Class WebRTCEndpoint<EndpointMetadata, TrackMetadata>

Adds track that will be sent to the RTC Engine.

Example

let localStream: MediaStream = new MediaStream();
try {
  localAudioStream = await navigator.mediaDevices.getUserMedia(
    AUDIO_CONSTRAINTS
  );
  localAudioStream
    .getTracks()
    .forEach((track) => localStream.addTrack(track));
} catch (error) {
  console.error("Couldn't get microphone permission:", error);
}

try {
  localVideoStream = await navigator.mediaDevices.getUserMedia(
    VIDEO_CONSTRAINTS
  );
  localVideoStream
    .getTracks()
    .forEach((track) => localStream.addTrack(track));
} catch (error) {
 console.error("Couldn't get camera permission:", error);
}

localStream
 .getTracks()
 .forEach((track) => webrtc.addTrack(track, localStream));
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Cleans up WebRTCEndpoint instance.

Tries to connect to the RTC Engine. If user is successfully connected then WebRTCEndpointEvents.connected will be emitted.

Example

let webrtc = new WebRTCEndpoint();
webrtc.connect({displayName: "Bob"});
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Disables track encoding so that it will be no longer sent to the server.

Example

const trackId = webrtc.addTrack(track, stream, {}, {enabled: true, activeEncodings: ["l", "m", "h"]});
webrtc.disableTrackEncoding(trackId, "l");
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Disconnects from the room. This function should be called when user disconnects from the room in a clean way e.g. by clicking a dedicated, custom button disconnect. As a result there will be generated one more media event that should be sent to the RTC Engine. Thanks to it each other endpoint will be notified that endpoint was removed in WebRTCEndpointEvents.endpointRemoved,

Enables track encoding so that it will be sent to the server.

Example

const trackId = webrtc.addTrack(track, stream, {}, {enabled: true, activeEncodings: ["l", "m", "h"]});
webrtc.disableTrackEncoding(trackId, "l");
// wait some time
webrtc.enableTrackEncoding(trackId, "l");
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Returns a snapshot of currently received remote endpoints.

Returns a snapshot of currently received remote tracks.

Example

if (webRTCEndpoint.getRemoteTracks()[trackId]?.simulcastConfig?.enabled) {
  webRTCEndpoint.setTargetTrackEncoding(trackId, encoding);
}
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Retrieves statistics related to the RTCPeerConnection. These statistics provide insights into the performance and status of the connection.

External

RTCPeerConnection#getStats()

See

MDN Web Docs: RTCPeerConnection.getStats()

Feeds media event received from RTC Engine to WebRTCEndpoint. This function should be called whenever some media event from RTC Engine was received and can result in WebRTCEndpoint generating some other media events.

Example

This example assumes phoenix channels as signalling layer. As phoenix channels require objects, RTC Engine encapsulates binary data into map with one field that is converted to object with one field on the TS side.

webrtcChannel.on("mediaEvent", (event) => webrtc.receiveMediaEvent(event.data));
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Removes a track from connection that was sent to the RTC Engine.

Example

// setup camera
let localStream: MediaStream = new MediaStream();
try {
  localVideoStream = await navigator.mediaDevices.getUserMedia(
    VIDEO_CONSTRAINTS
  );
  localVideoStream
    .getTracks()
    .forEach((track) => localStream.addTrack(track));
} catch (error) {
  console.error("Couldn't get camera permission:", error);
}

let trackId
localStream
 .getTracks()
 .forEach((track) => trackId = webrtc.addTrack(track, localStream));

// remove track
webrtc.removeTrack(trackId)
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Replaces a track that is being sent to the RTC Engine.

Example

// setup camera
let localStream: MediaStream = new MediaStream();
try {
  localVideoStream = await navigator.mediaDevices.getUserMedia(
    VIDEO_CONSTRAINTS
  );
  localVideoStream
    .getTracks()
    .forEach((track) => localStream.addTrack(track));
} catch (error) {
  console.error("Couldn't get camera permission:", error);
}
let oldTrackId;
localStream
 .getTracks()
 .forEach((track) => trackId = webrtc.addTrack(track, localStream));

// change camera
const oldTrack = localStream.getVideoTracks()[0];
let videoDeviceId = "abcd-1234";
navigator.mediaDevices.getUserMedia({
     video: {
       ...(VIDEO_CONSTRAINTS as {}),
       deviceId: {
         exact: videoDeviceId,
       },
     }
  })
  .then((stream) => {
    let videoTrack = stream.getVideoTracks()[0];
    webrtc.replaceTrack(oldTrackId, videoTrack);
  })
  .catch((error) => {
    console.error('Error switching camera', error);
  })
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Updates maximum bandwidth for the given simulcast encoding of the given track.

Sets track variant that server should send to the client library.

The variant will be sent whenever it is available. If chosen variant is temporarily unavailable, some other variant will be sent until the chosen variant becomes active again.

Example

webrtc.setTargetTrackEncoding(incomingTrackCtx.trackId, "l")
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Updates maximum bandwidth for the track identified by trackId. This value directly translates to quality of the stream and, in case of video, to the amount of RTP packets being sent. In case trackId points at the simulcast track bandwidth is split between all of the variant streams proportionally to their resolution.

Updates the metadata for the current endpoint.

Updates the metadata for a specific track.