draft-ietf-rtcweb-audio-01.txt   draft-ietf-rtcweb-audio-02.txt 
Network Working Group JM. Valin Network Working Group JM. Valin
Internet-Draft Mozilla Internet-Draft Mozilla
Intended status: Standards Track C. Bran Intended status: Standards Track C. Bran
Expires: May 27, 2013 Plantronics Expires: February 03, 2014 Plantronics
November 23, 2012 August 02, 2013
WebRTC Audio Codec and Processing Requirements WebRTC Audio Codec and Processing Requirements
draft-ietf-rtcweb-audio-01 draft-ietf-rtcweb-audio-02
Abstract Abstract
This document outlines the audio codec and processing requirements This document outlines the audio codec and processing requirements
for WebRTC client application and endpoint devices. for WebRTC client application and endpoint devices.
Status of this Memo Status of This Memo
This Internet-Draft is submitted in full conformance with the This Internet-Draft is submitted in full conformance with the
provisions of BCP 78 and BCP 79. provisions of BCP 78 and BCP 79.
Internet-Drafts are working documents of the Internet Engineering Internet-Drafts are working documents of the Internet Engineering
Task Force (IETF). Note that other groups may also distribute Task Force (IETF). Note that other groups may also distribute
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and may be updated, replaced, or obsoleted by other documents at any and may be updated, replaced, or obsoleted by other documents at any
time. It is inappropriate to use Internet-Drafts as reference time. It is inappropriate to use Internet-Drafts as reference
material or to cite them other than as "work in progress." material or to cite them other than as "work in progress."
This Internet-Draft will expire on May 27, 2013. This Internet-Draft will expire on February 03, 2014.
Copyright Notice Copyright Notice
Copyright (c) 2012 IETF Trust and the persons identified as the Copyright (c) 2013 IETF Trust and the persons identified as the
document authors. All rights reserved. document authors. All rights reserved.
This document is subject to BCP 78 and the IETF Trust's Legal This document is subject to BCP 78 and the IETF Trust's Legal
Provisions Relating to IETF Documents Provisions Relating to IETF Documents
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publication of this document. Please review these documents publication of this document. Please review these documents
carefully, as they describe your rights and restrictions with respect carefully, as they describe your rights and restrictions with respect
to this document. Code Components extracted from this document must to this document. Code Components extracted from this document must
include Simplified BSD License text as described in Section 4.e of include Simplified BSD License text as described in Section 4.e of
the Trust Legal Provisions and are provided without warranty as the Trust Legal Provisions and are provided without warranty as
described in the Simplified BSD License. described in the Simplified BSD License.
Table of Contents Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 2
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . . 3 2. Terminology . . . . . . . . . . . . . . . . . . . . . . . . . 2
2. Terminology . . . . . . . . . . . . . . . . . . . . . . . . . . 3 3. Codec Requirements . . . . . . . . . . . . . . . . . . . . . 2
3. Codec Requirements . . . . . . . . . . . . . . . . . . . . . . 3 4. Audio Level . . . . . . . . . . . . . . . . . . . . . . . . . 3
4. Audio Level . . . . . . . . . . . . . . . . . . . . . . . . . . 3 5. Acoustic Echo Cancellation (AEC) . . . . . . . . . . . . . . 4
5. Acoustic Echo Cancellation (AEC) . . . . . . . . . . . . . . . 4 6. Legacy VoIP Interoperability . . . . . . . . . . . . . . . . 4
6. Legacy VoIP Interoperability . . . . . . . . . . . . . . . . . 5 7. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 4
7. IANA Considerations . . . . . . . . . . . . . . . . . . . . . . 5 8. Security Considerations . . . . . . . . . . . . . . . . . . . 5
8. Security Considerations . . . . . . . . . . . . . . . . . . . . 5 9. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . 5
9. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . . 6 10. Normative References . . . . . . . . . . . . . . . . . . . . 5
10. Normative References . . . . . . . . . . . . . . . . . . . . . 6 Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 5
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . . 6
1. Introduction 1. Introduction
An integral part of the success and adoption of the Web Real Time An integral part of the success and adoption of the Web Real Time
Communications (WebRTC) will be the voice and video interoperability Communications (WebRTC) will be the voice and video interoperability
between WebRTC applications. This specification will outline the between WebRTC applications. This specification will outline the
audio processing and codec requirements for WebRTC client audio processing and codec requirements for WebRTC client
implementations. implementations.
2. Terminology 2. Terminology
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3. Codec Requirements 3. Codec Requirements
To ensure a baseline level of interoperability between WebRTC To ensure a baseline level of interoperability between WebRTC
clients, a minimum set of required codecs are specified below. While clients, a minimum set of required codecs are specified below. While
this section specifies the codecs that will be mandated for all this section specifies the codecs that will be mandated for all
WebRTC client implementations, it leaves the question of supporting WebRTC client implementations, it leaves the question of supporting
additional codecs to the will of the implementer. additional codecs to the will of the implementer.
