Skip to content

Repository for the RTPTransport specification of the WebRTC Working Group

License

Notifications You must be signed in to change notification settings

w3c/webrtc-rtptransport

Repository files navigation

WebRTC-RtpTransport

A proposed API that allows web applications to send and receive packets using the RTP/RTCP protocol, defined in RFC 3550.

The WebRTC-RtpTransport API is compatible with existing WebRTC APIs, including WebRTC-PC (RTCPeerConnection) and WebRTC Encoded Transform, and can be combined with WebCodecs. This allows applications to leverage existing APIs, simplifying the transition, while allowing applications to decide which pipeline stages to replace or keep.

The WebRTC-RtpTransport API enables web applications to support:

  • Custom payloads (ML-based audio codecs)
  • Custom packetization
  • Custom FEC
  • Custom RTX
  • Custom Jitter Buffer
  • Custom bandwidth estimate
  • Custom rate control (with built-in bandwidth estimate)
  • Custom bitrate allocation
  • Custom metadata (header extensions)
  • Custom RTCP messages
  • Custom RTCP message timing
  • RTP forwarding

For more information

See the explainer for more info.

See the Custom Packetization Use Case for some API info.

See the API outline

See the Specification

About

Repository for the RTPTransport specification of the WebRTC Working Group

Resources

License

Code of conduct

Stars

Watchers

Forks

Packages

No packages published