--- 1/draft-ietf-rtcweb-audio-01.txt 2013-08-02 10:14:28.194313767 -0700 +++ 2/draft-ietf-rtcweb-audio-02.txt 2013-08-02 10:14:28.210314182 -0700 @@ -1,70 +1,69 @@ Network Working Group JM. Valin Internet-Draft Mozilla Intended status: Standards Track C. Bran -Expires: May 27, 2013 Plantronics - November 23, 2012 +Expires: February 03, 2014 Plantronics + August 02, 2013 WebRTC Audio Codec and Processing Requirements - draft-ietf-rtcweb-audio-01 + draft-ietf-rtcweb-audio-02 Abstract This document outlines the audio codec and processing requirements 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 provisions of BCP 78 and BCP 79. 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 May 27, 2013. + This Internet-Draft will expire on February 03, 2014. 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. 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 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 . . . . . . . . . . . . . . . . . . . . . . . . . 3 - 2. Terminology . . . . . . . . . . . . . . . . . . . . . . . . . . 3 - 3. Codec Requirements . . . . . . . . . . . . . . . . . . . . . . 3 - 4. Audio Level . . . . . . . . . . . . . . . . . . . . . . . . . . 3 - 5. Acoustic Echo Cancellation (AEC) . . . . . . . . . . . . . . . 4 - 6. Legacy VoIP Interoperability . . . . . . . . . . . . . . . . . 5 - 7. IANA Considerations . . . . . . . . . . . . . . . . . . . . . . 5 - 8. Security Considerations . . . . . . . . . . . . . . . . . . . . 5 - 9. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . . 6 - 10. Normative References . . . . . . . . . . . . . . . . . . . . . 6 - Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . . 6 + 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 + 7. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 4 + 8. Security Considerations . . . . . . . . . . . . . . . . . . . 5 + 9. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . 5 + 10. Normative References . . . . . . . . . . . . . . . . . . . . 5 + Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 5 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 audio processing and codec requirements for WebRTC client implementations. 2. Terminology @@ -76,21 +75,22 @@ 3. Codec Requirements 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. 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 ptime of 20 - see section 4.5.14 of [RFC3551] 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. @@ -115,22 +114,23 @@ 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. + 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 @@ -196,43 +196,49 @@ [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 and Cullen Jennings. 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 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. [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. + + [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 Jean-Marc Valin Mozilla 650 Castro Street Mountain View, CA 94041 USA Email: jmvalin@jmvalin.ca - Cary Bran Plantronics 345 Encinial Street Santa Cruz, CA 95060 USA Phone: +1 206 661-2398 Email: cary.bran@plantronics.com