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Internet Engineering Task Force (IETF)                         JM. ValinRequest for Comments: 7874                                       MozillaCategory: Standards Track                                        C. BranISSN: 2070-1721                                              Plantronics                                                                May 2016WebRTC Audio Codec and Processing RequirementsAbstract   This document outlines the audio codec and processing requirements   for WebRTC endpoints.Status of This Memo   This is an Internet Standards Track document.   This document is a product of the Internet Engineering Task Force   (IETF).  It represents the consensus of the IETF community.  It has   received public review and has been approved for publication by the   Internet Engineering Steering Group (IESG).  Further information on   Internet Standards is available inSection 2 of RFC 5741.   Information about the current status of this document, any errata,   and how to provide feedback on it may be obtained athttp://www.rfc-editor.org/info/rfc7874.Copyright Notice   Copyright (c) 2016 IETF Trust and the persons identified as the   document authors.  All rights reserved.   This document is subject toBCP 78 and the IETF Trust's Legal   Provisions Relating to IETF Documents   (http://trustee.ietf.org/license-info) in effect on the date of   publication of this document.  Please review these documents   carefully, as they describe your rights and restrictions with respect   to this document.  Code Components extracted from this document must   include Simplified BSD License text as described in Section 4.e of   the Trust Legal Provisions and are provided without warranty as   described in the Simplified BSD License.Valin & Bran                 Standards Track                    [Page 1]

RFC 7874                      WebRTC Audio                      May 2016Table of Contents1.  Introduction  . . . . . . . . . . . . . . . . . . . . . . . .22.  Terminology . . . . . . . . . . . . . . . . . . . . . . . . .23.  Codec Requirements  . . . . . . . . . . . . . . . . . . . . .24.  Audio Level . . . . . . . . . . . . . . . . . . . . . . . . .45.  Acoustic Echo Cancellation (AEC)  . . . . . . . . . . . . . .46.  Legacy VoIP Interoperability  . . . . . . . . . . . . . . . .57.  Security Considerations . . . . . . . . . . . . . . . . . . .58.  References  . . . . . . . . . . . . . . . . . . . . . . . . .68.1.  Normative References  . . . . . . . . . . . . . . . . . .68.2.  Informative References  . . . . . . . . . . . . . . . . .6   Acknowledgements  . . . . . . . . . . . . . . . . . . . . . . . .7   Authors' Addresses  . . . . . . . . . . . . . . . . . . . . . . .71.  Introduction   An integral part of the success and adoption of Web Real-Time   Communications (WebRTC) will be the voice and video interoperability   between WebRTC applications.  This specification will outline the   audio processing and codec requirements for WebRTC endpoints.2.  Terminology   The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",   "SHOULD", "SHOULD NOT", "RECOMMENDED", "NOT RECOMMENDED", "MAY", and   "OPTIONAL" in this document are to be interpreted as described inRFC2119 [RFC2119].3.  Codec Requirements   To ensure a baseline level of interoperability between WebRTC   endpoints, a minimum set of required codecs are specified below.  If   other suitable audio codecs are available for the WebRTC endpoint to   use, it is RECOMMENDED that they also be included in the offer in   order to maximize the possibility of establishing the session without   the need for audio transcoding.   WebRTC endpoints are REQUIRED to implement the following audio   codecs:   o  Opus [RFC6716] with the payload format specified in [RFC7587].   o  PCMA and PCMU (as specified in ITU-T Recommendation G.711 [G.711])      with the payload format specified inSection 4.5.14 of [RFC3551].Valin & Bran                 Standards Track                    [Page 2]

