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WebRTC connectionState stuck at "new" - Safari only, works in Chrome and FF

I'm having trouble connecting to my local peer as remote with WebRTC video and audio. This issue is only happening in Safari on desktop and iOS. On Chrome and Firefox the issue is non-existant.
I'm assuming it has something to do with the fact that in Safari, it always asks if you want to allow audio/video but I'm not sure. That's just the only difference I can make out between the browsers. Even after selecting 'allow', the issue persists.
Reproduction steps:
In Chrome, open the initial local connection with audio/video
In Safari, open the remote connection and choose to enable audio/video
Result:
Local connection never makes an offer and the connectionState of the remote (Safari) gets stuck as new. See the following RTCPeerConnection object:
Here is the exact same object via the exact same steps, but in Chrome or Firefox:
Edit:
After more testing, I've found the following:
Below format: (First Connection) > (Second Connection)
Chrome > Chrome: Works
Chrome > Firefox: Works
Chrome > Safari: Doesn't work
Safari > Chrome: Works
Safari > Safari: Works
The issue doesn't seem to exist when using Safari for both sides of the connection...only when Safari is used as the secondary connection.
Here is my code:
import h from './helpers.js';
document.getElementById('close-chat').addEventListener('click', (e) => {
document.querySelector('#right').style.display = "none";
});
document.getElementById('open-chat').addEventListener('click', (e) => {
document.querySelector('#right').style.display = "flex";
});
window.addEventListener('load', () => {
sessionStorage.setItem('connected', 'false');
const room = h.getParam('room');
const user = h.getParam('user');
sessionStorage.setItem('username', user);
const username = sessionStorage.getItem('username');
if (!room) {
document.querySelector('#room-create').attributes.removeNamedItem('hidden');
}
else if (!username) {
document.querySelector('#username-set').attributes.removeNamedItem('hidden');
}
else {
let commElem = document.getElementsByClassName('room-comm');
for (let i = 0; i < commElem.length; i++) {
commElem[i].attributes.removeNamedItem('hidden');
}
var pc = [];
let socket = io('/stream');
var socketId = '';
var myStream = '';
var screen = '';
// Get user video by default
getAndSetUserStream();
socket.on('connect', () => {
console.log('Connected');
sessionStorage.setItem('remoteConnected', 'false');
h.connectedChat();
setTimeout(h.establishingChat, 3000);
setTimeout(h.oneMinChat, 60000);
setTimeout(h.twoMinChat, 120000);
setTimeout(h.threeMinChat, 180000);
setTimeout(h.fourMinChat, 240000);
setTimeout(h.fiveMinChat, 300000);
// Set socketId
socketId = socket.io.engine.id;
socket.emit('subscribe', {
room: room,
socketId: socketId
});
socket.on('new user', (data) => {
// OG user gets log when new user joins here.
console.log('New User');
console.log(data);
socket.emit('newUserStart', { to: data.socketId, sender: socketId });
pc.push(data.socketId);
init(true, data.socketId);
});
socket.on('newUserStart', (data) => {
console.log('New User Start');
console.log(data);
pc.push(data.sender);
init(false, data.sender);
});
socket.on('ice candidates', async (data) => {
console.log('Ice Candidates:');
console.log(data);
data.candidate ? await pc[data.sender].addIceCandidate(new RTCIceCandidate(data.candidate)) : '';
});
socket.on('sdp', async (data) => {
console.log('SDP:');
console.log(data);
if (data.description.type === 'offer') {
data.description ? await pc[data.sender].setRemoteDescription(new RTCSessionDescription(data.description)) : '';
h.getUserFullMedia().then(async (stream) => {
if (!document.getElementById('local').srcObject) {
h.setLocalStream(stream);
}
// Save my stream
myStream = stream;
stream.getTracks().forEach((track) => {
pc[data.sender].addTrack(track, stream);
});
let answer = await pc[data.sender].createAnswer();
await pc[data.sender].setLocalDescription(answer);
socket.emit('sdp', { description: pc[data.sender].localDescription, to: data.sender, sender: socketId });
}).catch((e) => {
console.error(e);
});
}
else if (data.description.type === 'answer') {
await pc[data.sender].setRemoteDescription(new RTCSessionDescription(data.description));
