RTCPeerConnection.addTrack()
The RTCPeerConnection
method
addTrack()
adds a new media track to the set of tracks
which will be transmitted to the other peer.>
Note: Adding a track to a connection triggers renegotiation by
firing a negotiationneeded
event. See
Starting negotiation in Signaling and video calling for details.
Syntax
rtpSender = rtcPeerConnection.addTrack(track, stream...);
Parameters
track
-
A
MediaStreamTrack
object representing the media track to add to the peer connection. stream...
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 stream
s. Instead, the stream
s 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.
Return value
The RTCRtpSender
object which will be used to transmit the media data.
Note: Every RTCRtpSender
is paired with an
RTCRtpReceiver
to make up an RTCRtpTransceiver
. The
associated receiver is muted (indicating that it is not able to deliver packets) until
and unless one or more streams are added to the receiver by the remote peer.
Exceptions
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.
Usage notes
Adding tracks to multiple streams
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 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 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.
Reused senders
This method may return either a new RTCRtpSender
or, under very specific
circumstances, an existing compatible sender which has not yet been used to transmit
data. Compatible reusable RTCRtpSender
instances meet these criteria:
- There is no track already associated with the sender.
-
The
RTCRtpTransceiver
associated with the sender has aRTCRtpReceiver
whosetrack
property specifies aMediaStreamTrack
whosekind
is the same as thekind
of thetrack
parameter specified when callingRTCPeerConnection.addTrack()
. This ensures that a transceiver only handles audio or video and never both. - The
RTCRtpTransceiver
'sstopped
property isfalse
. -
The
RTCRtpSender
being considered has never been used to send data. If the transceiver'scurrentDirection
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
'strack
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 itscurrentDirection
updated to include sending; if its current value is"recvonly"
, it becomes"sendrecv"
, and if its current value is"inactive"
, it becomes"sendonly"
.
New senders
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 specifiedtrack
and set ofstream
(s). -
A new
RTCRtpReceiver
is created with a newMediaStreamTrack
as itstrack
property (not the track specified as a parameter when callingaddTrack()
). This track'skind
is set to match thekind
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.
Example
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.
var mediaConstraints = {
audio: true, // We want an audio track
video: true, // ...and we want a video track
};
var desc = new RTCSessionDescription(sdp);
pc.setRemoteDescription(desc)
.then(function () {
return navigator.mediaDevices.getUserMedia(mediaConstraints);
})
.then(function (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.
Specifications
Specification |
---|
WebRTC 1.0: Real-Time Communication Between Browsers # dom-rtcpeerconnection-addtrack |
Browser compatibility
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