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
See the explainer for more info.
See the Custom Packetization Use Case for some API info.
See the proposed spec.