Network Working Group JM. Valin
Internet-Draft Mozilla
Intended status: Standards Track C. Bran
Expires: August 17, 2014 Plantronics
February 13, 2014
WebRTC Audio Codec and Processing Requirements
draft-ietf-rtcweb-audio-05
Abstract
This document outlines the audio codec and processing requirements
for WebRTC client application and endpoint devices.
Status of This Memo
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Table of Contents
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1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 2
2. Terminology . . . . . . . . . . . . . . . . . . . . . . . . . 2
3. Codec Requirements . . . . . . . . . . . . . . . . . . . . . 2
4. Audio Level . . . . . . . . . . . . . . . . . . . . . . . . . 3
5. Acoustic Echo Cancellation (AEC) . . . . . . . . . . . . . . 4
6. Legacy VoIP Interoperability . . . . . . . . . . . . . . . . 4
7. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 5
8. Security Considerations . . . . . . . . . . . . . . . . . . . 5
9. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . 5
10. Normative References . . . . . . . . . . . . . . . . . . . . 5
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 [Opus-RTP].
o G.711 PCMA and PCMU with the payload format specified in section
4.5.14 of [RFC3551].
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:
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+------------+--------------------------------+-----------+
|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.
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 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
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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.
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
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The codec requirements above will ensure, at a minimum, voice
interoperability capabilities between WebRTC client applications and
legacy phone systems.
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. 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.
[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.
[Opus-RTP]
Spittka, J., Vos, K., and JM. Valin, "RTP Payload Format
for Opus Codec", August 2013.
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Authors' Addresses
Jean-Marc Valin
Mozilla
650 Castro Street
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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