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Codec agnostic RTP payload format for video

Document Type Expired Internet-Draft (individual)
Expired & archived
Authors Sergio Garcia Murillo , Youenn Fablet , Dr. Alex Gouaillard
Last updated 2021-09-10 (Latest revision 2021-03-09)
Replaces draft-codec-agnostic-rtp-payload-format
RFC stream (None)
Intended RFC status (None)
Stream Stream state (No stream defined)
Consensus boilerplate Unknown
RFC Editor Note (None)
IESG IESG state Expired
Telechat date (None)
Responsible AD (None)
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This Internet-Draft is no longer active. A copy of the expired Internet-Draft is available in these formats:


RTP Media Chains usually rely on piping encoder output directly to packetizers. Media packetization formats often support a specific codec format and optimize RTP packets generation accordingly. With the development of Selective Forward Unit (SFU) solutions, that do not process media content server side, the need for media content processing at the origin and at the destination has arised. RTP Media Chains used e.g. in WebRTC solutions are increasingly relying on application-specific transforms that sit in-between encoder and packetizer on one end and in-between depacketizer and decoder on the other end. This use case has become so important, that the W3C is standardizing the capacity to access encoded content with the [WebRTCInsertableStreams] API proposal. An extremely popular use case is application level end-to-end encryption of media content, using for instance [SFrame]. Whatever the modification applied to the media content, RTP packetizers can no longer expect to use packetization formats that mandate media content to be in a specific codec format. In the extreme cases like encryption, where the RTP Payload is made completely opaque to the SFUs, some extra mechanism must also be added for them to be able to route the packets without depending on RTP payload or payload headers. The traditionnal process of creating a new RTP Payload specification per content would not be practical as we would need to make a new one for each codec-transform pair. This document describes a solution, which provides the following features in the case the encoded content has been modified before reaching the packetizer: - a paylaod agnostic RTP packetization format that can be used on any media content, - a negotiation mechanism for the above format and the inner payload, Both of the above mechanism are backward compatible with most of (S)RTP/RTCP mechanisms used for bandwidth estimation and congestion control in RTP/SRTP/webrtc, including but not limited to SSRC, RED, FEC, RTX, NACK, SR/RR, REMB, transport-wide-CC, TMBR, .... It as illustrated by existing implementations in chrome, safari, and Medooze. This document also describes a solution to allow SFUs to continue performing packet routing on top of this generic RTP packetization format. This document complements the SFrame (media encryption), and Dependency Descriptor (AV1 payload annex) documents to provide an End-to-End-Encryption solution that would sit on top of SRTP/Webrtc, use SFUs on the media back-end, and leverage W3C APIs in the browser. A high level description of such system will be provided as an informational I-D in the SFrame WG and then cited here.


Sergio Garcia Murillo
Youenn Fablet
Dr. Alex Gouaillard

(Note: The e-mail addresses provided for the authors of this Internet-Draft may no longer be valid.)