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RTCTransport

A proposed API that allows web applications to directly send and receive network packets peer-to-peer, targeting realtime media and similar traffic.

The RTCTransport 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
  • Packet 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

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Repository for the RTPTransport specification of the WebRTC Working Group

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