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rfc:rfc7874

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.

Valin & Bran Standards Track [Page 1] RFC 7874 WebRTC Audio May 2016

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].

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 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.

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 challenging

Valin & 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 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.

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.com

Valin & Bran Standards Track [Page 7]

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