WebRTC One-to-Many Audio-Broadcasting / Source Code

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Setup a new meeting:

Local Media Stream

Remote Media Streams

If 10 users join your broadcasted room, 20 RTP ports will be opened on your browser:

  1. 10 RTP ports for outgoing audio streams
  2. 10 RTP ports for incoming audio streams

Difference between one-way broadcasting and one-to-many broadcasting

For 10 users session, maximum 10 RTP ports for outgoing audio stream will be opened.

On each participant's side; only one incoming RTP port will be opened.

Unlike one-way broadcasting; one-to-many broadcasting experiment opens both outgoing as well as incoming RTP ports for each participant.

<script src="https://www.webrtc-experiment.com/one-to-many-audio-broadcasting/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

// 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

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

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