Network Working Group C. Perkins
Internet-Draft University of Glasgow
Intended status: Standards Track M. Westerlund
Expires: August 29, 2013 Ericsson
J. Ott
Aalto University
February 25, 2013
Web Real-Time Communication (WebRTC): Media Transport and Use of RTP
draft-ietf-rtcweb-rtp-usage-06
Abstract
The Web Real-Time Communication (WebRTC) framework provides support
for direct interactive rich communication using audio, video, text,
collaboration, games, etc. between two peers' web-browsers. This
memo describes the media transport aspects of the WebRTC framework.
It specifies how the Real-time Transport Protocol (RTP) is used in
the WebRTC context, and gives requirements for which RTP features,
profiles, and extensions need to be supported.
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
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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 29, 2013.
Copyright Notice
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publication of this document. Please review these documents
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Internet-Draft RTP for WebRTC February 2013
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Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . . 4
2. Rationale . . . . . . . . . . . . . . . . . . . . . . . . . . 4
3. Terminology . . . . . . . . . . . . . . . . . . . . . . . . . 5
4. WebRTC Use of RTP: Core Protocols . . . . . . . . . . . . . . 6
4.1. RTP and RTCP . . . . . . . . . . . . . . . . . . . . . . . 6
4.2. Choice of the RTP Profile . . . . . . . . . . . . . . . . 7
4.3. Choice of RTP Payload Formats . . . . . . . . . . . . . . 8
4.4. RTP Session Multiplexing . . . . . . . . . . . . . . . . . 8
4.5. RTP and RTCP Multiplexing . . . . . . . . . . . . . . . . 9
4.6. Reduced Size RTCP . . . . . . . . . . . . . . . . . . . . 10
4.7. Symmetric RTP/RTCP . . . . . . . . . . . . . . . . . . . . 10
4.8. Choice of RTP Synchronisation Source (SSRC) . . . . . . . 10
4.9. Generation of the RTCP Canonical Name (CNAME) . . . . . . 11
5. WebRTC Use of RTP: Extensions . . . . . . . . . . . . . . . . 11
5.1. Conferencing Extensions . . . . . . . . . . . . . . . . . 11
5.1.1. Full Intra Request (FIR) . . . . . . . . . . . . . . . 12
5.1.2. Picture Loss Indication (PLI) . . . . . . . . . . . . 13
5.1.3. Slice Loss Indication (SLI) . . . . . . . . . . . . . 13
5.1.4. Reference Picture Selection Indication (RPSI) . . . . 13
5.1.5. Temporal-Spatial Trade-off Request (TSTR) . . . . . . 13
5.1.6. Temporary Maximum Media Stream Bit Rate Request
(TMMBR) . . . . . . . . . . . . . . . . . . . . . . . 13
5.2. Header Extensions . . . . . . . . . . . . . . . . . . . . 14
5.2.1. Rapid Synchronisation . . . . . . . . . . . . . . . . 14
5.2.2. Client-to-Mixer Audio Level . . . . . . . . . . . . . 14
5.2.3. Mixer-to-Client Audio Level . . . . . . . . . . . . . 15
6. WebRTC Use of RTP: Improving Transport Robustness . . . . . . 15
6.1. Negative Acknowledgements and RTP Retransmission . . . . . 15
6.2. Forward Error Correction (FEC) . . . . . . . . . . . . . . 16
7. WebRTC Use of RTP: Rate Control and Media Adaptation . . . . . 17
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