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 Internet Engineering Task Force (IETF)
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Stream WG state Submitted to IESG for Publication
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IESG IESG state RFC 8834 (Proposed Standard)
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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


   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

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
   ( 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
   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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