WebRTC: Connecting multiple listeners to one client, one at a time - html

Question

I have been trying to get WebRTC to function with a broadcaster and multiple listeners but am stuck when it comes to transferal descriptions and candidates via signalling (with nodejs & socket.io).
I can get the process working between two browsers with a simple nodejs socket app which simply broadcasts the descriptions and candidates to other already connected clients, but when I attempt to store a description and connect with a newly opened browser, nothing happens.
What I basically need to understand is what do I need to provide to one browser, in order for it to begin communicating with another? The project I am working on requires the ability for listeners to join rooms, authenticate, and begin listening to whatever media is being sent.
Below is my client side code:
var audioContext = new webkitAudioContext()
var client = null
var configuration =
{
'iceServers':
[{
'url': 'stun:stun.example.org'
}]
}
$(function ()
{
window.RTCPeerConnection = window.RTCPeerConnection || window.webkitRTCPeerConnection || window.mozRTCPeerConnection
client = new RTCPeerConnection(configuration, { optional:[ { RtpDataChannels: true } ]})
client.onnegotiationneeded = function ()
{
console.log('Negotiation needed')
createOffer()
}
client.onicecandidate = function (event)
{
console.log('onicecandidate')
socket.emit('candidate', JSON.stringify({ 'candidate': event.candidate }))
}
client.onaddstream = function (event)
{
console.log('onaddstream')
$('#player').attr('src', URL.createObjectURL(event.stream))
player.play()
}
socket.on('candidate', function (event)
{
candidate(event)
})
socket.on('description', function (message)
{
if(!client) { return }
client.setRemoteDescription(new RTCSessionDescription(message.sdp), function () {
if (client.remoteDescription.type == 'offer')
client.createAnswer(function (description)
{
client.setLocalDescription(description, function ()
{
socket.emit('description', JSON.stringify({ 'sdp':client.localDescription }))
})
}, function (err)
{
console.log('error: ' + err)
})
}, function(err)
{
console.log('error: ' + err)
})
})
addStream()
})
function createOffer ()
{
if(!client) { return; }
client.createOffer(function (description)
{
console.log(description)
client.setLocalDescription(description, function ()
{
socket.emit('description', JSON.stringify({ 'sdp': client.localDescription }))
console.log('set local description')
})
})
}
function candidate (message)
{
if(message.candidate)
{
console.log('candidate')
client.addIceCandidate(new RTCIceCandidate(message.candidate))
}
}
function addStream ()
{
navigator.webkitGetUserMedia({audio: true, video: false}, function(stream)
{
client.addStream(stream)
})
}
And my signalling part of my server as it currently stands:
io.on 'connection', (socket) ->
socket.on 'description', (data) ->
parsed = JSON.parse data
socket.broadcast.emit 'description', parsed
socket.on 'candidate', (candidate) ->
parsed = JSON.parse candidate
socket.broadcast.emit 'candidate', parsed
I'd appreciate any insight into this. Thanks.

Answer 1

The "PeerConnection" as the name indicates can be used with only one other peer. You cannot cache the offer SDP generated by one PeerConnection instance to use it with more than one other peers.
In your case, you must create a PeerConnection for each browser that you want to send/receive audio and video from and then exchange the corresponding SDP offer and answers with those browsers via your signaling mechanism.
Please feel free to go through some of the links I have mentioned here to understand how WebRTC works.