}
});
socket.on('chat', (data) => {
h.addChat(data, 'remote');
});
});
function getAndSetUserStream() {
console.log('Get and set user stream.');
h.getUserFullMedia({ audio: true, video: true }).then((stream) => {
// Save my stream
myStream = stream;
h.setLocalStream(stream);
}).catch((e) => {
console.error(`stream error: ${e}`);
});
}
function sendMsg(msg) {
let data = {
room: room,
msg: msg,
sender: username
};
// Emit chat message
socket.emit('chat', data);
// Add localchat
h.addChat(data, 'local');
}
function init(createOffer, partnerName) {
console.log('P1:');
console.log(partnerName);
pc[partnerName] = new RTCPeerConnection(h.getIceServer());
console.log('P2:');
console.log(pc[partnerName]);
if (screen && screen.getTracks().length) {
console.log('Screen:');
console.log(screen);
screen.getTracks().forEach((track) => {
pc[partnerName].addTrack(track, screen); // Should trigger negotiationneeded event
});
}
else if (myStream) {
console.log('myStream:');
console.log(myStream);
myStream.getTracks().forEach((track) => {
pc[partnerName].addTrack(track, myStream); // Should trigger negotiationneeded event
});
}
else {
h.getUserFullMedia().then((stream) => {
console.log('Stream:');
console.log(stream);
// Save my stream
myStream = stream;
stream.getTracks().forEach((track) => {
console.log('Tracks:');
console.log(track);
pc[partnerName].addTrack(track, stream); // Should trigger negotiationneeded event
});
h.setLocalStream(stream);
}).catch((e) => {
console.error(`stream error: ${e}`);
});
}
// Create offer
if (createOffer) {
console.log('Create Offer');
pc[partnerName].onnegotiationneeded = async () => {
let offer = await pc[partnerName].createOffer();
console.log('Offer:');
console.log(offer);
await pc[partnerName].setLocalDescription(offer);
console.log('Partner Details:');
console.log(pc[partnerName]);
socket.emit('sdp', { description: pc[partnerName].localDescription, to: partnerName, sender: socketId });
};
}
// Send ice candidate to partnerNames
pc[partnerName].onicecandidate = ({ candidate }) => {
console.log('Send ICE Candidates:');
console.log(candidate);
socket.emit('ice candidates', { candidate: candidate, to: partnerName, sender: socketId });
};
// Add
pc[partnerName].ontrack = (e) => {
console.log('Adding partner video...');
let str = e.streams[0];
if (document.getElementById(`${partnerName}-video`)) {
document.getElementById(`${partnerName}-video`).srcObject = str;
}
else {
// Video elem
let newVid = document.createElement('video');
newVid.id = `${partnerName}-video`;
newVid.srcObject = str;
newVid.autoplay = true;
newVid.className = 'remote-video';
newVid.playsInline = true;
newVid.controls = true;
// Put div in main-section elem
document.getElementById('left').appendChild(newVid);
const video = document.getElementsByClassName('remote-video');
}
};
pc[partnerName].onconnectionstatechange = (d) => {
console.log('Connection State:');
console.log(pc[partnerName].iceConnectionState);
switch (pc[partnerName].iceConnectionState) {
case 'new':
console.log('New connection...!');
break;
case 'checking':
console.log('Checking connection...!');
break;
case 'connected':
console.log('Connected with dispensary!');
sessionStorage.setItem('remoteConnected', 'true');
h.establishedChat();
break;
case 'disconnected':
console.log('Disconnected');
sessionStorage.setItem('connected', 'false');
sessionStorage.setItem('remoteConnected', 'false');
h.disconnectedChat();
h.closeVideo(partnerName);
break;
case 'failed':
console.log('Failed');
sessionStorage.setItem('connected', 'false');
sessionStorage.setItem('remoteConnected', 'false');
h.disconnectedChat();
h.closeVideo(partnerName);
break;
case 'closed':
console.log('Closed');
sessionStorage.setItem('connected', 'false');
sessionStorage.setItem('remoteConnected', 'false');
h.disconnectedChat();
h.closeVideo(partnerName);
break;
}
};
pc[partnerName].onsignalingstatechange = (d) => {
switch (pc[partnerName].signalingState) {
case 'closed':
console.log("Signalling state is 'closed'");
h.closeVideo(partnerName);
break;
}
};
}
// Chat textarea
document.getElementById('chat-input').addEventListener('keypress', (e) => {
if (e.which === 13 && (e.target.value.trim())) {
e.preventDefault();
sendMsg(e.target.value);
setTimeout(() => {
e.target.value = '';
}, 50);
}
});
}
});

