WebRTC pause and resume stream

Question

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

jib · Accepted Answer · 2016-08-14T03:19:10.383

40

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.

edited Aug 14 '16 at 03:19

answered Mar 08 '16 at 16:05

jib

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The last sentence is unclear. Will "black video" and "silence" still use similar resources as before disabled (decoding/rendering, data transmission, etc)? – JSON Aug 13 '16 at 23:34
@JSON I've updated the answer to clarify that it hardly uses any resources. I confirmed it using stats in [this fiddle](https://jsfiddle.net/ec9ossmu/) (requires Firefox for spec-stats atm). Thanks for catching that it was unclear. – jib Aug 14 '16 at 03:18
Can you explain what happens to bandwidth? – Javad Asoodeh Sep 21 '20 at 08:37
I think i still receive track, but how can pause a particular transreceiver.sender(Audio/Video) from remote side with negotiation and without it ? – Javad Asoodeh Sep 21 '20 at 08:51
I changed direction attr from receiver side to 'inactive' and On the receiver side, the bandwidth decreased, do you think it's true way for pause audio/video ? – Javad Asoodeh Sep 21 '20 at 11:07
1

@J4v4d Right, to pause from the remote end is a different question, and changing `direction` is one way to go there. See [my blog post](https://blog.mozilla.org/webrtc/rtcrtptransceiver-explored/) for the myriad ways to control a track remotely. – jib Sep 21 '20 at 16:54
is there any way out for just pausing the steaming simply instead of receiving blank frames , because I am sending those frames to the backend for processing – Atif Shafi Oct 22 '21 at 12:54

score 1 · Answer 2 · edited May 23 '17 at 12:24

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

WebRTC pause and resume stream

2 Answers2

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