Network Working Group C. Bran
Internet-Draft Plantronics
Intended status: Standards Track C. Jennings
Expires: September 13, 2012 Cisco
JM. Valin
Mozilla
March 12, 2012
WebRTC Codec and Media Processing Requirements
draft-cbran-rtcweb-codec-02
Abstract
This document outlines the codec and media processing requirements
for WebRTC client application and endpoint devices.
Status of this Memo
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the Trust Legal Provisions and are provided without warranty as
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Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . . 3
2. Terminology . . . . . . . . . . . . . . . . . . . . . . . . . . 3
3. Codec Requirements . . . . . . . . . . . . . . . . . . . . . . 3
3.1. Audio Codec Requirements . . . . . . . . . . . . . . . . . 3
3.2. Video Codec Requirements . . . . . . . . . . . . . . . . . 3
4. Audio Level . . . . . . . . . . . . . . . . . . . . . . . . . . 4
5. Acoustic Echo Cancellation (AEC) . . . . . . . . . . . . . . . 5
6. Legacy VoIP Interoperability . . . . . . . . . . . . . . . . . 6
7. IANA Considerations . . . . . . . . . . . . . . . . . . . . . . 6
8. Security Considerations . . . . . . . . . . . . . . . . . . . . 6
9. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . . 6
10. Normative References . . . . . . . . . . . . . . . . . . . . . 6
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . . 7
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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
media 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
This section covers the audio and video codec requirements for WebRTC
client applications. To ensure a baseline level of interoperability
between WebRTC clients, a minimum set of required codecs are
specified below. While this section specifies the codecs that will
be mandated for all WebRTC client implementations, it leaves the
question of supporting additional codecs to the will of the
implementer.
3.1. Audio Codec Requirements
WebRTC clients are REQUIRED to implement the following audio codecs.
o PCMA/PCMU - 1 channel with a rate of 8000 Hz and a ptime of 20 -
see section 4.5.14 of [RFC3551]
o Telephone Event - [RFC4734]
o Opus [draft-ietf-codec-opus]
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, for all ptime
values up to 120 ms. Clients MAY use the offer/answer mechanism to
signal a preference for a particular mode or ptime.
3.2. Video Codec Requirements
The following feature list applies to all required video codecs.
Required video codecs:
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o MUST support at least 10 frames per second (fps) and SHOULD
support 30 fps
o If VP8 is supported, then it MUST support the bilinear and none
reconstruction filters
o OPTIONALLY offer support for additional color spaces
o MUST support a minimum resolution of 320X240
o SHOULD support resolutions of 1280x720, 720x480, 1024x768,
800x600, 640x480, 640 x 360 , 320x240
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.
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
same time maintaining sufficient dynamic range and reducing the risk
of clipping. The main drawbacks are that the resulting level is
about 12 dB lower than typical "commercial music" levels and it
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
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
what is recommended above without causing interoperability problems.
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.
SHOULD include an AEC and if not, SHOULD ensure that the speaker-to-
microphone gain is below unity at all frequencies to avoid
instability when none of the client has echo cancellation. For
clients that do not control the audio capture and playback devices
directly, it is RECOMMENDED to support echo cancellation between
devices running at slight different sampling rates, such as when a
webcam is used for microphone.
The client SHOULD allow either the entire AEC 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. It 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 it the case, it SHOULD be disabled by default.
Call control event notification to connected devices such as headsets
(what's that exactly?)
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6. Legacy VoIP Interoperability
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
The codec requirements have no additional security considerations
other than those captured in
[I-D.ekr-security-considerations-for-rtc-web].
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.
10. Normative References
[I-D.ekr-security-considerations-for-rtc-web]
Rescorla, E., "Security Considerations for RTC-Web",
May 2011.
[I-D.webm]
Google, Inc., "VP8 Data Format and Decoding Guide",
July 2010.
[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.
[RFC4734] Schulzrinne, H. and T. Taylor, "Definition of Events for
Modem, Fax, and Text Telephony Signals", RFC 4734,
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December 2006.
Authors' Addresses
Cary Bran
Plantronics
345 Encinial Street
Santa Cruz, CA 95060
USA
Phone: +1 206 661-2398
Email: cary.bran@plantronics.com
Cullen Jennings
Cisco
170 West Tasman Drive
San Jose, CA 95134
USA
Phone: +1 408 421-9990
Email: fluffy@cisco.com
Jean-Marc Valin
Mozilla
650 Castro Street
Mountain View, CA 94041
USA
Email: jmvalin@jmvalin.ca
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