RFC 7874                      WebRTC Audio                      May 2016   o  [RFC3389] comfort noise (CN).  WebRTC endpoints MUST support      [RFC3389] CN for streams encoded with G.711 or any other supported      codec that does not provide its own CN.  Since Opus provides its      own CN mechanism, the use of [RFC3389] CN with Opus is NOT      RECOMMENDED.  Use of Discontinuous Transmission (DTX) / CN by      senders is OPTIONAL.   o  the 'audio/telephone-event' media type as specified in [RFC4733].      The endpoints MAY send DTMF events at any time and SHOULD suppress      in-band dual-tone multi-frequency (DTMF) tones, if any.  DTMF      events generated by a WebRTC endpoint MUST have a duration of no      more than 8000 ms and no less than 40 ms.  The recommended default      duration is 100 ms for each tone.  The gap between events MUST be      no less than 30 ms; the recommended default gap duration is 70 ms.      WebRTC endpoints are not required to do anything with tones (as      specified inRFC 4733) sent to them, except gracefully drop them.      There is currently no API to inform JavaScript about the received      DTMF or other tones (as specified inRFC 4733).  WebRTC endpoints      are REQUIRED to be able to generate and consume the following      events:         +------------+--------------------------------+-----------+         |Event Code  | Event Name                     | Reference |         +------------+--------------------------------+-----------+         | 0          | DTMF digit "0"                 | [RFC4733] |         | 1          | DTMF digit "1"                 | [RFC4733] |         | 2          | DTMF digit "2"                 | [RFC4733] |         | 3          | DTMF digit "3"                 | [RFC4733] |         | 4          | DTMF digit "4"                 | [RFC4733] |         | 5          | DTMF digit "5"                 | [RFC4733] |         | 6          | DTMF digit "6"                 | [RFC4733] |         | 7          | DTMF digit "7"                 | [RFC4733] |         | 8          | DTMF digit "8"                 | [RFC4733] |         | 9          | DTMF digit "9"                 | [RFC4733] |         | 10         | DTMF digit "*"                 | [RFC4733] |         | 11         | DTMF digit "#"                 | [RFC4733] |         | 12         | DTMF digit "A"                 | [RFC4733] |         | 13         | DTMF digit "B"                 | [RFC4733] |         | 14         | DTMF digit "C"                 | [RFC4733] |         | 15         | DTMF digit "D"                 | [RFC4733] |         +------------+--------------------------------+-----------+   For all cases where the endpoint is able to process audio at a   sampling rate higher than 8 kHz, it is RECOMMENDED that Opus be   offered before PCMA/PCMU.  For Opus, all modes MUST be supported on   the decoder side.  The choice of encoder-side modes is left to the   implementer.  Endpoints MAY use the offer/answer mechanism to signal   a preference for a particular mode or ptime.Valin & Bran                 Standards Track                    [Page 3]

RFC 7874                      WebRTC Audio                      May 2016   For additional information on implementing codecs other than the   mandatory-to-implement codecs listed above, refer to [RFC7875].4.  Audio Level   It is desirable to standardize the "on the wire" audio level for   speech transmission to avoid users having to manually adjust the   playback and to facilitate mixing in conferencing applications.  It   is also desirable to be consistent with ITU-T Recommendations G.169   and G.115, which recommend an active audio level of -19 dBm0.   However, unlike G.169 and G.115, the audio for WebRTC is not   constrained to have a passband specified by G.712 and can in fact be   sampled at any sampling rate from 8 to 48 kHz and higher.  For this   reason, the level SHOULD be normalized by only considering   frequencies above 300 Hz, regardless of the sampling rate used.  The   level SHOULD also be adapted to avoid clipping, either by lowering   the gain to a level below -19 dBm0 or through the use of a   compressor.   Assuming linear 16-bit PCM with a value of +/-32767, -19 dBm0   corresponds to a root mean square (RMS) level of 2600.  Only active   speech should be considered in the RMS calculation.  If the endpoint   has control over the entire audio-capture path, as is typically the   case for a regular phone, then it is RECOMMENDED that the gain be   adjusted in such a way that an average speaker would have a level of   2600 (-19 dBm0) for active speech.  If the endpoint does not have   control over the entire audio capture, as is typically the case for a   software endpoint, then the endpoint SHOULD use automatic gain   control (AGC) to dynamically adjust the level to 2600 (-19 dBm0) +/-   6 dB.  For music- or desktop-sharing applications, the level SHOULD   NOT be automatically adjusted, and the endpoint SHOULD allow the user   to set the gain manually.   The RECOMMENDED filter for normalizing the signal energy is a second-   order Butterworth filter with a 300 Hz cutoff frequency.   It is common for the audio output on some devices to be "calibrated"   for playing back pre-recorded "commercial" music, which is typically   around 12 dB louder than the level recommended in this section.   Because of this, endpoints MAY increase the gain before playback.5.  Acoustic Echo Cancellation (AEC)   It is plausible that the dominant near-to-medium-term WebRTC usage   model will be people using the interactive audio and video   capabilities to communicate with each other via web browsers running   on a notebook computer that has a built-in microphone and speakers.   The notebook-as-communication-device paradigm presents challengingValin & Bran                 Standards Track                    [Page 4]

