Network Working Group C. Bran
Internet-Draft Plantronics
Intended status: Standards Track C. Jennings
Expires: September 11, 2012 Cisco
JM. Valin
March 12, 2012

WebRTC Codec and Media Processing Requirements


This document outlines the codec and media processing requirements for WebRTC client application and endpoint devices.

Status of this Memo

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Table of Contents

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.

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:

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 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?)

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. 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.
[RFC4734] Schulzrinne, H. and T. Taylor, "Definition of Events for Modem, Fax, and Text Telephony Signals", RFC 4734, December 2006.
[I-D.ekr-security-considerations-for-rtc-web] Rescorla, E.K., "Security Considerations for RTC-Web", May 2011.
[I-D.webm] Google, Inc., , "VP8 Data Format and Decoding Guide", July 2010.

Authors' Addresses

Cary Bran Plantronics 345 Encinial Street Santa Cruz, CA 95060 USA Phone: +1 206 661-2398 EMail:
Cullen Jennings Cisco 170 West Tasman Drive San Jose, CA 95134 USA Phone: +1 408 421-9990 EMail:
Jean-Marc Valin Mozilla 650 Castro Street Mountain View, CA 94041 USA EMail: