Audio is not working but video does on WebRTC - javascript

I'm setting my video-conference service for academic purposes using WebRTC and everything works fine except for the audio capture. When I go to my website, it will only ask permission for the video camera, but not for the audio, and of course it just won't be any audio...
My main.js
'use strict';
var isChannelReady = false;
var isInitiator = false;
var isStarted = false;
var localStream;
var pc;
var remoteStream;
var turnReady;
var pcConfig = {
'iceServers': [{
'urls': 'stun:stun.l.google.com:19302'
}]
};
// Set up audio and video regardless of what devices are present.
var sdpConstraints = {
offerToReceiveAudio: true,
offerToReceiveVideo: true
};
/////////////////////////////////////////////
var room = 'foo';
// Could prompt for room name:
// room = prompt('Enter room name:');
var socket = io.connect();
if (room !== '') {
socket.emit('create or join', room);
console.log('Attempted to create or join room', room);
}
socket.on('created', function(room) {
console.log('Created room ' + room);
isInitiator = true;
});
socket.on('full', function(room) {
console.log('Room ' + room + ' is full');
});
socket.on('join', function (room){
console.log('Another peer made a request to join room ' + room);
console.log('This peer is the initiator of room ' + room + '!');
isChannelReady = true;
});
socket.on('joined', function(room) {
console.log('joined: ' + room);
isChannelReady = true;
});
socket.on('log', function(array) {
console.log.apply(console, array);
});
////////////////////////////////////////////////
function sendMessage(message) {
console.log('Client sending message: ', message);
socket.emit('message', message);
}
// This client receives a message
socket.on('message', function(message) {
console.log('Client received message:', message);
if (message === 'got user media') {
maybeStart();
} else if (message.type === 'offer') {
if (!isInitiator && !isStarted) {
maybeStart();
}
pc.setRemoteDescription(new RTCSessionDescription(message));
doAnswer();
} else if (message.type === 'answer' && isStarted) {
pc.setRemoteDescription(new RTCSessionDescription(message));
} else if (message.type === 'candidate' && isStarted) {
var candidate = new RTCIceCandidate({
sdpMLineIndex: message.label,
candidate: message.candidate
});
pc.addIceCandidate(candidate);
} else if (message === 'bye' && isStarted) {
handleRemoteHangup();
}
});
////////////////////////////////////////////////////
var localVideo = document.querySelector('#localVideo');
var remoteVideo = document.querySelector('#remoteVideo');
navigator.mediaDevices.getUserMedia({
audio: true,
video: true
})
.then(gotStream)
.catch(function(e) {
alert('getUserMedia() error: ' + e.name);
});
function gotStream(stream) {
console.log('Adding local stream.');
localVideo.src = window.URL.createObjectURL(stream);
localStream = stream;
sendMessage('got user media');
if (isInitiator) {
maybeStart();
}
}
var constraints = {
audio: true,
video: true
};
console.log('Getting user media with constraints', constraints);
if (location.hostname !== 'localhost') {
requestTurn(
'https://computeengineondemand.appspot.com/turn?username=41784574&key=4080218913'
);
}
function maybeStart() {
console.log('>>>>>>> maybeStart() ', isStarted, localStream, isChannelReady);
if (!isStarted && typeof localStream !== 'undefined' && isChannelReady) {
console.log('>>>>>> creating peer connection');
createPeerConnection();
pc.addStream(localStream);
isStarted = true;
console.log('isInitiator', isInitiator);
if (isInitiator) {
doCall();
}
}
}
window.onbeforeunload = function() {
sendMessage('bye');
};
/////////////////////////////////////////////////////////
function createPeerConnection() {
try {
pc = new RTCPeerConnection(null);
pc.onicecandidate = handleIceCandidate;
pc.onaddstream = handleRemoteStreamAdded;
pc.onremovestream = handleRemoteStreamRemoved;
console.log('Created RTCPeerConnnection');
} catch (e) {
console.log('Failed to create PeerConnection, exception: ' + e.message);
alert('Cannot create RTCPeerConnection object.');
return;
}
}
function handleIceCandidate(event) {
console.log('icecandidate 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.');
}
}
function handleRemoteStreamAdded(event) {
console.log('Remote stream added.');
remoteVideo.src = window.URL.createObjectURL(event.stream);
remoteStream = event.stream;
}
function handleCreateOfferError(event) {
console.log('createOffer() error: ', event);
}
function doCall() {
console.log('Sending offer to peer');
pc.createOffer(setLocalAndSendMessage, handleCreateOfferError);
}
function doAnswer() {
console.log('Sending answer to peer.');
pc.createAnswer().then(
setLocalAndSendMessage,
onCreateSessionDescriptionError
);
}
function setLocalAndSendMessage(sessionDescription) {
// Set Opus as the preferred codec in SDP if Opus is present.
// sessionDescription.sdp = preferOpus(sessionDescription.sdp);
pc.setLocalDescription(sessionDescription);
console.log('setLocalAndSendMessage sending message', sessionDescription);
sendMessage(sessionDescription);
}
function onCreateSessionDescriptionError(error) {
trace('Failed to create session description: ' + error.toString());
}
function requestTurn(turnURL) {
var turnExists = false;
for (var i in pcConfig.iceServers) {
if (pcConfig.iceServers[i].url.substr(0, 5) === 'turn:') {
turnExists = true;
turnReady = true;
break;
}
}
if (!turnExists) {
console.log('Getting TURN server from ', turnURL);
// No TURN server. Get one from computeengineondemand.appspot.com:
var xhr = new XMLHttpRequest();
xhr.onreadystatechange = function() {
if (xhr.readyState === 4 && xhr.status === 200) {
var turnServer = JSON.parse(xhr.responseText);
console.log('Got TURN server: ', turnServer);
pcConfig.iceServers.push({
'url': 'turn:' + turnServer.username + '#' + turnServer.turn,
'credential': turnServer.password
});
turnReady = true;
}
};
xhr.open('GET', turnURL, true);
xhr.send();
}
}
function handleRemoteStreamAdded(event) {
console.log('Remote stream added.');
remoteVideo.src = window.URL.createObjectURL(event.stream);
remoteStream = event.stream;
}
function handleRemoteStreamRemoved(event) {
console.log('Remote stream removed. Event: ', event);
}
function hangup() {
console.log('Hanging up.');
stop();
sendMessage('bye');
}
function handleRemoteHangup() {
console.log('Session terminated.');
stop();
isInitiator = false;
}
function stop() {
isStarted = false;
// isAudioMuted = false;
// isVideoMuted = false;
pc.close();
pc = null;
}
///////////////////////////////////////////
// Set Opus as the default audio codec if it's present.
