WebRTC pause and resume stream
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.
You should try using renegotiation, I believe the difference still exists how it is done in chrome and firefox:
In chrome, you just call
addStreamorremoveStreamon thePeerConnectionobject to add/ remove the stream, then create and exchangesdp.In firefox, there is no direct
removeStream, you need to use RTCRtpSender andaddTrackandremoveTrackmethods, you can take a look at this question