I read several other questions on a related issue, but none answered my question. I have an odd issue where I am able to use WebRTC to audio chat from chrome to firefox but not firefox to chrome.
Basically, when a user wishes to audio chat, he/she clicks a button #audioChatBtn
, which uses getUserMedia()
to setup a stream. The thing is, clicking #audioChatBtn
from Firefox doesn't fire the onaddstream
callback on Chrome, but clicking the button from Chrome fires onaddstream
on Firefox. Thus, I can audio chat from Chrome to Firefox but not the other way around. I have been trying to figure this out for several hours, but I'm hoping maybe someone here has an answer.
Relevant source:
var configuration = {
'iceServers': [
{ url: 'stun:stun.l.google.com:19302' },
{ url: 'stun:stun1.l.google.com:19302' },
{ url: 'stun:stun2.l.google.com:19302' },
{ url: 'stun:stun3.l.google.com:19302' },
{ url: 'stun:stun4.l.google.com:19302' }
]
};
var pc = RTCPeerConnection(configuration);
var myStream = null;
var currentAudioIndex = 0; // Number of created channels
var myAudioEnabled = false;
// send any ice candidates to the other peer
pc.onicecandidate = function (evt) {
if (evt.candidate)
$(document).trigger("persistState", { mode: 'rtc', 'candidate': evt.candidate });
};
// let the 'negotiationneeded' event trigger offer generation
pc.onnegotiationneeded = function () {
pc.createOffer(localDescCreated, logError);
}
// once remote stream arrives, play it in the audio element
pc.onaddstream = function (evt) {
console.log('creating and binding audio');
var idx = (currentAudioIndex++);
var audioElement = $('#audio' + idx);
if (audioElement.length == 0) {
var audio = $('<audio id="audio' + idx + '" autoplay>');
$('body').append(audio);
audioElement = $('#audio' + idx);
}
var audioObject = audioElement[0];
attachMediaStream(audioObject, evt.stream);
};
function localDescCreated(desc) {
pc.setLocalDescription(desc, function () {
$(document).trigger("persistState", { mode: 'rtc', 'sdp': pc.localDescription });
}, logError);
}
function logError(e) {
bootbox.alert("Audio chat could not be started.");
}
function hasGetUserMedia() {
return !!(navigator.getUserMedia || navigator.webkitGetUserMedia ||
navigator.mozGetUserMedia || navigator.msGetUserMedia);
}
server.onPersist = function(msg) {
if (msg.mode == "rtc") {
if (msg.sdp)
pc.setRemoteDescription(new RTCSessionDescription(msg.sdp), function () {
// if we received an offer, we need to answer
if (pc.remoteDescription.type == 'offer')
pc.createAnswer(localDescCreated, logError);
}, logError);
else
pc.addIceCandidate(new RTCIceCandidate(msg.candidate));
}
}
// On click, start audio chat from this user.
$('#audioChatBtn').click(function() {
if (!hasGetUserMedia()) {
bootbox.alert('Audio conferencing is not supported by your browser. (Currently only supported by Chrome, Firefox, and Opera web browsers.)');
return;
}
if (myAudioEnabled) {
myStream.stop();
displayAlert('Streaming closed', 'Audio chat is off');
$('#audioChatBtn').removeClass('btn-success').addClass('btn-primary');
} else {
getUserMedia({ video: false, audio: true }, function (localMediaStream) {
myStream = localMediaStream;
pc.addStream(localMediaStream);
displayAlert('Streaming...', 'Audio chat is enabled');
$('#audioChatBtn').removeClass('btn-primary').addClass('btn-success');
}, logError);
}
myAudioEnabled = !myAudioEnabled;
});
What I've tried
- Tried using
'optional': [{ 'DtlsSrtpKeyAgreement': 'true' }]
in the configuration after reading this question - Tried creating a new RTCPeerConnection() each request
- Tried using native browser functions instead of adapter.js.
- Explored Web Audio API instead of
getUserMedia()