function preferOpus(sdp) {
var sdpLines = sdp.split('\r\n');
var mLineIndex;
// Search for m line.
for (var i = 0; i < sdpLines.length; i++) {
if (sdpLines[i].search('m=audio') !== -1) {
mLineIndex = i;
break;
}
}
if (mLineIndex === null) {
return sdp;
}
// If Opus is available, set it as the default in m line.
for (i = 0; i < sdpLines.length; i++) {
if (sdpLines[i].search('opus/48000') !== -1) {
var opusPayload = extractSdp(sdpLines[i], /:(\d+) opus\/48000/i);
if (opusPayload) {
sdpLines[mLineIndex] = setDefaultCodec(sdpLines[mLineIndex],
opusPayload);
}
break;
}
}
// Remove CN in m line and sdp.
sdpLines = removeCN(sdpLines, mLineIndex);
sdp = sdpLines.join('\r\n');
return sdp;
}
function extractSdp(sdpLine, pattern) {
var result = sdpLine.match(pattern);
return result && result.length === 2 ? result[1] : null;
}
// Set the selected codec to the first in m line.
function setDefaultCodec(mLine, payload) {
var elements = mLine.split(' ');
var newLine = [];
var index = 0;
for (var i = 0; i < elements.length; i++) {
if (index === 3) { // Format of media starts from the fourth.
newLine[index++] = payload; // Put target payload to the first.
}
if (elements[i] !== payload) {
newLine[index++] = elements[i];
}
}
return newLine.join(' ');
}
// Strip CN from sdp before CN constraints is ready.
function removeCN(sdpLines, mLineIndex) {
var mLineElements = sdpLines[mLineIndex].split(' ');
// Scan from end for the convenience of removing an item.
for (var i = sdpLines.length - 1; i >= 0; i--) {
var payload = extractSdp(sdpLines[i], /a=rtpmap:(\d+) CN\/\d+/i);
if (payload) {
var cnPos = mLineElements.indexOf(payload);
if (cnPos !== -1) {
// Remove CN payload from m line.
mLineElements.splice(cnPos, 1);
}
// Remove CN line in sdp
sdpLines.splice(i, 1);
}
}
sdpLines[mLineIndex] = mLineElements.join(' ');
return sdpLines;
}
and my index.js (server)
var fs = require('fs');
var http = require('http');
var https = require('https');
var cors = require('cors');
var privateKey = fs.readFileSync('host.key', 'utf8');
var certificate = fs.readFileSync('host.cert', 'utf8');
var os = require('os');
var credentials = {key: privateKey, cert: certificate};
var express = require('express');
var app = express();
app.use(cors());
var path = require('path');
app.use(express.static(path.join(__dirname, 'public')));
app.get('/', function(req, res){
res.sendFile(__dirname + '/index.html');
//console.log('Servidor en marxa.');
});
console.log('Servidor ON');
var httpServer = http.createServer(app);
var httpsServer = https.createServer(credentials,app);
httpServer.listen(80,"0.0.0.0");
httpsServer.listen(443,"0.0.0.0");
var io = require("socket.io");
io = io.listen(httpServer);
io = io.listen(httpsServer);
//EL SERVER ESTA CONFIGURAT I FUNCIONA//
var usersConnected = [];
var nUsersConnected = 0;
io.sockets.on('connection',function(socket){
function log() {
var array = ['Message from server:'];
array.push.apply(array, arguments);
socket.emit('log', array);
}
usersConnected.push(socket.id);
nUsersConnected++;
log('Welcome to WebRTC Server: Miguel & Pavel');
console.log('New user connected: '+socket.id+' #Users connected = '+nUsersConnected);
socket.on('message', function(message) {
log('Client said: ', message);
// for a real app, would be room-only (not broadcast)
socket.broadcast.emit('message', message);
});
socket.on('create or join', function(room) {
log('Received request to create or join room ' + room);
var numClients = io.sockets.sockets.length;
log('Room ' + room + ' now has ' + numClients + ' client(s)');
if (numClients === 1) {
socket.join(room);
log('Client ID ' + socket.id + ' created room ' + room);
socket.emit('created', room, socket.id);
} else if (numClients === 2) {
log('Client ID ' + socket.id + ' joined room ' + room);
io.sockets.in(room).emit('join', room);
socket.join(room);
socket.emit('joined', room, socket.id);
io.sockets.in(room).emit('ready');
} else { // max two clients
socket.emit('full', room);
}
});
socket.on('ipaddr', function() {
var ifaces = os.networkInterfaces();
for (var dev in ifaces) {
ifaces[dev].forEach(function(details) {
if (details.family === 'IPv4' && details.address !== '127.0.0.1') {
socket.emit('ipaddr', details.address);
}
});
}
});
socket.on('disconnect', function (message) {
nUsersConnected--;
console.log('User: '+socket.id+' disconnected #Users connected = '+nUsersConnected);
/*if(usuari1 == socket.id){
io.to(usuari2).emit('message', {type:'hangup'});
}else{
io.to(usuari1).emit('message', {type:'hangup'});
}*/
//tots els usuaris s'han desconnectat
if(nUsersConnected == 0){
console.log('All users have been deleted');
nUsersConnected = 0;
usersConnected = [];
}
})
socket.on('bye', function(){
console.log('received bye');
});
});
Thanks!

When I go to my website, it will only ask permission for the video camera, but not for the audio, and of course it just won't be any audio
Am suspecting issue with your system microphone.
If your microphone is proper, then your localstream should have audioTracks localStream.getAudioTracks(). And if we unmute localVideo element, it should echo the audio.
Try this demo https://webrtc.github.io/samples/src/content/devices/input-output/ to test your microphone. Update your question with the results.
Google stopped providing turn servers with below url, so you need to setup your TurnServer. Try CoTURN.
requestTurn(
'https://computeengineondemand.appspot.com/turn?username=41784574&key=4080218913'
);

Related

Javascript WebRTC Failed to set remote answer sdp: Called in wrong state: kHaveRemoteOffer and Called in wrong state: kStable

I can't get my WebRTC code to work properly.. I did everything right I believe and it's still not working. There is something strange why ontrack gets called so early maybe it's suppose to be like that.
The website uses javascript code, the server code I didn't post but thats where WebSockets connect is just a exchanger, what you send to server it sends the same information back to the other partner (stranger) you are connected too.
