





Internet Engineering Task Force (IETF)                         JM. Valin
Request for Comments: 7874                                       Mozilla
Category: Standards Track                                        C. Bran
ISSN: 2070-1721                                              Plantronics
                                                                May 2016


             WebRTC Audio Codec and Processing Requirements

Abstract

   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 in Section 2 of RFC 5741.

   Information about the current status of this document, any errata,
   and how to provide feedback on it may be obtained at
   http://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 to BCP 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.









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Table of Contents

   1.  Introduction  . . . . . . . . . . . . . . . . . . . . . . . .   2
   2.  Terminology . . . . . . . . . . . . . . . . . . . . . . . . .   2
   3.  Codec Requirements  . . . . . . . . . . . . . . . . . . . . .   2
   4.  Audio Level . . . . . . . . . . . . . . . . . . . . . . . . .   4
   5.  Acoustic Echo Cancellation (AEC)  . . . . . . . . . . . . . .   4
   6.  Legacy VoIP Interoperability  . . . . . . . . . . . . . . . .   5
   7.  Security Considerations . . . . . . . . . . . . . . . . . . .   5
   8.  References  . . . . . . . . . . . . . . . . . . . . . . . . .   6
     8.1.  Normative References  . . . . . . . . . . . . . . . . . .   6
     8.2.  Informative References  . . . . . . . . . . . . . . . . .   6
   Acknowledgements  . . . . . . . . . . . . . . . . . . . . . . . .   7
   Authors' Addresses  . . . . . . . . . . . . . . . . . . . . . . .   7

1.  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 in RFC
   2119 [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 in Section 4.5.14 of [RFC3551].






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   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 in RFC 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 in RFC 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.



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   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 challenging



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   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.






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RFC 7874                      WebRTC Audio                      May 2016


8.  References

8.1.  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.





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   [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.com











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