It would be helpful to see the console logs from a failed (stuck in the "new" state) Safari run.
One possibility is that Safari isn't doing the full ice candidate gathering. As Phillip Hancke noted, seeing the SDP would help figure out if that's happening. As would seeing the console logs. In the past, Safari has had various quirks and bugs related to candidate gathering.
One way to force Safari to gather candidates is to explicitly set offerToReceiveAudio and offerToReceiveVideo:
await pc[partnerName].createOffer({ offerToReceiveAudio: true, offerToReceiveVideo: true })

No sound when making outgoing call via JsSIP (Asterisk)

I'm trying to debug the existent system, where calls are made via Asteriks.
When accepting incoming calls, everything works fine, but on making outgoing call there is apparently no sound (but I accept 'addstream' event and attach stream to audio). Production code takes 500 lines, but this code does pretty much the same, but doesn't work as well
const socket = new JsSIP.WebSocketInterface('wss://callwss.agdevelopments.net');
socket.via_transport = 'wss';
const configuration = {
password: "SIP4003!",
realm: "callws,s.agdevelopments.net",
register: true,
session_timers: false,
uri: "sip:4003#callwss.agdevelopments.net",
sockets: [socket]
}
const ua = new JsSIP.UA(configuration)
// Setup events
ua.on('connected', function () {
console.log('Connected')
})
ua.on('disconnected', function () {
console.log('Connected')
})
// Make a call
const eventHandlers = {
'progress': function (e) {
console.log('call is in progress');
},
'failed': function (e) {
console.log('call failed with cause: ' + (e.data ? e.data.cause : 'no cause'), e);
},
'ended': function (e) {
console.log('call ended with cause: ' + (e.data ? e.data.cause : 'no cause'), e);
},
'confirmed': function (e) {
console.log('call confirmed');
},
'addstream': (e) => {
console.log('Add stream (event handlers)')
audio.srcObject = e.stream
audio.play()
}
};
const options = {
'eventHandlers': eventHandlers,
'mediaConstraints': {'audio': true, 'video': false}
};
const audio = new window.Audio()
ua.on('registered', function () {
const session = ua.call('0513887341', options)
if (session.connection) {
console.log('Connection is valid')
session.connection.addEventListener('addstream', e => {
console.log('Add stream')
audio.srcObject = e.stream
audio.play()
})
session.on('addstream', function(e){
// set remote audio stream (to listen to remote audio)
// remoteAudio is <audio> element on page
const remoteAudio = audio
remoteAudio.src = window.URL.createObjectURL(e.stream);
remoteAudio.play();
});
session.connection.addEventListener('peerconnection', e => {
console.log('Peer connection')
audio.srcObject = e.stream
audio.play()
})
} else {
console.log('Connection is null')
}
})
ua.on('newRTCSession', (data) => {
console.log('New RTC Session')
const session = data.session
session.on('addstream', function(e){
// set remote audio stream (to listen to remote audio)
// remoteAudio is <audio> element on page
const remoteAudio = audio
remoteAudio.src = window.URL.createObjectURL(e.stream);
remoteAudio.play();
});
})
ua.start()
Also attaching screenshots from Asterisk. The first one is outgoing call with no sound, the second is incoming with sound