Server code looks like this little sample
private void writeStranger(UserProfile you, String msg) {
UserProfile stranger = you.stranger;
if(stranger != null)
sendMessage(stranger.getWebSocket(), msg);
}
public void sendMessage(WebSocket websocket, String msg) {
try {
websocket.send(msg);
} catch ( WebsocketNotConnectedException e ) {
disconnnectClient(websocket);
}
}
//...
case "ice_candidate":
JSONObject candidatePackage = (JSONObject) packet.get(1);
JSONObject candidate = (JSONObject) candidatePackage.get("candidate");
obj = new JSONObject();
list = new JSONArray();
list.put("iceCandidate");
obj.put("candidate", candidate);
list.put(obj);
System.out.println("Sent = " + list.toString());
writeStranger(you, list.toString()); //send ice candidate to stranger
break;
case "send_answer":
JSONObject sendAnswerPackage = (JSONObject) packet.get(1);
JSONObject answer = (JSONObject) sendAnswerPackage.get("answer");
obj = new JSONObject();
list = new JSONArray();
list.put("getAnswer");
obj.put("answer", answer);
list.put(obj);
System.out.println("Sent = " + list.toString());
writeStranger(you, list.toString()); //send answer to stranger
break;
case "send_offer":
JSONObject offerPackage = (JSONObject) packet.get(1);
JSONObject offer = (JSONObject) offerPackage.get("offer");
obj = new JSONObject();
list = new JSONArray();
list.put("getOffer");
obj.put("offer", offer);
list.put(obj);
System.out.println("Sent = " + list.toString());
writeStranger(you, list.toString()); //send ice candidate to stranger
break;
Here are my outputs.
RAW Text: https://pastebin.com/raw/FL8g29gG
JSON colored: https://pastebin.com/FL8g29gG
My javascript Code below
var ws;
var peerConnection, localStream;
var rtc_server = {
iceServers: [
{urls: "stun:stun.l.google.com:19302"},
{urls: "stun:stun.services.mozilla.com"},
{urls: "stun:stun.stunprotocol.org:3478"},
{url: "stun:stun.l.google.com:19302"},
{url: "stun:stun.services.mozilla.com"},
{url: "stun:stun.stunprotocol.org:3478"},
]
}
//offer SDP's tells other peers what you would like
var rtc_media_constraints = {
mandatory: {
OfferToReceiveAudio: true,
OfferToReceiveVideo: true
}
};
var rtc_peer_options = {
optional: [
{DtlsSrtpKeyAgreement: true}, //To make Chrome and Firefox to interoperate.
]
}
var PeerConnection = RTCPeerConnection || window.PeerConnection || window.webkitPeerConnection || window.webkitRTCPeerConnection || window.mozRTCPeerConnection;
var IceCandidate = RTCIceCandidate || window.mozRTCIceCandidate || window.RTCIceCandidate;
var SessionDescription = RTCSessionDescription || window.mozRTCSessionDescription || window.RTCSessionDescription;
var getUserMedia = navigator.getUserMedia || navigator.webkitGetUserMedia || navigator.mozGetUserMedia || navigator.msGetUserMedia;
function hasSupportForVideoChat() {
return window.RTCPeerConnection && window.RTCIceCandidate && window.RTCSessionDescription && navigator.mediaDevices && navigator.mediaDevices.getUserMedia && (RTCPeerConnection.prototype.addStream || RTCPeerConnection.prototype.addTrack) ? true : false;
}
function loadMyCameraStream() {
if (getUserMedia) {
getUserMedia.call(navigator, { video: {facingMode: "user", aspectRatio: 4 / 3/*height: 272, width: 322*/}, audio: { echoCancellation : true } },
function(localMediaStream) {
//Add my video
$("div#videoBox video#you")[0].muted = true;
$("div#videoBox video#you")[0].autoplay = true;
$("div#videoBox video#you").attr('playsinline', '');
$("div#videoBox video#you").attr('webkit-playsinline', '');
$("div#videoBox video#you")[0].srcObject = localMediaStream;
localStream = localMediaStream;
},
function(e) {
addStatusMsg("Your Video has error : " + e);
}
);
} else {
addStatusMsg("Your browser does not support WebRTC (Camera/Voice chat).");
return;
}
}
function loadStrangerCameraStream() {
if(!hasSupportForVideoChat())
return;
peerConnection = new PeerConnection(rtc_server, rtc_peer_options);
if (peerConnection.addTrack !== undefined)
localStream.getTracks().forEach(track => peerConnection.addTrack(track, localStream));
else
peerConnection.addStream(localStream);
peerConnection.onicecandidate = function(e) {
if (!e || !e.candidate)
return;
ws.send(JSON.stringify(['ice_candidate', {"candidate": e.candidate}]));
};
if (peerConnection.addTrack !== undefined) {
//newer technology
peerConnection.ontrack = function(e) {
//e.streams.forEach(stream => doAddStream(stream));
addStatusMsg("ontrack called");
//Add stranger video
$("div#videoBox video#stranger").attr('playsinline', '');
$("div#videoBox video#stranger").attr('webkit-playsinline', '');
$('div#videoBox video#stranger')[0].srcObject = e.streams[0];
$("div#videoBox video#stranger")[0].autoplay = true;
};
} else {
//older technology
peerConnection.onaddstream = function(e) {
addStatusMsg("onaddstream called");
//Add stranger video
$("div#videoBox video#stranger").attr('playsinline', '');
$("div#videoBox video#stranger").attr('webkit-playsinline', '');
$('div#videoBox video#stranger')[0].srcObject = e.stream;
$("div#videoBox video#stranger")[0].autoplay = true;
};
}
peerConnection.createOffer(
function(offer) {
peerConnection.setLocalDescription(offer, function () {
//both offer and peerConnection.localDescription are the same.
addStatusMsg('createOffer, localDescription: ' + JSON.stringify(peerConnection.localDescription));
//addStatusMsg('createOffer, offer: ' + JSON.stringify(offer));
ws.send(JSON.stringify(['send_offer', {"offer": peerConnection.localDescription}]));
},
function(e) {
addStatusMsg('createOffer, set description error' + e);
});
},
function(e) {
addStatusMsg("createOffer error: " + e);
},
rtc_media_constraints
);
}
function closeStrangerCameraStream() {
$('div#videoBox video#stranger')[0].srcObject = null
if(peerConnection)
peerConnection.close();
}
function iceCandidate(candidate) {
//ICE = Interactive Connectivity Establishment
if(peerConnection)
peerConnection.addIceCandidate(new IceCandidate(candidate));
else
addStatusMsg("peerConnection not created error");
addStatusMsg("Peer Ice Candidate = " + JSON.stringify(candidate));
}
function getAnswer(answer) {
if(!hasSupportForVideoChat())
return;
if(peerConnection) {
peerConnection.setRemoteDescription(new SessionDescription(answer), function() {
console.log("get answer ok");
addStatusMsg("peerConnection, SessionDescription answer is ok");
},
function(e) {
addStatusMsg("peerConnection, SessionDescription fail error: " + e);
});
}
}
function getOffer(offer) {
if(!hasSupportForVideoChat())
return;
addStatusMsg("peerConnection, setRemoteDescription offer: " + JSON.stringify(offer));
if(peerConnection) {
peerConnection.setRemoteDescription(new SessionDescription(offer), function() {
peerConnection.createAnswer(
function(answer) {
peerConnection.setLocalDescription(answer);
addStatusMsg("create answer sent: " + JSON.stringify(answer));
ws.send(JSON.stringify(['send_answer', {"answer": answer}]));
},
function(e) {
addStatusMsg("peerConnection, setRemoteDescription create answer fail: " + e);
}
);
});
}
}
My website where I use it: https://www.camspark.com/
Fixed myself I figured out I had 2 problems with this code.
