Media Transport and Use of RTP in WebRTC
RFC 8834, “Media Transport and Use of RTP in WebRTC”, is a Proposed Standard document published in January 2021 by C. Perkins, M. Westerlund, J. Ott. The canonical text is published by the RFC Editor.
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.
What “Proposed Standard” means
An entry-level standards-track specification: stable, peer-reviewed and a solid basis for implementation, though it may still evolve before becoming an Internet Standard.
The canonical text of RFC 8834 is hosted at rfc-editor.org. Available in HTML,TXT,PDF,XML.
- RFC 8833 Application-Layer Protocol Negotiation for WebRTC
- RFC 8835 Transports for WebRTC
- RFC 8832 WebRTC Data Channel Establishment Protocol
- RFC 8836 Congestion Control Requirements for Interactive Real-Time Media
- RFC 8831 WebRTC Data Channels
- RFC 8837 Differentiated Services Code Point Packet Markings for WebRTC QoS
- RFC 8830 WebRTC MediaStream Identification in the Session Description Protocol
- RFC 8838 Trickle ICE: Incremental Provisioning of Candidates for the Interactive Connectivity Establishment Protocol