Media Transport and Use of RTP in WebRTC
RFC 8834
| Document | Type | RFC - Proposed Standard (January 2021; No errata) | |
|---|---|---|---|
| Authors | Colin Perkins , Magnus Westerlund , Joerg Ott | ||
| Last updated | 2021-01-18 | ||
| Replaces | draft-perkins-rtcweb-rtp-usage | ||
| Stream | IETF | ||
| Formats | plain text html xml pdf htmlized bibtex | ||
| Reviews | |||
| Stream | WG state | Submitted to IESG for Publication | |
| Document shepherd | Ted Hardie | ||
| Shepherd write-up | Show (last changed 2015-02-10) | ||
| IESG | IESG state | RFC 8834 (Proposed Standard) | |
| Action Holders |
(None)
|
||
| Consensus Boilerplate | Yes | ||
| Telechat date | |||
| Responsible AD | Alissa Cooper | ||
| Send notices to | (None) | ||
| IANA | IANA review state | Version Changed - Review Needed | |
| IANA action state | No IANA Actions | ||
Internet Engineering Task Force (IETF) C. Perkins
Request for Comments: 8834 University of Glasgow
Category: Standards Track M. Westerlund
ISSN: 2070-1721 Ericsson
J. Ott
Technical University Munich
January 2021
Media Transport and Use of RTP in WebRTC
Abstract
The framework for Web Real-Time Communication (WebRTC) 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 is an Internet Standards Track document.
This document is a product of the Internet Engineering Task Force
(IETF). It represents the consensus of the IETF community. It has
received public review and has been approved for publication by the
Internet Engineering Steering Group (IESG). Further information on
Internet Standards is available in Section 2 of RFC 7841.
Information about the current status of this document, any errata,
and how to provide feedback on it may be obtained at
https://www.rfc-editor.org/info/rfc8834.
Copyright Notice
Copyright (c) 2021 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
(https://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. Rationale
3. Terminology
4. WebRTC Use of RTP: Core Protocols
4.1. RTP and RTCP
4.2. Choice of the RTP Profile
4.3. Choice of RTP Payload Formats
4.4. Use of RTP Sessions
4.5. RTP and RTCP Multiplexing
4.6. Reduced Size RTCP
4.7. Symmetric RTP/RTCP
4.8. Choice of RTP Synchronization Source (SSRC)
4.9. Generation of the RTCP Canonical Name (CNAME)
4.10. Handling of Leap Seconds
5. WebRTC Use of RTP: Extensions
5.1. Conferencing Extensions and Topologies
5.1.1. Full Intra Request (FIR)
5.1.2. Picture Loss Indication (PLI)
5.1.3. Slice Loss Indication (SLI)
5.1.4. Reference Picture Selection Indication (RPSI)
5.1.5. Temporal-Spatial Trade-Off Request (TSTR)
5.1.6. Temporary Maximum Media Stream Bit Rate Request (TMMBR)
5.2. Header Extensions
5.2.1. Rapid Synchronization
5.2.2. Client-to-Mixer Audio Level
5.2.3. Mixer-to-Client Audio Level
5.2.4. Media Stream Identification
5.2.5. Coordination of Video Orientation
6. WebRTC Use of RTP: Improving Transport Robustness
6.1. Negative Acknowledgements and RTP Retransmission
6.2. Forward Error Correction (FEC)
7. WebRTC Use of RTP: Rate Control and Media Adaptation
7.1. Boundary Conditions and Circuit Breakers
7.2. Congestion Control Interoperability and Legacy Systems
8. WebRTC Use of RTP: Performance Monitoring
9. WebRTC Use of RTP: Future Extensions
10. Signaling Considerations
11. WebRTC API Considerations
12. RTP Implementation Considerations
12.1. Configuration and Use of RTP Sessions
12.1.1. Use of Multiple Media Sources within an RTP Session
12.1.2. Use of Multiple RTP Sessions
12.1.3. Differentiated Treatment of RTP Streams
12.2. Media Source, RTP Streams, and Participant Identification
12.2.1. Media Source Identification
12.2.2. SSRC Collision Detection
12.2.3. Media Synchronization Context
13. Security Considerations
14. IANA Considerations
15. References
15.1. Normative References
15.2. Informative References
Acknowledgements
Authors' Addresses
1. Introduction
The Real-time Transport Protocol (RTP) [RFC3550] provides a framework
for delivery of audio and video teleconferencing data and other real-
time media applications. Previous work has defined the RTP protocol,
along with numerous profiles, payload formats, and other extensions.
When combined with appropriate signaling, these form the basis for
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