First problem was then createOffer() must only be sent by 1 person not both people.. you have to randomly pick which person which does the createOffer().
Second problem is the ICE Candidate's you have to create a queue/array for both sides, which holds all the incoming ice_candidates. Only do the peerConnection.addIceCandidate(new IceCandidate(candidate)); when the response to createOffer() is received and the setRemoteDescription from createOffer() response is set up.
Both getAnswer() and getOffer() use exactly same code, but one is received for 1 client while the other is received for the other client. Both need to flush the IceCandidates array when either of them is triggered.. Maybe if anyone wants you could combine both functions into 1 function as the code is the same.
Final working code looks like this
var ws;
var peerConnection, localStream;
//STUN = (Session Traversal Utilities for NAT)
var rtc_server = {
iceServers: [
{urls: "stun:stun.l.google.com:19302"},
{urls: "stun:stun.services.mozilla.com"},
{urls: "stun:stun.stunprotocol.org:3478"},
{url: "stun:stun.l.google.com:19302"},
{url: "stun:stun.services.mozilla.com"},
{url: "stun:stun.stunprotocol.org:3478"},
]
}
//offer SDP = [Session Description Protocol] tells other peers what you would like
var rtc_media_constraints = {
mandatory: {
OfferToReceiveAudio: true,
OfferToReceiveVideo: true
}
};
var rtc_peer_options = {
optional: [
{DtlsSrtpKeyAgreement: true}, //To make Chrome and Firefox to interoperate.
]
}
var finishSDPVideoOffer = false;
var isOfferer = false;
var iceCandidates = [];
var PeerConnection = RTCPeerConnection || window.PeerConnection || window.webkitPeerConnection || window.webkitRTCPeerConnection || window.mozRTCPeerConnection;
var IceCandidate = RTCIceCandidate || window.mozRTCIceCandidate || window.RTCIceCandidate;
var SessionDescription = RTCSessionDescription || window.mozRTCSessionDescription || window.RTCSessionDescription;
var getUserMedia = navigator.getUserMedia || navigator.webkitGetUserMedia || navigator.mozGetUserMedia || navigator.msGetUserMedia;
function hasSupportForVideoChat() {
return window.RTCPeerConnection && window.RTCIceCandidate && window.RTCSessionDescription && navigator.mediaDevices && navigator.mediaDevices.getUserMedia && (RTCPeerConnection.prototype.addStream || RTCPeerConnection.prototype.addTrack) ? true : false;
}
function loadMyCameraStream() {
if (getUserMedia) {
getUserMedia.call(navigator, { video: {facingMode: "user", aspectRatio: 4 / 3/*height: 272, width: 322*/}, audio: { echoCancellation : true } },
function(localMediaStream) {
//Add my video
$("div#videoBox video#you")[0].muted = true;
$("div#videoBox video#you")[0].autoplay = true;
$("div#videoBox video#you").attr('playsinline', '');
$("div#videoBox video#you").attr('webkit-playsinline', '');
$("div#videoBox video#you")[0].srcObject = localMediaStream;
localStream = localMediaStream;
},
function(e) {
addStatusMsg("Your Video has error : " + e);
}
);
} else {
addStatusMsg("Your browser does not support WebRTC (Camera/Voice chat).");
return;
}
}
function loadStrangerCameraStream(isOfferer_) {
if(!hasSupportForVideoChat())
return;
//Only add pending ICE Candidates when getOffer() is finished.
finishSDPVideoOfferOrAnswer = false;
iceCandidates = []; //clear ICE Candidates array.
isOfferer = isOfferer_;
peerConnection = new PeerConnection(rtc_server, rtc_peer_options);
if (peerConnection.addTrack !== undefined)
localStream.getTracks().forEach(track => peerConnection.addTrack(track, localStream));
else
peerConnection.addStream(localStream);
peerConnection.onicecandidate = function(e) {
if (!e || !e.candidate)
return;
ws.send(JSON.stringify(['ice_candidate', {"candidate": e.candidate}]));
};
if (peerConnection.addTrack !== undefined) {
//newer technology
peerConnection.ontrack = function(e) {
//e.streams.forEach(stream => doAddStream(stream));
addStatusMsg("ontrack called");
//Add stranger video
$("div#videoBox video#stranger").attr('playsinline', '');
$("div#videoBox video#stranger").attr('webkit-playsinline', '');
$('div#videoBox video#stranger')[0].srcObject = e.streams[0];
$("div#videoBox video#stranger")[0].autoplay = true;
};
} else {
//older technology
peerConnection.onaddstream = function(e) {
addStatusMsg("onaddstream called");
//Add stranger video
$("div#videoBox video#stranger").attr('playsinline', '');
$("div#videoBox video#stranger").attr('webkit-playsinline', '');
$('div#videoBox video#stranger')[0].srcObject = e.stream;
$("div#videoBox video#stranger")[0].autoplay = true;
};
}
if(isOfferer) {
peerConnection.createOffer(
function(offer) {
peerConnection.setLocalDescription(offer, function () {
//both offer and peerConnection.localDescription are the same.