The issue was solved from IT side. There was no problems in JsSIP or code

Remote video not display using webrtc scaledrone

Hi can someone check is there something wrong with my code cause it not displayig the remote video. The log always return 'Start web RTC as 'offerer'' for both local and remote video. And I think mostly the sample code almost similar with my code so I dont know. Much love appreciate if you guys can help me. Thanks!
if (!location.hash) {
location.hash = Math.floor(Math.random() * 0xFFFFFF).toString(16);
}
const roomHash = location.hash.substring(1);
//const roomHash = '5dfd9b';
console.log("ROOM ID: >> " +roomHash);
// TODO: Replace with your own channel ID
const drone = new ScaleDrone('AUpfMxm16E9bfdgA');
// Room name needs to be prefixed with 'observable-'
//const roomName = 'observable-' + roomHash;
const roomName = 'observable-testPHR';
const configuration = {
iceServers: [{
urls: 'stun:stun.l.google.com:19302'
}]
};
let room;
let pc;
function onSuccess() {
console.log("Connection sucess");
};
function onError(error) {
console.log("Connection failed!");
//console.error(error);
};
drone.on('open', error => {
if (error) {
console.log( " Error open drone >>");
return console.error(error);
}
console.log(" Drone open >>")
room = drone.subscribe(roomName);
room.on('open', error => {
if (error) {
onError(error);
}
});
// We're connected to the room and received an array of 'members'
// connected to the room (including us). Signaling server is ready.
room.on('members', members => {
console.log('MEMBERS', members);
// If we are the second user to connect to the room we will be creating the offer
const isOfferer = members.length === 2;
startWebRTC(isOfferer);
});
});
// Send signaling data via Scaledrone
function sendMessage(message) {
console.log("Sending signal via scaledrone >>");
drone.publish({
room: roomName,
message
});
}
function startWebRTC(isOfferer) {
console.log('Starting WebRTC in as', isOfferer ? 'offerer' : 'waiter');
pc = new RTCPeerConnection(configuration);
console.log(" Test A ");
// 'onicecandidate' notifies us whenever an ICE agent needs to deliver a
// message to the other peer through the signaling server
pc.onicecandidate = event => {
console.log("Send Message to Candidate");
if (event.candidate) {
sendMessage({'candidate': event.candidate});
}
};
console.log(" Test B ");
// If user is offerer let the 'negotiationneeded' event create the offer
if (isOfferer) {
console.log(" Create Offer ");
pc.onnegotiationneeded = () => {
pc.createOffer().then(localDescCreated).catch(onError);
}
}
console.log(" Test C ");
// When a remote stream arrives display it in the #remoteVideo element
pc.ontrack = event => {
console.log("Display remote video >>>")
const stream = event.streams[0];
console.log(" Stream : >>" +stream);
if (!remoteVideo.srcObject || remoteVideo.srcObject.id !== stream.id) {
remoteVideo.srcObject = stream;
}
};
console.log(" Test D ");
navigator.mediaDevices.getUserMedia({
audio: true,
video: true,
}).then(stream => {
console.log(" Display Local Video >> ");
// Display your local video in #localVideo element
localVideo.srcObject = stream;
// Add your stream to be sent to the conneting peer
stream.getTracks().forEach(track => pc.addTrack(track, stream));
}, onError);
console.log(" Test E ");
// Listen to signaling data from Scaledrone
room.on('data', (message, client) => {
// Message was sent by us
if (client.id === drone.clientId) {
return;
}
console.log(" Test F ");
if (message.sdp) {
// This is called after receiving an offer or answer from another peer
pc.setRemoteDescription(new RTCSessionDescription(message.sdp), () => {
// When receiving an offer lets answer it
if (pc.remoteDescription.type === 'offer') {
console.log(" Answer call ");
pc.createAnswer().then(localDescCreated).catch(onError);
}
}, onError);
} else if (message.candidate) {
// Add the new ICE candidate to our connections remote description
pc.addIceCandidate(
new RTCIceCandidate(message.candidate), onSuccess, onError
);
}
});
}
function localDescCreated(desc) {
pc.setLocalDescription(
desc,
() => sendMessage({'sdp': pc.localDescription}),
onError
);
}

