--- 1/draft-ietf-rtcweb-audio-00.txt 2012-11-23 09:14:16.669341961 +0100 +++ 2/draft-ietf-rtcweb-audio-01.txt 2012-11-23 09:14:16.681341833 +0100 @@ -1,19 +1,19 @@ Network Working Group JM. Valin Internet-Draft Mozilla Intended status: Standards Track C. Bran -Expires: March 11, 2013 Plantronics - September 7, 2012 +Expires: May 27, 2013 Plantronics + November 23, 2012 WebRTC Audio Codec and Processing Requirements - draft-ietf-rtcweb-audio-00 + draft-ietf-rtcweb-audio-01 Abstract This document outlines the audio codec and processing requirements for WebRTC client application and endpoint devices. Status of this Memo This Internet-Draft is submitted in full conformance with the provisions of BCP 78 and BCP 79. @@ -21,21 +21,21 @@ Internet-Drafts are working documents of the Internet Engineering Task Force (IETF). Note that other groups may also distribute working documents as Internet-Drafts. The list of current Internet- Drafts is at http://datatracker.ietf.org/drafts/current/. Internet-Drafts are draft documents valid for a maximum of six months and may be updated, replaced, or obsoleted by other documents at any time. It is inappropriate to use Internet-Drafts as reference material or to cite them other than as "work in progress." - This Internet-Draft will expire on March 11, 2013. + This Internet-Draft will expire on May 27, 2013. Copyright Notice Copyright (c) 2012 IETF Trust and the persons identified as the document authors. All rights reserved. This document is subject to BCP 78 and the IETF Trust's Legal Provisions Relating to IETF Documents (http://trustee.ietf.org/license-info) in effect on the date of publication of this document. Please review these documents @@ -81,21 +81,21 @@ WebRTC client implementations, it leaves the question of supporting additional codecs to the will of the implementer. WebRTC clients are REQUIRED to implement the following audio codecs. o Opus [RFC6716], with any ptime value up to 120 ms 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] - o Telephone Event - [RFC4734] + o Telephone Event - [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 @@ -151,37 +151,37 @@ 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. + WebRTC clients SHOULD include an AEC 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 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. + 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 client applications and legacy phone systems. 7. IANA Considerations This document makes no request of IANA. @@ -207,22 +207,22 @@ Rescorla, E., "Security Considerations for RTC-Web", May 2011. [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, + [RFC4733] Schulzrinne, H. and T. Taylor, "RTP Payload for DTMF + Digits, Telephony Tones, and Telephony Signals", RFC 4733, December 2006. Authors' Addresses Jean-Marc Valin Mozilla 650 Castro Street Mountain View, CA 94041 USA