addStatusMsg('createOffer, localDescription: ' + JSON.stringify(peerConnection.localDescription));
//addStatusMsg('createOffer, offer: ' + JSON.stringify(offer));
ws.send(JSON.stringify(['send_offer', {"offer": peerConnection.localDescription}]));
},
function(e) {
addStatusMsg('createOffer, set description error' + e);
});
},
function(e) {
addStatusMsg("createOffer error: " + e);
},
rtc_media_constraints
);
}
}
function closeStrangerCameraStream() {
$('div#videoBox video#stranger')[0].srcObject = null
if(peerConnection)
peerConnection.close();
}
function iceCandidate(candidate) {
//ICE = Interactive Connectivity Establishment
if(!finishSDPVideoOfferOrAnswer) {
iceCandidates.push(candidate);
addStatusMsg("Queued iceCandidate");
return;
}
if(!peerConnection) {
addStatusMsg("iceCandidate peerConnection not created error.");
return;
}
peerConnection.addIceCandidate(new IceCandidate(candidate));
addStatusMsg("Added on time, Peer Ice Candidate = " + JSON.stringify(candidate));
}
function getAnswer(answer) {
if(!hasSupportForVideoChat())
return;
if(!peerConnection) {
addStatusMsg("getAnswer peerConnection not created error.");
return;
}
peerConnection.setRemoteDescription(new SessionDescription(answer), function() {
addStatusMsg("getAnswer SessionDescription answer is ok");
finishSDPVideoOfferOrAnswer = true;
while (iceCandidates.length) {
var candidate = iceCandidates.shift();
try {
peerConnection.addIceCandidate(new IceCandidate(candidate));
addStatusMsg("Adding queued ICE Candidates");
} catch(e) {
addStatusMsg("Error adding queued ICE Candidates error:" + e);
}
}
iceCandidates = [];
},
function(e) {
addStatusMsg("getAnswer SessionDescription fail error: " + e);
});
}
function getOffer(offer) {
if(!hasSupportForVideoChat())
return;
if(!peerConnection) {
addStatusMsg("getOffer peerConnection not created error.");
return;
}
addStatusMsg("getOffer setRemoteDescription offer: " + JSON.stringify(offer));
peerConnection.setRemoteDescription(new SessionDescription(offer), function() {
finishSDPVideoOfferOrAnswer = true;
while (iceCandidates.length) {
var candidate = iceCandidates.shift();
try {
peerConnection.addIceCandidate(new IceCandidate(candidate));
addStatusMsg("Adding queued ICE Candidates");
} catch(e) {
addStatusMsg("Error adding queued ICE Candidates error:" + e);
}
}
iceCandidates = [];
if(!isOfferer) {
peerConnection.createAnswer(
function(answer) {
peerConnection.setLocalDescription(answer);
addStatusMsg("getOffer create answer sent: " + JSON.stringify(answer));
ws.send(JSON.stringify(['send_answer', {"answer": answer}]));
},
function(e) {
addStatusMsg("getOffer setRemoteDescription create answer fail: " + e);
}
);
}
});
}
Here is the patch I did on server-side WebSocket (Java) server.
//JSON
//["connected", {videoChatOfferer: true}]
//["connected", {videoChatOfferer: false}]
JSONObject obj = new JSONObject();
JSONArray list = new JSONArray();
list.put("loadStrangerCameraStream");
obj.put("videoChatOfferer", true); //first guy offerer for WebRTC.
list.put(obj);
server.sendMessage(websocket, list.toString()); //connected to chat partner
obj.put("videoChatOfferer", false); //second guy isn't offerer.
list.put(obj);
server.sendMessage(stranger.getWebSocket(), list.toString()); //connected to chat partner

Failed to set session description: OperationError: Failed to set remote answer sdp: Called in wrong state: STATE_INPROGRESS

I used Angular JS : - Getting Room Id/Token from server side to connect web socket
Following code used in application -
app.controller("videoCallingController",
["$scope", "$location", "$rootScope", "$localStorage", 'AuthenticationService', "CommonService", "videoService",
function ($scope, $location, $rootScope, $localStorage, AuthenticationService, CommonService, videoService) {
$scope.UserInfo = getLoggedInUserDetails();
if ($scope.UserInfo == null) {
window.location.replace("/login.html#/");
}
$scope.offerConversationDetails = JSON.parse(localStorage.getItem('offerConversationDetails')); //$rootScope.offerConversationDetails;
$scope.IsCallBtn = false;
$scope.IsCallEnd = false;
$scope.IsRemoteStreaming = false;
//------- start custom Web socket ----------------
var ws;
function initWS(groupname) {
// var groupname = "TestRoom";
ws = new WebSocket("wss://" + location.host + "/api/" + "api/Values/Get?id=" + groupname);
ws.onopen = function () { };
ws.onmessage = function (evt) {
var signal = null;
if (!pc1) { answerCall(); }
signal = JSON.parse(evt.data);
if (signal.sdp) {
console.log("Received SDP from remote peer.");
pc1.setRemoteDescription(new RTCSessionDescription(signal.sdp));
createAndSendAnswer(signal.sdp);
}
else if (signal.candidate) {
console.log("Received ICECandidate from remote peer.");
//pc1.addIceCandidate(new RTCIceCandidate(signal.candidate));
addIceCandidate(signal);
} else if (signal.closeConnection) {
console.log("Received 'close call' signal from remote peer.");
//endCall();
}
};
ws.onerror = function (evt) {
console.log(evt.message);
};
ws.onclose = function () {
console.log("disconnected");
};
}
function sendWS(msg) {
if (ws.readyState == WebSocket.OPEN) {
ws.send(msg);
}
}
function closeWS() {
ws.close();
}
//------- end custom Web socket ----------------
//---- start Html element selectors ------------
//var callButton = document.getElementById('callButton');
//callButton.onclick = initiateCall;
var startTime;
var localVideo = document.getElementById('localVideo');
var remoteVideo = document.getElementById('remoteVideo');
localVideo.addEventListener('loadedmetadata', function () {
trace('Local video videoWidth: ' + this.videoWidth +
'px, videoHeight: ' + this.videoHeight + 'px');
});
remoteVideo.addEventListener('loadedmetadata', function () {
trace('Remote video videoWidth: ' + this.videoWidth +
'px, videoHeight: ' + this.videoHeight + 'px');
});
remoteVideo.onresize = function () {
trace('Remote video size changed to ' +
remoteVideo.videoWidth + 'x' + remoteVideo.videoHeight);
// We'll use the first onsize callback as an indication that video has started
// playing out.