How to send and receive desktop capture stream generated via getUsermedia()

I am making a Screen sharing app with WebRTC + Socket.io and stuck at a place.
Connected with two browser using WebRTC + Socket.io and can send text
I am taking support from codelab but it is not for stream.(If solution is based on this link then highly helpful)
How can I send getUserMedia() stream:
dataChannel.send(stream);
And receive same stream on channel.onmessage():
I am getting event.data as '[object MediaStream]' not stream.
channel.onmessage = function(event){
// unable to get correct stream
// event.data is "[object MediaStream]" in string
}
function createPeerConnection(isInitiator, config) {
console.log('Creating Peer connection as initiator?', isInitiator, 'config:', config);
peerConn = new RTCPeerConnection(config);
// send any ice candidates to the other peer
peerConn.onicecandidate = function (event) {
console.log('onIceCandidate event:', event);
if (event.candidate) {
sendMessage({
type: 'candidate',
label: event.candidate.sdpMLineIndex,
id: event.candidate.sdpMid,
candidate: event.candidate.candidate
});
} else {
console.log('End of candidates.');
}
};
if (isInitiator) {
console.log('Creating Data Channel');
dataChannel = peerConn.createDataChannel("screen");
onDataChannelCreated(dataChannel);
console.log('Creating an offer');
peerConn.createOffer(onLocalSessionCreated, logError);
} else {
peerConn.ondatachannel = function (event) {
console.log('ondatachannel:', event.channel);
dataChannel = event.channel;
onDataChannelCreated(dataChannel);
};
}
}
It is working fine for string or json i.e. dataChannel.send('Hello');
I have created a wiki page for same: wiki
Please help.

Please try something like this: (explanation at the end of the code)
var btnShareYourCamera = document.querySelector('#share-your-camera');
var localVideo = document.querySelector('#local-video');
var remoteVideo = document.querySelector('#remote-video');
var websocket = new WebSocket('wss://path-to-server:port/');
websocket.onmessage = function(event) {
var data = JSON.parse(event.data);
if (data.sdp) {
if (data.sdp.type === 'offer') {
getUserMedia(function(video_stream) {
localVideo.srcObject = video_stream;
answererPeer(new RTCSessionDescription(data.sdp), video_stream);
});
}
if (data.sdp.type === 'answer') {
offerer.setRemoteDescription(new RTCSessionDescription(data.sdp));
}
}
if (data.candidate) {
addIceCandidate((offerer || answerer), new RTCIceCandidate(data.candidate));
}
};
var iceTransportPolicy = 'all';
var iceTransportLimitation = 'udp';
function addIceCandidate(peer, candidate) {
if (iceTransportLimitation === 'tcp') {
if (candidate.candidate.toLowerCase().indexOf('tcp') === -1) {
return; // ignore UDP
}
}
peer.addIceCandidate(candidate);
}
var offerer, answerer;
var iceServers = {
iceServers: [{
'urls': [
'stun:stun.l.google.com:19302',
'stun:stun1.l.google.com:19302',
'stun:stun2.l.google.com:19302',