RFC 7874                      WebRTC Audio                      May 2016   echo cancellation problems, the specific remedy of which will not be   mandated here.  However, while no specific algorithm or standard will   be required by WebRTC-compatible endpoints, echo cancellation will   improve the user experience and should be implemented by the endpoint   device.   WebRTC endpoints SHOULD include an AEC or some other form of echo   control.  On general-purpose platforms (e.g., a PC), it is common for   the analog-to-digital converter (ADC) for audio capture and the   digital-to-analog converter (DAC) for audio playback to use different   clocks.  In these cases, such as when a webcam is used for capture   and a separate soundcard is used for playback, the sampling rates are   likely to differ slightly.  Endpoint AECs SHOULD be robust to such   conditions, unless they are shipped along with hardware that   guarantees capture and playback to be sampled from the same clock.   Endpoints SHOULD allow the entire AEC and/or the nonlinear processing   (NLP) to be turned off for applications, such as music, that do not   behave well with the spectral attenuation methods typically used in   NLP.  Similarly, endpoints SHOULD have the ability to detect the   presence of a headset and disable echo cancellation.   For some applications where the remote endpoint may not have an echo   canceller, the local endpoint MAY include a far-end echo canceller,   but when included, it SHOULD be disabled by default.6.  Legacy VoIP Interoperability   The codec requirements above will ensure, at a minimum, voice   interoperability capabilities between WebRTC endpoints and legacy   phone systems that support G.711.7.  Security Considerations   For security considerations regarding the codecs themselves, please   refer to their specifications, including [RFC6716], [RFC7587],   [RFC3551], [RFC3389], and [RFC4733].  Likewise, consult the RTP base   specification for RTP-based security considerations.  WebRTC security   is further discussed in [WebRTC-SEC], [WebRTC-SEC-ARCH], and   [WebRTC-RTP-USAGE].   Using the guidelines in [RFC6562], implementers should consider   whether the use of variable bitrate is appropriate for their   application.  Encryption and authentication issues are beyond the   scope of this document.Valin & Bran                 Standards Track                    [Page 5]

RFC 7874                      WebRTC Audio                      May 20168.  References8.1.  Normative References   [RFC2119]  Bradner, S., "Key words for use in RFCs to Indicate              Requirement Levels",BCP 14,RFC 2119,              DOI 10.17487/RFC2119, March 1997,              <http://www.rfc-editor.org/info/rfc2119>.   [RFC3551]  Schulzrinne, H. and S. Casner, "RTP Profile for Audio and              Video Conferences with Minimal Control", STD 65,RFC 3551,              DOI 10.17487/RFC3551, July 2003,              <http://www.rfc-editor.org/info/rfc3551>.   [RFC3389]  Zopf, R., "Real-time Transport Protocol (RTP) Payload for              Comfort Noise (CN)",RFC 3389, DOI 10.17487/RFC3389,              September 2002, <http://www.rfc-editor.org/info/rfc3389>.   [RFC4733]  Schulzrinne, H. and T. Taylor, "RTP Payload for DTMF              Digits, Telephony Tones, and Telephony Signals",RFC 4733,              DOI 10.17487/RFC4733, December 2006,              <http://www.rfc-editor.org/info/rfc4733>.   [RFC6716]  Valin, JM., Vos, K., and T. Terriberry, "Definition of the              Opus Audio Codec",RFC 6716, DOI 10.17487/RFC6716,              September 2012, <http://www.rfc-editor.org/info/rfc6716>.   [RFC6562]  Perkins, C. and JM. Valin, "Guidelines for the Use of              Variable Bit Rate Audio with Secure RTP",RFC 6562,              DOI 10.17487/RFC6562, March 2012,              <http://www.rfc-editor.org/info/rfc6562>.   [RFC7587]  Spittka, J., Vos, K., and JM. Valin, "RTP Payload Format              for the Opus Speech and Audio Codec",RFC 7587,              DOI 10.17487/RFC7587, June 2015,              <http://www.rfc-editor.org/info/rfc7587>.   [G.711]    ITU-T, "Pulse code modulation (PCM) of voice frequencies",              ITU-T Recommendation G.711, November 1988,              <http://www.itu.int/rec/T-REC-G.711-198811-I/en>.8.2.  Informative References   [WebRTC-SEC]              Rescorla, E.,"Security Considerations for WebRTC", Work              in Progress,draft-ietf-rtcweb-security-08, February 2015.Valin & Bran                 Standards Track                    [Page 6]

RFC 7874                      WebRTC Audio                      May 2016   [WebRTC-SEC-ARCH]              Rescorla, E.,"WebRTC Security Architecture", Work in              Progress,draft-ietf-rtcweb-security-arch-11, March 2015.   [WebRTC-RTP-USAGE]              Perkins, C., Westerlund, M., and J. Ott, "Web Real-Time              Communication (WebRTC): Media Transport and Use of RTP",              Work in Progress,draft-ietf-rtcweb-rtp-usage-26, March              2016.   [RFC7875]  Proust, S., Ed., "Additional WebRTC Audio Codecs for              Interoperability",RFC 7875, DOI 10.17487/RFC7875, May              2016, <http://www.rfc-editor.org/info/rfc7875>.Acknowledgements   This document incorporates ideas and text from various other   documents.  In particular, we would like to acknowledge, and say   thanks for, work we incorporated from Harald Alvestrand and Cullen   Jennings.Authors' Addresses   Jean-Marc Valin   Mozilla   331 E. Evelyn Avenue   Mountain View, CA  94041   United States   Email: jmvalin@jmvalin.ca   Cary Bran   Plantronics   345 Encinial Street   Santa Cruz, CA  95060   United States   Phone: +1 206 661-2398   Email: cary.bran@plantronics.comValin & Bran                 Standards Track                    [Page 7]