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Signaling and video calling - Web APIs
ement.
...connection) { alert("you can't start a call because you already have one open!"); } else { var clickedusername = evt.target.textcontent; if (clickedusername === myusername) { alert("i'm afraid i can't let you talk to yourself.
...g the stream with audio from the user's microphone and video from their webcam.
...s microphone and webcam to that.
Using DTMF with WebRTC - Web APIs
let's take a look at the javascript code next.
...audio to the connection.
... some browsers have not yet implemented rtcpeerconnection.addtrack(), therefore requiring the use of the obsolete addstream() method, we use this boolean to determine whether or not the user agent supports addtrack(); if it doesn't, we'll fall back to addstream().
...= handlecallersignalingstatechangeevent; callerpc.onicegatheringstatechange = handlecallergatheringstatechangeevent; receiverpc = new rtcpeerconnection(); receiverpc.onicecandidate = handlereceivericeevent; if (hasaddtrack) { receiverpc.ontrack = handlereceivertrackevent; } else { receiverpc.onaddstream = handlereceiveraddstreamevent; } navigator.mediadevices.getusermedia(
mediaconstraints) .then(gotstream) .catch(err => log(err.message)); } after creating the rtcpeerconnection for the caller (callerpc), we look to see if it has an addtrack() method.