--- 1/draft-ietf-rtcweb-audio-05.txt 2014-09-05 16:14:34.281967867 -0700 +++ 2/draft-ietf-rtcweb-audio-06.txt 2014-09-05 16:14:34.297968257 -0700 @@ -1,19 +1,19 @@ Network Working Group JM. Valin Internet-Draft Mozilla Intended status: Standards Track C. Bran -Expires: August 17, 2014 Plantronics - February 13, 2014 +Expires: March 9, 2015 Plantronics + September 5, 2014 WebRTC Audio Codec and Processing Requirements - draft-ietf-rtcweb-audio-05 + draft-ietf-rtcweb-audio-06 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 August 17, 2014. + This Internet-Draft will expire on March 9, 2015. Copyright Notice Copyright (c) 2014 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 @@ -39,26 +39,27 @@ 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 carefully, as they describe your rights and restrictions with respect to this document. Code Components extracted from this document must include Simplified BSD License text as described in Section 4.e of the Trust Legal Provisions and are provided without warranty as described in the Simplified BSD License. Table of Contents + 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 + 6. Legacy VoIP Interoperability . . . . . . . . . . . . . . . . 5 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 @@ -81,20 +82,26 @@ 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 [RFC3389] comfort noise (CN). Receivers MUST support RFC3389 CN + for streams encoded with G.711 or any other supported codec that + does not provide its own CN. Since Opus provides its own CN + mechanism, the use of RFC3389 CN with Opus is NOT RECOMMENDED. + Use of DTX/CN by senders is OPTIONAL. + 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: +------------+--------------------------------+-----------+ |Event Code | Event Name | Reference | +------------+--------------------------------+-----------+ | 0 | DTMF digit "0" | RFC4733 | | 1 | DTMF digit "1" | RFC4733 | | 2 | DTMF digit "2" | RFC4733 | @@ -210,40 +218,43 @@ 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. + [RFC3389] Zopf, R., "Real-time Transport Protocol (RTP) Payload for + Comfort Noise (CN)", RFC 3389, September 2002. + [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. Authors' Addresses Jean-Marc Valin Mozilla - 650 Castro Street + 331 E. Evelyn Avenue Mountain View, CA 94041 USA Email: jmvalin@jmvalin.ca Cary Bran Plantronics 345 Encinial Street Santa Cruz, CA 95060 USA