increase the gain before playback. 150 5. Acoustic Echo Cancellation (AEC) 152 It is plausible that the dominant near to mid-term WebRTC usage model 153 will be people using the interactive audio and video capabilities to 154 communicate with each other via web browsers running on a notebook 155 computer that has built-in microphone and speakers. The notebook-as- 156 communication-device paradigm presents challenging echo cancellation 157 problems, the specific remedy of which will not be mandated here. 158 However, while no specific algorithm or standard will be required by 159 WebRTC compatible clients, echo cancellation will improve the user 160 experience and should be implemented by the endpoint device. 162 WebRTC clients SHOULD include an AEC and if that is not possible, the 163 clients SHOULD ensure that the speaker-to-microphone gain is below 164 unity at all frequencies to avoid instability when none of the client 165 has echo cancellation. For clients that do not control the audio 166 capture and playback devices directly, it is RECOMMENDED to support 167 echo cancellation between devices running at slight different 168 sampling rates, such as when a webcam is used for microphone. 170 The client SHOULD allow either the entire AEC or the non-linear 171 processing (NLP) to be turned off for applications, such as music, 172 that do not behave well with the spectral attenuation methods 173 typically used in NLPs. It SHOULD have the ability to detect the 174 presence of a headset and disable echo cancellation. 176 For some applications where the remote client may not have an echo 177 canceller, the local client MAY include a far-end echo canceller, but 178 if that is the case, it SHOULD be disabled by default. 180 6. Legacy VoIP Interoperability 182 The codec requirements above will ensure, at a minimum, voice 183 interoperability capabilities between WebRTC client applications and 184 legacy phone systems. 186 7. IANA Considerations 188 This document makes no request of IANA. 190 Note to RFC Editor: this section may be removed on publication as an 191 RFC. 193 8. Security Considerations 195 The codec requirements have no additional security considerations 196 other than those captured in 197 [I-D.ekr-security-considerations-for-rtc-web]. 199 9. Acknowledgements 201 This draft incorporates ideas and text from various other drafts. In 202 particularly we would like to acknowledge, and say thanks for, work 203 we incorporated from Harald Alvestrand and Cullen Jennings. 205 10. Normative References 207 [RFC2119] Bradner, S., "Key words for use in RFCs to Indicate 208 Requirement Levels", BCP 14, RFC 2119, March 1997. 210 [RFC3551] Schulzrinne, H. and S. Casner, "RTP Profile for Audio and 211 Video Conferences with Minimal Control", STD 65, RFC 3551, 212 July 2003. 214 [RFC4733] Schulzrinne, H. and T. Taylor, "RTP Payload for DTMF 215 Digits, Telephony Tones, and Telephony Signals", RFC 4733, 216 December 2006. 218 [RFC6716] Valin, JM., Vos, K., and T. Terriberry, "Definition of the 219 Opus Audio Codec", RFC 6716, September 2012. 221 [Opus-RTP] 222 Spittka, J., Vos, K., and JM. Valin, "RTP Payload Format 223 for Opus Codec", August 2013. 225 [I-D.ekr-security-considerations-for-rtc-web] 226 Rescorla, E.K., "Security Considerations for RTC-Web", May 227 2011. 229 Authors' Addresses 231 Jean-Marc Valin 232 Mozilla 233 650 Castro Street 234 Mountain View, CA 94041 235 USA 237 Email: jmvalin@jmvalin.ca 238 Cary Bran 239 Plantronics 240 345 Encinial Street 241 Santa Cruz, CA 95060 242 USA 244 Phone: +1 206 661-2398 245 Email: cary.bran@plantronics.com