RFC 8834: Media Transport and Use of RTP in WebRTC
In plain English — editorial summary, not part of the RFC
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.
Document record
- Document ID
- RFC8834
- Published
- January 2021
- Authors
- C. Perkins; M. Westerlund; J. Ott
- Status
- PROPOSED STANDARD
- Stream
- IETF
- Area
- wit
- Pages
- 39
- Also known as
- —
Related documents
Ranked automatically by shared keywords, IETF area and stream — not by editorial selection.
- RFC 8833Application-Layer Protocol Negotiation (ALPN) for WebRTCCurrent
January 2021
- RFC 8835Transports for WebRTCCurrent
January 2021
- RFC 8832WebRTC Data Channel Establishment ProtocolCurrent
January 2021
- RFC 8837Differentiated Services Code Point (DSCP) Packet Markings for WebRTC QoSCurrent
January 2021
- RFC 8831WebRTC Data ChannelsCurrent
January 2021
- RFC 8829JavaScript Session Establishment Protocol (JSEP)Obsoleted
January 2021
- RFC 8828WebRTC IP Address Handling RequirementsCurrent
January 2021
- RFC 8827WebRTC Security ArchitectureCurrent
January 2021
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The document record above — title, authors, date, status, stream, area, relationships, DOI and errata — is imported verbatim from the public RFC Editor index. The “in plain English” section is editorial: written by The metasystema editorial team, not part of the RFC. Where the two differ, the RFC text governs.
Last checked against the RFC Editor index on . RFCs are never revised after publication; changes are issued as new documents.
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