'stun:stun.l.google.com:19302?transport=udp',
]
}],
iceTransportPolicy: iceTransportPolicy,
rtcpMuxPolicy: 'require',
bundlePolicy: 'max-bundle'
};
// https://https;//cdn.webrtc-experiment.com/IceServersHandler.js
if (typeof IceServersHandler !== 'undefined') {
iceServers.iceServers = IceServersHandler.getIceServers();
}
var mediaConstraints = {
OfferToReceiveAudio: true,
OfferToReceiveVideo: true
};
/* offerer */
function offererPeer(video_stream) {
offerer = new RTCPeerConnection(iceServers);
offerer.idx = 1;
video_stream.getTracks().forEach(function(track) {
offerer.addTrack(track, video_stream);
});
offerer.ontrack = function(event) {
remoteVideo.srcObject = event.streams[0];
};
offerer.onicecandidate = function(event) {
if (!event || !event.candidate) return;
websocket.send(JSON.stringify({
candidate: event.candidate
}));
};
offerer.createOffer(mediaConstraints).then(function(offer) {
offerer.setLocalDescription(offer).then(function() {
websocket.send(JSON.stringify({
sdp: offer
}));
});
});
}
/* answerer */
function answererPeer(offer, video_stream) {
answerer = new RTCPeerConnection(iceServers);
answerer.idx = 2;
video_stream.getTracks().forEach(function(track) {
answerer.addTrack(track, video_stream);
});
answerer.ontrack = function(event) {
remoteVideo.srcObject = event.streams[0];
};
answerer.onicecandidate = function(event) {
if (!event || !event.candidate) return;
websocket.send(JSON.stringify({
candidate: event.candidate
}));
};
answerer.setRemoteDescription(offer).then(function() {
answerer.createAnswer(mediaConstraints).then(function(answer) {
answerer.setLocalDescription(answer).then(function() {
websocket.send(JSON.stringify({
sdp: answer
}));
});
});
});
}
var video_constraints = {
mandatory: {},
optional: []
};
function getUserMedia(successCallback) {
function errorCallback(e) {
alert(JSON.stringify(e, null, '\t'));
}
var mediaConstraints = {
video: true,
audio: true
};
navigator.mediaDevices.getUserMedia(mediaConstraints).then(successCallback).catch(errorCallback);
}
btnShareYourCamera.onclick = function() {
getUserMedia(function(video_stream) {
localVideo.srcObject = video_stream;
offererPeer(video_stream);
});
};
You must attach stream using peer.addTrack as you can see in the above example
You must receive remote stream using peer.ontrack as you can see in the above example
i.e. use addTrack to attach your camera and use ontrack to receive remote camera.
You must never send your stream using dataChannel.send. Both are totally different protocols. A MediaStream must be shared using RTP; not SCTP. RTP is used only if you call peer.addTrack method to attach your camera stream.
This process happens before you open or join a room.
See single-page demo here: https://www.webrtc-experiment.com/getStats/
HTML for above code snippet:
<button id="share-your-camera"></button>
<video id="local-video" controls autoplay playsinline></video>
<video id="remote-video" controls autoplay playsinline></video>

