Intent to Prototype: RTPTransport WebRTC API
Tony Herre
Contact emails
he...@google.comExplainer
https://github.com/w3c/webrtc-rtptransport/blob/main/explainer-use-case-1.mdhttps://github.com/w3c/webrtc-rtptransport/blob/main/explainer-use-case-2.md
Specification
NoneSummary
An API allowing web apps to directly interact with the RTP packets sent and received by WebRTC PeerConnections, allowing customised encoders, packetisers and congestion control mechanisms.
Blink component
Blink>WebRTCMotivation
WebRTC PeerConnections are a blackbox, provided with raw media and externally configured to encode & send / receive & decode audio and video tracks, applying built-in generic congestion control algorithms and RTP serialization. There's significant interest to be able to integrate external components which can be tuned to the specific usecase of an app and developed independently: - using alternative bandwidth estimation algorithms run in JS or WASM - encoding media using new technologies such as WebCodecs or encoders compiled to WASM etc These are explored in the explainers: "Use Case 1" regarding external encoders & packetizers, "Use Case 2" regarding external congestion control / bandwidth estimation.
Initial public proposal
https://github.com/w3c/webrtc-rtptransportTAG review
NoneTAG review status
PendingRisks
Interoperability and Compatibility
None
Gecko: No signal
WebKit: No signal
Web developers: No signals
Other signals:
WebView application risks
Does this intent deprecate or change behavior of existing APIs, such that it has potentially high risk for Android WebView-based applications?
None
Debuggability
None
Is this feature fully tested by web-platform-tests?
NoFlag name on chrome://flags
NoneFinch feature name
NoneNon-finch justification
NoneRequires code in //chrome?
FalseEstimated milestones
No milestones specified