draft-ietf-rtcweb-audio-04.txt		draft-ietf-rtcweb-audio-05.txt
	Network Working Group JM. Valin		Network Working Group JM. Valin
	Internet-Draft Mozilla		Internet-Draft Mozilla
	Intended status: Standards Track C. Bran		Intended status: Standards Track C. Bran
	Expires: July 31, 2014 Plantronics		Expires: August 17, 2014 Plantronics
	January 27, 2014		February 13, 2014
	WebRTC Audio Codec and Processing Requirements		WebRTC Audio Codec and Processing Requirements
	draft-ietf-rtcweb-audio-04		draft-ietf-rtcweb-audio-05
	Abstract		Abstract
	This document outlines the audio codec and processing requirements		This document outlines the audio codec and processing requirements
	for WebRTC client application and endpoint devices.		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		This Internet-Draft is submitted in full conformance with the
	provisions of BCP 78 and BCP 79.		provisions of BCP 78 and BCP 79.
	skipping to change at page 1, line 32		skipping to change at page 1, line 32
	Internet-Drafts are working documents of the Internet Engineering		Internet-Drafts are working documents of the Internet Engineering
	Task Force (IETF). Note that other groups may also distribute		Task Force (IETF). Note that other groups may also distribute
	working documents as Internet-Drafts. The list of current Internet-		working documents as Internet-Drafts. The list of current Internet-
	Drafts is at http://datatracker.ietf.org/drafts/current/.		Drafts is at http://datatracker.ietf.org/drafts/current/.
	Internet-Drafts are draft documents valid for a maximum of six months		Internet-Drafts are draft documents valid for a maximum of six months
	and may be updated, replaced, or obsoleted by other documents at any		and may be updated, replaced, or obsoleted by other documents at any
	time. It is inappropriate to use Internet-Drafts as reference		time. It is inappropriate to use Internet-Drafts as reference
	material or to cite them other than as "work in progress."		material or to cite them other than as "work in progress."
	This Internet-Draft will expire on July 31, 2014.		This Internet-Draft will expire on August 17, 2014.
	Copyright Notice		Copyright Notice
	Copyright (c) 2014 IETF Trust and the persons identified as the		Copyright (c) 2014 IETF Trust and the persons identified as the
	document authors. All rights reserved.		document authors. All rights reserved.
	This document is subject to BCP 78 and the IETF Trust's Legal		This document is subject to BCP 78 and the IETF Trust's Legal
	Provisions Relating to IETF Documents		Provisions Relating to IETF Documents
	(http://trustee.ietf.org/license-info) in effect on the date of		(http://trustee.ietf.org/license-info) in effect on the date of
	publication of this document. Please review these documents		publication of this document. Please review these documents
	skipping to change at page 2, line 39		skipping to change at page 2, line 39
	3. Codec Requirements		3. Codec Requirements
	To ensure a baseline level of interoperability between WebRTC		To ensure a baseline level of interoperability between WebRTC
	clients, a minimum set of required codecs are specified below. If		clients, a minimum set of required codecs are specified below. If
	other suitable audio codecs are available for the browser to use, it		other suitable audio codecs are available for the browser to use, it
	is RECOMMENDED that they are also be included in the offer in order		is RECOMMENDED that they are also be included in the offer in order
	to maximize the possibility to establish the session without the need		to maximize the possibility to establish the session without the need
	for audio transcoding.		for audio transcoding.
	WebRTC clients are REQUIRED to implement the following audio codecs.		WebRTC clients are REQUIRED to implement the following audio codecs:
	o Opus [RFC6716], with the payload format specified in [Opus-RTP]		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 any		o G.711 PCMA and PCMU with the payload format specified in section
	ptime value up to 120 ms - see section 4.5.14 of [RFC3551]		4.5.14 of [RFC3551].
	o The audio/telephone-event media format as specified in [RFC4733].		o The audio/telephone-event media format as specified in [RFC4733].
	WebRTC clients are REQUIRED to be able to generate and consume the		WebRTC clients are REQUIRED to be able to generate and consume the
	following events:		following events:
	+------------+--------------------------------+-----------+		+------------+--------------------------------+-----------+
	|Event Code | Event Name | Reference |		|Event Code | Event Name | Reference |
	+------------+--------------------------------+-----------+		+------------+--------------------------------+-----------+
	| 0 | DTMF digit "0" | RFC4733 |		| 0 | DTMF digit "0" | RFC4733 |
	| 1 | DTMF digit "1" | RFC4733 |		| 1 | DTMF digit "1" | RFC4733 |
End of changes. 6 change blocks.
	9 lines changed or deleted		8 lines changed or added
This html diff was produced by rfcdiff 1.41. The latest version is available from http://tools.ietf.org/tools/rfcdiff/