Meeting.js » A WebRTC Library for Media Streaming

<script src="https://webrtc.github.io/adapter/adapter-latest.js"></script>
<script src="https://www.webrtc-experiment.com/CodecsHandler.js"></script>
<script src="https://www.webrtc-experiment.com/IceServersHandler.js"></script>
<script src="https://www.webrtc-experiment.com/meeting.js"></script>

var meeting = new Meeting('meeting-unique-id');

// on getting local or remote streams
meeting.onaddstream = function(e) {
    // e.type == 'local' ---- it is local media stream
    // e.type == 'remote' --- it is remote media stream
    document.body.appendChild(e.video);
};

// custom signaling channel
// you can use each and every signaling channel
// any websocket, socket.io, or XHR implementation
// any SIP server
// anything! etc.
meeting.openSignalingChannel = function(callback) {
    return io.connect().on('message', callback);
};

// check pre-created meeting rooms
// it is useful to auto-join
// or search pre-created sessions
meeting.check('meeting room name');

document.getElementById('setup-new-meeting').onclick = function() {
    meeting.setup('meeting room name');
};

// if someone leaves; just remove his video
meeting.onuserleft = function(userid) {
    var video = document.getElementById(userid);
    if(video) video.parentNode.removeChild(video);
};

// to leave a meeting room
meeting.leave();

// removing existing bandwidth lines
sdp = sdp.replace( /b=AS([^\r\n]+\r\n)/g , '');

// setting "outgoing" audio RTP port's bandwidth to "50kbit/s"
sdp = sdp.replace( /a=mid:audio\r\n/g , 'a=mid:audio\r\nb=AS:50\r\n');

// setting "outgoing" video RTP port's bandwidth to "256kbit/s"
sdp = sdp.replace( /a=mid:video\r\n/g , 'a=mid:video\r\nb=AS:256\r\n');