if (startTime) {
var elapsedTime = window.performance.now() - startTime;
trace('Setup time: ' + elapsedTime.toFixed(3) + 'ms');
startTime = null;
}
};
//---- end Html element selectors ------------
var localStream;
var pc1;
// var pc2;
var offerOptions = {
offerToReceiveAudio: 1,
offerToReceiveVideo: 1
};
function getName(pc) {
return 'pc1'; //(pc === pc1) ? 'pc1' : 'pc2';
}
function getOtherPc(pc) {
return pc1; //(pc === pc1) ? pc2 : pc1;
}
function gotStream(stream) {
trace('Received local stream');
//localVideo = attachMediaStream(localVideo, stream);
// attachMediaStream(localVideo, stream);
console.log(attachMediaStream(localVideo, stream));
localStream = stream;
// callButton.disabled = false;
muteAudio(stream);
}
function muteAudio(stream) {
var AudioTrack = stream.getAudioTracks()[0];
//var localAudioBtn = document.getElementById('localAudioBtn');
var muteBtn = document.getElementById('localAudioMuteBtn');
var unMuteBtn = document.getElementById('localAudioUnMuteBtn');
//localAudioBtn.style.visibility = 'visible';
unMuteBtn.onclick = MuteUnMute;
muteBtn.onclick = MuteUnMute;
function MuteUnMute() {
AudioTrack.enabled = !AudioTrack.enabled;
muteBtn.className = 'btn btn-info btn-lg' + (AudioTrack.enabled ? '' : ' hidden');
unMuteBtn.className = 'btn btn-warning btn-lg' + (AudioTrack.enabled ? ' hidden' : '');
//localAudioBtn.className = 'button button-mute' + (AudioTrack.enabled ? '' : ' muted');
};
}
function gumFailed(e) {
alert('getUserMedia() error: ' + e.name);
}
function start() {
trace('Requesting local stream');
// startButton.disabled = true;
var constraints = {
audio: true,
video: { width: 1280, height: 720 } //true
};
if (typeof Promise === 'undefined') {
navigator.getUserMedia(constraints, gotStream, gumFailed);
} else {
navigator.mediaDevices.getUserMedia(constraints)
.then(gotStream)
.catch(gumFailed);
}
}
var Room = {};
Room.createNewRoom = function () {
if ($scope.offerConversationDetails.type == 'owner') {
videoService.CreateNewRoom($scope.offerConversationDetails.offerId, function (resp) {
if (resp.status === iresponseStatus.success) {
console.log("CreateNewRoom");
console.log(resp.result);
$scope.roomDetails = resp.result;
$rootScope.offerRoomId = $scope.roomDetails.RoomId;
Room.getRoomStatusById($scope.roomDetails.RoomId);
}
else if (resp.status === iresponseStatus.error) {
if (resp.error.code == 5) {
callDateTimeExpired(resp.error.message);
}
displayNoty("error", resp.error.message);
console.log(resp.error);
setTimeout(function () { Room.createNewRoom(); }, 10000);
}
});
}
}
Room.getRoomToken = function () {
if ($scope.offerConversationDetails.type == 'winner') {
videoService.GetRoomToken($scope.offerConversationDetails.offerId, function (resp) {
if (resp.status === iresponseStatus.success) {
console.log("Get Room Token");
console.log(resp.result);
$scope.roomDetails = resp.result;
$rootScope.offerRoomId = $scope.roomDetails.RoomId;
initWS($scope.roomDetails.RoomToken);
}
else if (resp.status === iresponseStatus.error) {
console.log(resp.error);
if (resp.error.code == 5) {
callDateTimeExpired(resp.error.message);
}
setTimeout(function () { Room.getRoomToken(); }, 1000);
}
});
}
}
Room.getRoomStatusById = function (roomId) {
if ($scope.offerConversationDetails.type == 'owner') {
videoService.GetRoomStatusById(roomId, function (resp) {
if (resp.status === iresponseStatus.success) {
if (resp.result == 2) //Active
{
initWS($scope.roomDetails.RoomToken);
$scope.IsCallBtn = true;
}
else {
//setTimeout(Room.getRoomStatusById(roomId), 100);
setTimeout(function () { Room.getRoomStatusById(roomId); }, 100);
}
}
else if (resp.status === iresponseStatus.error) {
console.log(resp.error);
if (resp.error.code == 5) {
callDateTimeExpired(resp.error.message);
}
//setTimeout(Room.getRoomStatusById(roomId), 100);
setTimeout(function () { Room.getRoomStatusById(roomId); }, 100);
}
});
}
}
Room.updateRoomConsumption = function () {
if ($scope.offerConversationDetails.type == 'owner') {
updateRoomConsumption_interval = setInterval(UpdateRoomConsumption(), 5000);
}
}
var updateRoomConsumption_interval = null;
function UpdateRoomConsumption() {
if ($scope.offerConversationDetails.ChatDurationSeconds > $scope.seconds) {
videoService.UpdateRoomConsumption($scope.roomDetails.RoomId, $scope.seconds, function (resp) {
if (resp.status === iresponseStatus.success) {
// $scope.seconds += 5;
//setTimeout(function () { Room.updateRoomConsumption(); }, 5000);
}
else if (resp.status === iresponseStatus.error) {
console.log(resp.error);
}
});
}
else {
EndCall();
clearInterval(updateRoomConsumption_interval);
}
$scope.seconds = $scope.seconds + 5;
}
//initWS();
start();
Room.createNewRoom();
Room.getRoomToken();
function prepareCall() {
// callButton.disabled = true;
trace('Starting call');
startTime = 4410; //window.performance.now();
var videoTracks = localStream.getVideoTracks();
var audioTracks = localStream.getAudioTracks();
if (videoTracks.length > 0) {
trace('Using video device: ' + videoTracks[0].label);
}
if (audioTracks.length > 0) {
trace('Using audio device: ' + audioTracks[0].label);
}
// var servers = null;
var servers = {
iceServers: [
//{ url: "stun:23.21.150.121" },
{ url: "stun:stun.1.google.com:19302" }
//{ url: !isFirefox ? 'stun:stun.l.google.com:19302' : 'stun:23.21.150.121' }
]
};
pc1 = new RTCPeerConnection(servers);
trace('Created local peer connection object pc1');
pc1.onicecandidate = function (e) {
onIceCandidate(pc1, e);
};
// pc2 = new RTCPeerConnection(servers);
//trace('Created remote peer connection object pc2');
//pc2.onicecandidate = function (e) {
// onIceCandidate(pc2, e);
//};
pc1.oniceconnectionstatechange = function (e) {
onIceStateChange(pc1, e);
};
//pc2.oniceconnectionstatechange = function (e) {
// onIceStateChange(pc2, e);
//};
pc1.onaddstream = gotRemoteStream;
};
// run start(true) to initiate a call
$scope.initiateCall = function () {
prepareCall();
pc1.addStream(localStream);
trace('Added local stream to pc1');
trace('pc1 createOffer start');
pc1.createOffer(onCreateOfferSuccess, onCreateSessionDescriptionError,
offerOptions);
};
function answerCall() {
prepareCall();
pc1.addStream(localStream);
trace('Added local stream to pc1');
//createAndSendAnswer();
};
function onCreateSessionDescriptionError(error) {
trace('Failed to create session description: ' + error.toString());
}
function onCreateOfferSuccess(desc) {
trace('Offer from pc1\n' + desc.sdp);
trace('pc1 setLocalDescription start');
pc1.setLocalDescription(desc, function () {
onSetLocalSuccess(pc1);
sendWS(JSON.stringify({ "sdp": desc }));
}, onSetSessionDescriptionError);
}
function createAndSendAnswer(desc) {
// trace('pc2 setRemoteDescription start');
pc1.setRemoteDescription(desc, function () {
onSetRemoteSuccess(pc1);
}, onSetSessionDescriptionError);
trace('pc2 createAnswer start');
//Since the 'remote' side has no media stream we need
// to pass in the right constraints in order for it to
// accept the incoming offer of audio and video.
