Network Working Group                                          JM. Valin
Internet-Draft                                                   Mozilla
Intended status: Standards Track                                 C. Bran
Expires: April 27, November 1, 2015                                    Plantronics
                                                        October 24, 2014
                                                          April 30, 2015

             WebRTC Audio Codec and Processing Requirements
                       draft-ietf-rtcweb-audio-07
                       draft-ietf-rtcweb-audio-08

Abstract

   This document outlines the audio codec and processing requirements
   for WebRTC client application and endpoint devices. endpoints.

Status of This Memo

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   provisions of BCP 78 and BCP 79.

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   This Internet-Draft will expire on April 27, November 1, 2015.

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   Copyright (c) 2014 2015 IETF Trust and the persons identified as the
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Table of Contents

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

1.  Introduction

   An integral part of the success and adoption of the 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 client endpoint
   implementations.

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
   clients,
   endpoints, a minimum set of required codecs are specified below.  If
   other suitable audio codecs are available for the browser to use, it
   is RECOMMENDED that they are also be included in the offer in order
   to maximize the possibility to establish the session without the need
   for audio transcoding.

   WebRTC clients endpoints are REQUIRED to implement the following audio
   codecs:

   o  Opus [RFC6716] with the payload format specified in
      [I-D.ietf-payload-rtp-opus].

   o  G.711 PCMA and PCMU with the payload format specified in section
      4.5.14 of [RFC3551].

   o  [RFC3389] comfort noise (CN).  Receivers 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 DTX/CN by senders is OPTIONAL.

   o  The audio/telephone-event media format as specified in [RFC4733].
      WebRTC clients 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  |
         +------------+--------------------------------+-----------+

   For all cases where the client 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.
   Clients  Endpoints MAY use the offer/answer mechanism to signal
   a preference for a particular mode or ptime.

   For additional information on implementing codecs other than the
   mandatory-to-implement codecs listed above, refer to
   [I-D.ietf-rtcweb-audio-codecs-for-interop].

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 kHz to 48 kHz and up.  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 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 client 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 active speech have a level of 2600 (-19 dBm0) for an average
   speaker.  If the client endpoint does not have control over the entire audio
   capture, as is typically the case for a software client, endpoint, then the
   client
   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
   client 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, clients endpoints MAY increase the gain before playback.

5.  Acoustic Echo Cancellation (AEC)

   It is plausible that the dominant near to mid-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 built-in microphone and speakers.  The notebook-as-
   communication-device paradigm presents challenging 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 clients, endpoints, echo cancellation will improve the user
   experience and should be implemented by the endpoint device.

   WebRTC clients endpoints SHOULD include an AEC or some other form of echo
   control and if that is not possible, the clients SHOULD ensure that
   the speaker-to-microphone gain is below unity at all frequencies to
   avoid instability when none of the client has echo
   control.  For
   clients that do not control  On general purpose platforms (e.g.  PC), it is common for
   the audio capture ADC and the audio playback hardware,
   it is RECOMMENDED DAC to support echo cancellation between devices
   running at slightly use different sampling rates,
   clocks.  In these cases, such as when a webcam is used for microphone.

   Clients 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 non-linear
   processing (NLP) to be turned off for applications, such as music,
   that do not behave well with the spectral attenuation methods
   typically used in NLPs.  Similarly, clients endpoints SHOULD have the ability
   to detect the presence of a headset and disable echo cancellation.

   For some applications where the remote client endpoint may not have an echo
   canceller, the local client endpoint MAY include a far-end echo canceller,
   but if that is the case, it SHOULD be disabled by default.

6.  Legacy VoIP Interoperability

   The codec requirements above will ensure, at a minimum, voice
   interoperability capabilities between WebRTC client endpoints applications
   and legacy phone systems that support G.711.

7.  IANA Considerations

   This document makes no request of IANA.

   Note to RFC Editor: this section may be removed on publication as an
   RFC.

8.  Security Considerations

   For security considerations regarding the codecs themselves please
   refer their specifications, including [RFC6716],
   [I-D.ietf-payload-rtp-opus], [RFC3551], [RFC3389], and [RFC4733].
   Likewise, consult the RTP base specification for security RTP-based
   security considerations.  WebRTC security is further discussed in
   [I-D.ietf-rtcweb-security] and [I-D.ietf-rtcweb-security-arch] and
   [I-D.ietf-rtcweb-rtp-usage].

   Implementers should consider whether the use of VBR is appropriate
   for their application based on [RFC6562].  Encryption and
   authentication issues are beyond the scope of this document.

9.  Acknowledgements

   This draft incorporates ideas and text from various other drafts.  In
   particularly we would like to acknowledge, and say thanks for, work
   we incorporated from Harald Alvestrand and Cullen Jennings.

10.  References
10.1.  Normative References

   [RFC2119]  Bradner, S., "Key words for use in RFCs to Indicate
              Requirement Levels", BCP 14, RFC 2119, March 1997.

   [RFC3551]  Schulzrinne, H. and S. Casner, "RTP Profile for Audio and
              Video Conferences with Minimal Control", STD 65, RFC 3551,
              July 2003.

   [RFC3389]  Zopf, R., "Real-time Transport Protocol (RTP) Payload for
              Comfort Noise (CN)", RFC 3389, September 2002.

   [RFC4733]  Schulzrinne, H. and T. Taylor, "RTP Payload for DTMF
              Digits, Telephony Tones, and Telephony Signals", RFC 4733,
              December 2006.

   [RFC6716]  Valin, JM., Vos, K., and T. Terriberry, "Definition of the
              Opus Audio Codec", RFC 6716, September 2012.

   [RFC6562]  Perkins, C. and JM. Valin, "Guidelines for the Use of
              Variable Bit Rate Audio with Secure RTP", RFC 6562, March
              2012.

   [I-D.ietf-payload-rtp-opus]
              Spittka, J., Vos, K., and J. Valin, "RTP Payload Format
              for the Opus Speech and Audio Codec", draft-ietf-payload-rtp-
              opus-03 draft-ietf-payload-
              rtp-opus-11 (work in progress), July 2014. April 2015.

10.2.  Informative References

   [I-D.ietf-rtcweb-security]
              Rescorla, E., "Security Considerations for WebRTC", draft-
              ietf-rtcweb-security-08 (work in progress), February 2015.

   [I-D.ietf-rtcweb-security-arch]
              Rescorla, E., "WebRTC Security Architecture", draft-ietf-
              rtcweb-security-arch-11 (work in progress), March 2015.

   [I-D.ietf-rtcweb-rtp-usage]
              Perkins, C., Westerlund, M., and J. Ott, "Web Real-Time
              Communication (WebRTC): Media Transport and Use of RTP",
              draft-ietf-rtcweb-rtp-usage-23 (work in progress), March
              2015.

   [I-D.ietf-rtcweb-audio-codecs-for-interop]
              Proust, S., Berger, E., Feiten, B., Bogineni, K., Lei, M.,
              and E. Marocco, "Additional WebRTC audio codecs for
              interoperability with legacy networks.", draft-ietf-
              rtcweb-audio-codecs-for-interop-00
              interoperability.", draft-ietf-rtcweb-audio-codecs-for-
              interop-01 (work in progress),
              September 2014. January 2015.

Authors' Addresses

   Jean-Marc Valin
   Mozilla
   331 E. Evelyn Avenue
   Mountain View, CA  94041
   USA

   Email: jmvalin@jmvalin.ca

   Cary Bran
   Plantronics
   345 Encinial Street
   Santa Cruz, CA  95060
   USA

   Phone: +1 206 661-2398
   Email: cary.bran@plantronics.com