Internet DRAFT - draft-ietf-rtcweb-audio
draft-ietf-rtcweb-audio
Network Working Group JM. Valin
Internet-Draft Mozilla
Intended status: Standards Track C. Bran
Expires: October 23, 2016 Plantronics
April 21, 2016
WebRTC Audio Codec and Processing Requirements
draft-ietf-rtcweb-audio-11
Abstract
This document outlines the audio codec and processing requirements
for WebRTC 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 October 23, 2016.
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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. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 5
8. Security Considerations . . . . . . . . . . . . . . . . . . . 5
9. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . 6
10. References . . . . . . . . . . . . . . . . . . . . . . . . . 6
10.1. Normative References . . . . . . . . . . . . . . . . . . 6
10.2. Informative References . . . . . . . . . . . . . . . . . 7
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 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 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 are also be included in the offer in
order to maximize the possibility to establish 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 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). 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 DTX/CN by senders is OPTIONAL.
o The audio/telephone-event media format as specified in [RFC4733].
The endpoints MAY send DTMF events at any time and SHOULD suppress
in-band 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 RFC 4733 tones sent to them,
except gracefully drop them. There is currently no API to inform
JavaScript about the received DTMF or other RFC 4733 tones.
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
[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 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 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 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
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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. PC), it is common for
the audio capture ADC and the audio playback DAC 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 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, 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 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 endpoints 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], [RFC7587],
[RFC3551], [RFC3389], and [RFC4733]. Likewise, consult the RTP base
specification for 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].
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Implementers should consider whether the use of variable bitrate 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
particular 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,
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>.
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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-25 (work in progress), June
2015.
[I-D.ietf-rtcweb-audio-codecs-for-interop]
Proust, S., "Additional WebRTC audio codecs for
interoperability.", draft-ietf-rtcweb-audio-codecs-for-
interop-05 (work in progress), February 2016.
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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