pc1.createAnswer(onCreateAnswerSuccess, onCreateSessionDescriptionError);
};
function onSetLocalSuccess(pc) {
trace(getName(pc) + ' setLocalDescription complete');
}
function onSetRemoteSuccess(pc) {
trace(getName(pc) + ' setRemoteDescription complete');
}
function onSetSessionDescriptionError(error) {
trace('Failed to set session description: ' + error.toString());
}
function gotRemoteStream(event) {
// remoteVideo = attachMediaStream(remoteVideo, e.stream);
if (event != null && event != undefined && event.stream != null && event.stream != undefined) {
$scope.offerConversationDetails.ChatDurationSeconds = $scope.offerConversationDetails.ChatDuration * 60;
$scope.IsDisplayTimer = true; $scope.IsRemoteStreaming = true; $scope.IsCallBtn = false;
$scope.$apply();
initializeClock('clockdiv', $scope.offerConversationDetails.ChatDurationSeconds);
$scope.seconds = 0;
Room.updateRoomConsumption($scope.seconds);
//event.stream.oninactive = function ()
//{
// if ($scope.IsCallEnd == false)
// hangup();
// alert("remote streaming get stop");
//}
}
console.log(attachMediaStream(remoteVideo, event.stream));
trace('pc2 received remote stream');
}
function onCreateAnswerSuccess(desc) {
//trace('Answer from pc2:\n' + desc.sdp);
//trace('pc2 setLocalDescription start');
//pc2.setLocalDescription(desc, function () {
// onSetLocalSuccess(pc2);
//}, onSetSessionDescriptionError);
trace('pc1 setRemoteDescription start');
pc1.setLocalDescription(desc, function () {
onSetRemoteSuccess(pc1);
sendWS(JSON.stringify({ "sdp": desc }));
}, onSetSessionDescriptionError);
}
function onIceCandidate(pc, event) {
if (!event || !event.candidate) return;
sendWS(JSON.stringify({ "candidate": event.candidate }));
//var candidate = event.candidate;
//if (candidate) {
// getOtherPc(pc).addIceCandidate(new RTCIceCandidate(event.candidate),
// function () {
// onAddIceCandidateSuccess(pc);
// },
// function (err) {
// onAddIceCandidateError(pc, err);
// }
// );
// trace(getName(pc) + ' ICE candidate: \n' + event.candidate.candidate);
//}
}
function addIceCandidate(event) {
pc1.addIceCandidate(new RTCIceCandidate(event.candidate),
function () {
onAddIceCandidateSuccess(pc1);
},
function (err) {
onAddIceCandidateError(pc1, err);
}
);
trace(pc1 + ' ICE candidate: \n' + event.candidate.candidate);
}
function onAddIceCandidateSuccess(pc) {
trace(getName(pc) + ' addIceCandidate success');
}
function onAddIceCandidateError(pc, error) {
trace(getName(pc) + ' failed to add ICE Candidate: ' + error.toString());
}
function onIceStateChange(pc, event) {
if (pc) {
trace(getName(pc) + ' ICE state: ' + pc.iceConnectionState);
console.log('ICE state change event: ', event);
if (pc.iceConnectionState == 'disconnected' || pc.iceConnectionState == 'failed') {
EndCall();
}
}
}
function EndCall() {
hangup();
console.log("chatduration =" + $scope.offerConversationDetails.ChatDurationSeconds);
console.log("consumption =" + $scope.seconds);
//if ($scope.IsManuallyEnded != true)
//{
alertify.alert('Mepleez Conversation', 'Your conversation got finished. Please give conversation feedback to us.',
function () {
//hangup();
window.location.replace("/video.html#/experience");
});
//}
}
function callDateTimeExpired(message) {
alertify.alert('Mepleez Conversation', message,
function () {
// window.location.replace("/video.html#/experience");
window.location.replace("/index.html#/");
});
}
$scope.onEndCallBtnClick = function () {
//alertify.prompt('Cancel call', 'Please enter the reason for cancelling call', 'Busy with other stuff '
// , function (evt, value) {
// }
// , function () { });
alertify.confirm('End Call', 'Are you sure?', function () {
$scope.IsManuallyEnded = true;
hangup();
window.location.replace("/video.html#/experience");
}
, function () { });
}
function hangup() {
$scope.IsCallEnd = true;
trace('Ending call');
pc1.close();
// pc2.close();
pc1 = null;
// pc2 = null;
//hangupButton.disabled = true;
// callButton.disabled = false;
localStream.getTracks().forEach(function (track) {
track.stop();
});
localVideo.src = "";
remoteVideo.src = "";
closeWS();
// window.location.replace("/video.html#/experience");
}
$scope.dragOptions = {
start: function (e) {
console.log("STARTING");
},
drag: function (e) {
console.log("DRAGGING");
},
stop: function (e) {
console.log("STOPPING");
},
container: 'container'
}
There is some special characters will be getting in SDP after serlizing it.
So to remove it use below code
// Workaround
function maybeAddLineBreakToEnd(sdp) {
var endWithLineBreak = new RegExp(/\n$/);
if (!endWithLineBreak.test(sdp)) {
return sdp + '\n';
}
return sdp;
}
function gotDescription(desc) {
var offer = desc;
offerSdpTextarea.value = desc.sdp;
var sdp = offerSdpTextarea.value;
sdp = maybeAddLineBreakToEnd(sdp);
console.log(sdp);
sdp = sdp.replace(/\n/g, '\r\n');
offer.sdp = sdp;
pc1.setLocalDescription(offer,
onSetOfferSDPSuccess,
onSetSDPError);
trace('Modified Offer from localPeerConnection \n' + sdp);
// sendWS(JSON.stringify({ "sdp": desc }));
sendWS(JSON.stringify(offer));
}

how to use attachMediaStream?

I tried to test an exercise with WebRTC socket.io for a video call and chat .