WebRTC clients are REQUIRED to implement the following audio codecs. WebRTC clients are REQUIRED to implement the following audio codecs.
o Opus [RFC6716], with any ptime value up to 120 ms o Opus [RFC6716], with the payload format specified in [Opus-RTP]
and any ptime value up to 120 ms
o G.711 PCMA and PCMU with one channel, a rate of 8000 Hz and a o G.711 PCMA and PCMU with one channel, a rate of 8000 Hz and a
ptime of 20 - see section 4.5.14 of [RFC3551] ptime of 20 - see section 4.5.14 of [RFC3551]
o Telephone Event - [RFC4733] o Telephone Event - [RFC4733]
For all cases where the client is able to process audio at a sampling 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 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 PCMA/PCMU. For Opus, all modes MUST be supported on the decoder
side. The choice of encoder-side modes is left to the implementer. side. The choice of encoder-side modes is left to the implementer.
skipping to change at page 4, line 22 skipping to change at page 3, line 32
compressor. compressor.
AUTHORS' NOTE: The idea of using the same level as what the ITU-T AUTHORS' NOTE: The idea of using the same level as what the ITU-T
recommends is that it should improve inter-operability while at the recommends is that it should improve inter-operability while at the
same time maintaining sufficient dynamic range and reducing the risk same time maintaining sufficient dynamic range and reducing the risk
of clipping. The main drawbacks are that the resulting level is of clipping. The main drawbacks are that the resulting level is
about 12 dB lower than typical "commercial music" levels and it about 12 dB lower than typical "commercial music" levels and it
leaves room for ill-behaved clients to be much louder than a normal leaves room for ill-behaved clients to be much louder than a normal
client. While using music-type levels is not really an option (it client. While using music-type levels is not really an option (it
would require using the same compressor-limitors that studios use), would require using the same compressor-limitors that studios use),
it would be possible to have a level slightly higher (e.g. 3 dB) than it would be possible to have a level slightly higher (e.g. 3 dB)
what is recommended above without causing interoperability problems. than what is recommended above without causing interoperability
problems.
Assuming 16-bit PCM with a value of +/-32767, -19 dBm0 corresponds to 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 a root mean square (RMS) level of 2600. Only active speech should be
considered in the RMS calculation. If the client has control over considered in the RMS calculation. If the client has control over
the entire audio capture path, as is typically the case for a regular 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 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 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 speaker. If the client does not have control over the entire audio
capture, as is typically the case for a software client, then the capture, as is typically the case for a software client, then the
client SHOULD use automatic gain control (AGC) to dynamically adjust client SHOULD use automatic gain control (AGC) to dynamically adjust
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[I-D.ekr-security-considerations-for-rtc-web]. [I-D.ekr-security-considerations-for-rtc-web].
9. Acknowledgements 9. Acknowledgements
This draft incorporates ideas and text from various other drafts. In This draft incorporates ideas and text from various other drafts. In
particularly we would like to acknowledge, and say thanks for, work particularly we would like to acknowledge, and say thanks for, work
we incorporated from Harald Alvestrand and Cullen Jennings. we incorporated from Harald Alvestrand and Cullen Jennings.
10. Normative References 10. Normative References
[I-D.ekr-security-considerations-for-rtc-web]
Rescorla, E., "Security Considerations for RTC-Web",
May 2011.
[RFC2119] Bradner, S., "Key words for use in RFCs to Indicate [RFC2119] Bradner, S., "Key words for use in RFCs to Indicate
Requirement Levels", BCP 14, RFC 2119, March 1997. Requirement Levels", BCP 14, RFC 2119, March 1997.
[RFC3551] Schulzrinne, H. and S. Casner, "RTP Profile for Audio and [RFC3551] Schulzrinne, H. and S. Casner, "RTP Profile for Audio and
Video Conferences with Minimal Control", STD 65, RFC 3551, Video Conferences with Minimal Control", STD 65, RFC 3551,
July 2003. July 2003.
[RFC4733] Schulzrinne, H. and T. Taylor, "RTP Payload for DTMF [RFC4733] Schulzrinne, H. and T. Taylor, "RTP Payload for DTMF
Digits, Telephony Tones, and Telephony Signals", RFC 4733, Digits, Telephony Tones, and Telephony Signals", RFC 4733,
December 2006. December 2006.
[RFC6716] Valin, JM., Vos, K., and T. Terriberry, "Definition of the
Opus Audio Codec", RFC 6716, September 2012.
[Opus-RTP]
Spittka, J., Vos, K., and JM. Valin, "RTP Payload Format
for Opus Codec", August 2013.
[I-D.ekr-security-considerations-for-rtc-web]
Rescorla, E.K., "Security Considerations for RTC-Web", May
2011.
Authors' Addresses Authors' Addresses
Jean-Marc Valin Jean-Marc Valin
Mozilla Mozilla
650 Castro Street 650 Castro Street
Mountain View, CA 94041 Mountain View, CA 94041
USA USA
Email: jmvalin@jmvalin.ca Email: jmvalin@jmvalin.ca
Cary Bran Cary Bran
Plantronics Plantronics
345 Encinial Street 345 Encinial Street
Santa Cruz, CA 95060 Santa Cruz, CA 95060
USA USA
Phone: +1 206 661-2398 Phone: +1 206 661-2398
Email: cary.bran@plantronics.com Email: cary.bran@plantronics.com
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