Not receiving video, onicecandidate is not executing

So I was following this tutorial to learn how to implement a WebRTC server-client setup. Once I got that working I wanted to split the client into two parts, one sender and one receiver. Now they can establish a connection with each other but the receiver never gets the stream from the sender.
I managed to determine that the code flow between the original code and split versions remains the same, except that neither peer executes the onicecandidate event.
According to this I need to explicitly include OfferToReceiveAudio: true and OfferToReceiveVideo: true since I'm using Chrome, which I did but it didn't seem to make any difference.
Currently, they both receive SDP from each other, there is a local and remote description in the peerConnection, and iceGatheringState is "new" but iceConnectionState is "checking" (unlike the second link where he states it should also be "new")
How come they aren't exchanging ICE candidates when it's split in two like this?
Sender.js
const HTTPSPort = 3434;
const domain = '127.0.0.1';
const wssHost = 'wss://' + domain + ':' + HTTPSPort + '/websocket/';
// Feed settings
const video = true;
const audio = true;
const constraints = { "audio": audio, "video": video };
var videoContainer = null, feed = null,
pC = null, wsc = new WebSocket(wssHost),
pCConfig = [
{ 'url': 'stun:stun.services.mozilla.com' },
{ 'url': 'stun:stun.l.google.com:19302' }
];
function pageReady() {
// Check browser WebRTC availability
navigator.mediaDevices.getUserMedia(constraints).then(function (stream) {
videoContainer = document.getElementById('videoFeed');
// Get the feed and show it in the local video element
feed = stream;
videoContainer.srcObject = feed;
}).catch(function () {
alert("Sorry, your browser does not support WebRTC!");
});
}
wsc.onmessage = function (evt) {
if (!pC) {
// Initiate peerConnection
pC = new RTCPeerConnection(pCConfig);
// Send any ice candidates to the other peer
pC.onicecandidate = onIceCandidateHandler;
pC.addStream(feed);
}
// Read the message
var signal = JSON.parse(evt.data);
if (signal.sdp) {
log('Received SDP from remote peer.');
pC.setRemoteDescription(new RTCSessionDescription(signal.sdp));
answerCall();
} else if (signal.candidate) {
log('Received ICECandidate from remote peer.');
pC.addIceCandidate(new RTCIceCandidate(signal.candidate));
}
};
function answerCall() {
pC.createAnswer().then(function (answer) {
var ans = new RTCSessionDescription(answer);
pC.setLocalDescription(ans).then(function () {
wsc.send(JSON.stringify({ 'sdp': ans }));
}).catch(errorHandler);
}).catch(errorHandler);
}
function onIceCandidateHandler(evt) {
if (!evt || !evt.candidate) return;
wsc.send(JSON.stringify({ 'candidate': evt.candidate }));
};
Receiver.js
const HTTPSPort = 3434;
const domain = '127.0.0.1';
const wssHost = 'wss://' + domain + ':' + HTTPSPort + '/websocket/';
var remoteVideo = null,
pC = null, wsc = new WebSocket(wssHost),
pCConfig = [
{ 'url': 'stun:stun.services.mozilla.com' },
{ 'url': 'stun:stun.l.google.com:19302' }
],
mediaConstraints = {
mandatory: {
OfferToReceiveAudio: true,
OfferToReceiveVideo: true
}
};
function pageReady() {
remoteVideo = document.getElementById('remoteVideo');
icebutton = document.getElementById('checkICE');
icebutton.addEventListener('click', function (evt) {
console.log(pC);
})
};
wsc.onopen = function () {
// Initiates peerConnection
pC = new RTCPeerConnection(pCConfig);
// Send any ICE candidates to the other peer
pC.onicecandidate = onIceCandidateHandler;
// Once remote stream arrives, show it in the remote video element
pC.onaddstream = onAddStreamHandler;
// Offer a connection to the server
createAndSendOffer();
};
function createAndSendOffer() {
pC.createOffer(mediaConstraints).then(function (offer) {
var off = new RTCSessionDescription(offer);
pC.setLocalDescription(off).then(function () {
wsc.send(JSON.stringify({ 'sdp': off }));
}).catch(errorHandler);
}).catch(errorHandler);
}
wsc.onmessage = function (evt) {
// Read the message
var signal = JSON.parse(evt.data);
if (signal.sdp) {
console.log('Received SDP from remote peer.');
pC.setRemoteDescription(new RTCSessionDescription(signal.sdp));
} else if (signal.candidate) {
console.log('Received ICECandidate from remote peer.');
pC.addIceCandidate(new RTCIceCandidate(signal.candidate));
}
};
function onIceCandidateHandler(evt) {
if (!evt || !evt.candidate) return;
wsc.send(JSON.stringify({ 'candidate': evt.candidate }));
};
function onAddStreamHandler(evt) {
// Set remote video stream as source for remote video HTML element
remoteVideo.srcObject = evt.stream;
};

You forgot iceServers. Change
pCConfig = [
{ 'url': 'stun:stun.services.mozilla.com' },
{ 'url': 'stun:stun.l.google.com:19302' }
];
to
pCConfig = {
iceServers: [
{ urls: 'stun:stun.l.google.com:19302' }
]
};
Additionally:
url has been deprecated, use urls.
The mozilla stun server has been decommissioned, so save yourself some time and exclude it.
mandatory and OfferToReceiveAudio have been deprecated. Use offerToReceiveAudio.
mandatory and OfferToReceiveVideo have been deprecated. Use offerToReceiveVideo.
Tips
Kudos for using promises (unlike the tutorial)! Note you can return them to flatten things:
function createAndSendOffer() {
return pC.createOffer(mediaConstraints).then(function (offer) {
return pC.setLocalDescription(offer);
})
.then(function () {
wsc.send(JSON.stringify({ sdp: pC.localDescription }));
})
.catch(errorHandler);
}