A11 TypeScript API - v0.2.1
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    Class WebRtcWireStream

    Client-side WireStream over a browser-compatible WebRTC data channel.

    WebRTC negotiation runs through a connected SignallingTransport; application messages then use the peer-to-peer data channel. The signalling endpoint must remain alive while ICE candidates and descriptions are being exchanged. A11 packetizes messages below SCTP limits and applies the same half-close, drain, abort, and deadline lifecycle as its WebSocket transport.

    Create the endpoint, call start, and await wait for full completion. The returned stream is client-side; accepting server-side data channels is handled by the native service transport.

    Implements

    Index
    • get dataChannel(): RTCDataChannel

      Underlying data channel for advanced browser integration.

      Returns RTCDataChannel

    • get peerConnection(): RTCPeerConnection

      Underlying peer connection for diagnostics and browser statistics.

      Returns RTCPeerConnection