The RTCPeerConnection method addTrack() adds a new media track to the set of tracks which will be transmitted to the other peer.

track

A MediaStreamTrack object representing the media track to add to the peer connection.

stream1, …, streamN Optional

One or more local MediaStream objects to which the track should be added.

The specified track doesn't necessarily have to already be part of any of the specified streams. Instead, the streams are a way to group tracks together on the receiving end of the connection, making sure they are synchronized. Any tracks that are added to the same stream on the local end of the connection will be on the same stream on the remote end.

The RTCRtpSender object which will be used to transmit the media data.

InvalidAccessError DOMException

Thrown if the specified track (or all of its underlying streams) is already part of the RTCPeerConnection.

InvalidStateError DOMException

Thrown if the RTCPeerConnection is closed.

After the track parameter, you can optionally specify one or more MediaStream objects to add the track to. Only tracks are sent from one peer to another, not streams. Since streams are specific to each peer, specifying one or more streams means the other peer will create a corresponding stream (or streams) automatically on the other end of the connection, and will then automatically add the received track to those streams.

Streamless tracks

If no streams are specified, then the track is streamless. This is perfectly acceptable, although it will be up to the remote peer to decide what stream to insert the track into, if any. This is a very common way to use addTrack() when building many types of simple applications, where only one stream is needed. For example, if all you're sharing with the remote peer is a single stream with an audio track and a video track, you don't need to deal with managing what track is in what stream, so you might as well just let the transceiver handle it for you.

Here's an example showing a function that uses getUserMedia() to obtain a stream from a user's camera and microphone, then adds each track from the stream to the peer connection, without specifying a stream for each track:

async function openCall(pc) {
  const gumStream = await navigator.mediaDevices.getUserMedia({
    video: true,
    audio: true,
  });
  for (const track of gumStream.getTracks()) {
    pc.addTrack(track);
  }
}

The result is a set of tracks being sent to the remote peer, with no stream associations. The handler for the track event on the remote peer will be responsible for determining what stream to add each track to, even if that means adding them all to the same stream. The ontrack handler might look like this:

let inboundStream = null;

pc.ontrack = (ev) => {
  if (ev.streams && ev.streams[0]) {
    videoElem.srcObject = ev.streams[0];
  } else {
    if (!inboundStream) {
      inboundStream = new MediaStream();
      videoElem.srcObject = inboundStream;
    }
    inboundStream.addTrack(ev.track);
  }
};

Here, the track event handler adds the track to the first stream specified by the event, if a stream is specified. Otherwise, the first time ontrack is called, a new stream is created and attached to the video element, and then the track is added to the new stream. From then on, new tracks are added to that stream.

You could also just create a new stream for each track received:

pc.ontrack = (ev) => {
  if (ev.streams && ev.streams[0]) {
    videoElem.srcObject = ev.streams[0];
  } else {
    let inboundStream = new MediaStream(ev.track);
    videoElem.srcObject = inboundStream;
  }
};

Associating tracks with specific streams

By specifying a stream and allowing RTCPeerConnection to create streams for you, the streams' track associations are automatically managed for you by the WebRTC infrastructure. This includes things like changes to the transceiver's direction and tracks being halted using removeTrack().

For example, consider this function that an application might use to begin streaming a device's camera and microphone input over an RTCPeerConnection to a remote peer:

async function openCall(pc) {
  const gumStream = await navigator.mediaDevices.getUserMedia({
    video: true,
    audio: true,
  });
  for (const track of gumStream.getTracks()) {
    pc.addTrack(track, gumStream);
  }
}

The remote peer might then use a track event handler that looks like this:

pc.ontrack = ({ streams: [stream] }) => (videoElem.srcObject = stream);

This sets the video element's current stream to the one that contains the track that's been added to the connection.

This method returns a new RTCRtpSender or an existing instance for reuse. An RTCRtpSender instance is only compatible for reuse if it meets the following criteria:

• There is no track already associated with the sender.
• The RTCRtpTransceiver associated with the sender has a RTCRtpReceiver whose track property specifies a MediaStreamTrack whose kind is the same as the kind of the track parameter specified when calling RTCPeerConnection.addTrack(). This ensures that a transceiver only handles audio or video and never both.
• The RTCRtpTransceiver.currentDirection property is not "stopped".
• The RTCRtpSender being considered has never been used to send data. If the transceiver's currentDirection has ever been "sendrecv" or "sendonly", the sender can't be reused.

If all of those criteria are met, the sender gets reused, which results in these changes occurring to the existing RTCRtpSender and its RTCRtpTransceiver:

• The RTCRtpSender's track is set to the specified track.
• The sender's set of associated streams is set to the list of streams passed into this method, stream....
• The associated RTCRtpTransceiver has its currentDirection updated to indicate that it is sending; if its current value is "recvonly", it becomes "sendrecv", and if its current value is "inactive", it becomes "sendonly".

If no existing sender exists that can be reused, a new one is created. This also results in the creation of the associated objects that must exist. The process of creating a new sender results in these changes:

• The new RTCRtpSender is created with the specified track and set of stream(s).
• A new RTCRtpReceiver is created with a new MediaStreamTrack as its track property (not the track specified as a parameter when calling addTrack()). This track's kind is set to match the kind of the track provided as an input parameter.
• A new RTCRtpTransceiver is created and associated with the new sender and receiver.
• The new transceiver's direction is set to "sendrecv".
• The new transceiver is added to the RTCPeerConnection's set of transceivers.

This example is drawn from the code presented in the article Signaling and video calling and its corresponding sample code. It comes from the handleVideoOfferMsg() method there, which is called when an offer message is received from the remote peer.

const mediaConstraints = {
  audio: true, // We want an audio track
  video: true, // And we want a video track
};

const desc = new RTCSessionDescription(sdp);

pc.setRemoteDescription(desc)
  .then(() => navigator.mediaDevices.getUserMedia(mediaConstraints))
  .then((stream) => {
    previewElement.srcObject = stream;

    stream.getTracks().forEach((track) => pc.addTrack(track, stream));
  });

This code takes SDP which has been received from the remote peer and constructs a new RTCSessionDescription to pass into setRemoteDescription(). Once that succeeds, it uses MediaDevices.getUserMedia() to obtain access to the local webcam and microphone.

If that succeeds, the resulting stream is assigned as the source for a <video> element which is referenced by the variable previewElement.

The final step is to begin sending the local video across the peer connection to the caller. This is done by adding each track in the stream by iterating over the list returned by MediaStream.getTracks() and passing them to addTrack() along with the stream which they're a component of.

• WebRTC
• Introduction to the Real-time Transport Protocol (RTP)
• track