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Versions: (draft-cbran-rtcweb-codec) 00 01 02 03 04 05 06 07 08 09 10 11 RFC 7874

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


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

Abstract

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

Status of This Memo

   This Internet-Draft is submitted in full conformance with the
   provisions of BCP 78 and BCP 79.

   Internet-Drafts are working documents of the Internet Engineering
   Task Force (IETF).  Note that other groups may also distribute
   working documents as Internet-Drafts.  The list of current Internet-
   Drafts is at http://datatracker.ietf.org/drafts/current/.

   Internet-Drafts are draft documents valid for a maximum of six months
   and may be updated, replaced, or obsoleted by other documents at any
   time.  It is inappropriate to use Internet-Drafts as reference
   material or to cite them other than as "work in progress."

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

Copyright Notice

   Copyright (c) 2014 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 . . . . . . . . . . . . . . . . . . . . . . . . .   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
     10.2.  Informative References . . . . . . . . . . . . . . . . .   6
   Authors' Addresses  . . . . . . . . . . . . . . . . . . . . . . .   6

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
   implementations.

2.  Terminology

   The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
   "SHOULD", "SHOULD 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, 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 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].





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   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 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 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 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



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   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 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 does not have control over the entire audio
   capture, as is typically the case for a software client, then the
   client 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 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 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, echo cancellation will improve the user
   experience and should be implemented by the endpoint device.

   WebRTC clients 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 the audio capture and playback hardware,
   it is RECOMMENDED to support echo cancellation between devices
   running at slightly different sampling rates, such as when a webcam
   is used for microphone.




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   Clients 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 SHOULD have the ability to detect the
   presence of a headset and disable echo cancellation.

   For some applications where the remote client may not have an echo
   canceller, the local client 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 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

   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.



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   [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 Opus Speech and Audio Codec", draft-ietf-payload-rtp-
              opus-03 (work in progress), July 2014.

10.2.  Informative References

   [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 (work in progress),
              September 2014.

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







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