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WebRTC pause and resume stream

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I am trying to use WebRTC to build a web application that needs to pause/resume the video/audio stream when some events trigger. I have tried the getTracks()[0].stop() but I am not sure how to resume the stream. Are there any advice on that? thanks

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Ev3rlasting Avatar asked Mar 08 '16 02:03

Ev3rlasting


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How do I pause a Webrtc stream?

In chrome, you just call addStream or removeStream on the PeerConnection object to add/ remove the stream, then create and exchange sdp .

What is MediaStreamTrack?

The MediaStreamTrack interface represents a single media track within a stream; typically, these are audio or video tracks, but other track types may exist as well.


2 Answers

getTracks()[0].stop() is permanent.

Use getTracks()[0].enabled = false instead. To unpause getTracks()[0].enabled = true.

This will replace your video with black, and your audio with silence.

Try it (use https fiddle for Chrome):

var pc1 = new RTCPeerConnection(), pc2 = new RTCPeerConnection();    navigator.mediaDevices.getUserMedia({ video: true, audio: true })    .then(stream => pc1.addStream(video1.srcObject = stream))    .catch(log);    var mute = () => video1.srcObject.getTracks().forEach(t => t.enabled = !t.enabled);    var add = (pc, can) => can && pc.addIceCandidate(can).catch(log);  pc1.onicecandidate = e => add(pc2, e.candidate);  pc2.onicecandidate = e => add(pc1, e.candidate);    pc2.onaddstream = e => video2.srcObject = e.stream;  pc1.onnegotiationneeded = e =>    pc1.createOffer().then(d => pc1.setLocalDescription(d))    .then(() => pc2.setRemoteDescription(pc1.localDescription))    .then(() => pc2.createAnswer()).then(d => pc2.setLocalDescription(d))    .then(() => pc1.setRemoteDescription(pc2.localDescription))    .catch(log);    var log = msg => div.innerHTML += "<br>" + msg;
<video id="video1" height="120" width="160" autoplay muted></video>  <video id="video2" height="120" width="160" autoplay></video><br>  <input type="checkbox" onclick="mute()">mute</input><div id="div"></div>  <script src="https://webrtc.github.io/adapter/adapter-latest.js"></script>

PeerConnections basically stop sending packets in this muted state, so it is highly efficient.

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jib Avatar answered Sep 29 '22 23:09

jib


You should try using renegotiation, I believe the difference still exists how it is done in chrome and firefox:

  • In chrome, you just call addStream or removeStream on the PeerConnection object to add/ remove the stream, then create and exchange sdp.

  • In firefox, there is no direct removeStream, you need to use RTCRtpSender and addTrack and removeTrack methods, you can take a look at this question

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mido Avatar answered Sep 29 '22 23:09

mido