I could not get the console errors on firefox but not attacking me the two local / remote stream between them .
my browser code is this
'use strict';
var p = navigator.mediaDevices.getUserMedia ({video: true, audio: true});
// Clean-up function:
// collect garbage before unloading browser's window
window.onbeforeunload = function(e){
hangup();
}
// Data channel information
var sendChannel, receiveChannel;
var sendButton = document.getElementById("sendButton");
var sendTextarea = document.getElementById("dataChannelSend");
var receiveTextarea = document.getElementById("dataChannelReceive");
// HTML5 <video> elements
var localVideo = document.querySelector('#localVideo');
var remoteVideo = document.querySelector('#remoteVideo');
// Handler associated with Send button
sendButton.onclick = sendData;
// Flags...
var isChannelReady = false;
var isInitiator = false;
var isStarted = false;
// WebRTC data structures
// Streams
var SStream;
var BStream;
var localStream;
var remoteStream;
// PeerConnection
var pc;
var pc_constraints = {
'optional': [
{'DtlsSrtpKeyAgreement': true}
]};
var sdpConstraints = {};
// Let's get started: prompt user for input (room name)
var room = prompt('Enter room name:');
// Connect to signaling server
var socket = io.connect("http://localhost:8181");
// Send 'Create or join' message to singnaling server
if (room !== '') {
console.log('Create or join room ', room);
socket.emit('create or join',room);
}
// Server-mediated message exchanging...
// 1. Server-->Client...
// this peer is the initiator
socket.on('created', function (room){
console.log('Created room ' + room);
isInitiator = true;
// Call getUserMedia()
p.then (function (MediaStream) {
SStream=MediaStream;
localVideo.src = window.URL.createObjectURL (MediaStream);
localVideo.onloadedmetadata = function (e) {
console.log("add local stream");
sendMessage('got user media');
};
});
p.catch (function (err) {console.log (err.name);});
checkAndStart();
});
// Handle 'join' message coming back from server:
// another peer is joining the channel
socket.on('join',function(room){
console.log('this peer is the iniator of room '+room+' !');
isChannelReady=true;
})
// Handle 'joined' message coming back from server:
// this is the second peer joining the channel
socket.on('joined', function (room){
console.log('This peer has joined room ' + room);
isChannelReady = true;
// Call getUserMedia()
p.then (function (MediaStream) {
isChannelReady = true;
BStream=MediaStream;
remoteVideo.src = window.URL.createObjectURL (MediaStream);
remoteVideo.onloadedmetadata = function (e) {
console.log("add remote stream");
sendMessage('got user media');
};
});
p.catch (function (err) {console.log (err.name);});
});
// Server-sent log message...
socket.on('log', function (array){
console.log.apply(console, array);
});
// Receive message from the other peer via the signaling server
socket.on('message', function (message){
console.log('Received message: ', message);
if (message === 'got user media') {
console.log('sono in if');
checkAndStart();
}
else if (message.type === 'offer') {
if (!isInitiator && !isStarted) {
checkAndStart();
}
pc.setRemoteDescription(new RTCSessionDescription(message));
doAnswer();
}
else if (message.type === 'answer' && isStarted) {
pc.setRemoteDescription(new RTCSessionDescription(message));
}
else if (message.type === 'candidate' && isStarted) {
var candidate = new RTCIceCandidate({sdpMLineIndex:message.label,candidate:message.candidate});
pc.addIceCandidate(candidate);
}
else if (message === 'bye' && isStarted) {
handleRemoteHangup();
}
});
// 2. Client-->Server
// Send message to the other peer via the signaling server
function sendMessage(message){
console.log('Sending message: ', message);
socket.emit('message', message);
}
// Channel negotiation trigger function
function checkAndStart() {
if (!isStarted && isChannelReady) {
createPeerConnection();
isStarted = true;
if (isInitiator) {
doCall();
}
}
}
// PeerConnection management...
function createPeerConnection() {
try {
/*posso aggiungere turn, google ecc...*/
pc = new RTCPeerConnection({ iceServers: [{ urls: 'stun:93.33.165.245' }] }, {
'optional': [
{'DtlsSrtpKeyAgreement': true}
]});
if(isInitiator){
SStream.getTracks().forEach(track => pc.addTrack(track,SStream));
pc.ontrack= handleRemoteStreamAdded;}
else{
BStream.getTracks().forEach(track => pc.addTrack(track,BStream));}
pc.onicecandidate = handleIceCandidate;
console.log('create RTCPeerConnection');
}
catch (e) {
console.log('Failed to create PeerConnection, exception: ' + e.message);
alert('Cannot create RTCPeerConnection object.');
return;
}
pc.ontrack= handleRemoteStreamAdded;
pc.onremovestream = handleRemoteStreamRemoved;
if (isInitiator) {
try {
// Create a reliable data channel
sendChannel = pc.createDataChannel("sendDataChannel",
{reliable: true});
console.log('Created send data channel');
} catch (e) {
alert('Failed to create data channel. ');
console.log('createDataChannel() failed with exception: ' + e.message);
}
sendChannel.onopen = handleSendChannelStateChange;
sendChannel.onmessage = handleMessage;
sendChannel.onclose = handleSendChannelStateChange;
} else { // Joiner
pc.ondatachannel = gotReceiveChannel;
}
}
// Data channel management
function sendData() {
var data = sendTextarea.value;
if(isInitiator) sendChannel.send(data);
else receiveChannel.send(data);
console.log('Sent data: ' + data);
}
// Handlers...
function gotReceiveChannel(event) {
console.log('Receive Channel Callback');
receiveChannel = event.channel;
receiveChannel.onmessage = handleMessage;
receiveChannel.onopen = handleReceiveChannelStateChange;
receiveChannel.onclose = handleReceiveChannelStateChange;
}
function handleMessage(event) {
console.log('Received message: ' + event.data);
receiveTextarea.value += event.data + '\n';
}
function handleSendChannelStateChange() {
var readyState = sendChannel.readyState;
console.log('Send channel state is: ' + readyState);
// If channel ready, enable user's input
if (readyState == "open") {
dataChannelSend.disabled = false;
dataChannelSend.focus();
dataChannelSend.placeholder = "";
sendButton.disabled = false;
} else {
dataChannelSend.disabled = true;
sendButton.disabled = true;
}
}
function handleReceiveChannelStateChange() {
var readyState = receiveChannel.readyState;
console.log('Receive channel state is: ' + readyState);
// If channel ready, enable user's input
if (readyState == "open") {
dataChannelSend.disabled = false;
dataChannelSend.focus();
dataChannelSend.placeholder = "";
sendButton.disabled = false;
} else {
dataChannelSend.disabled = true;
sendButton.disabled = true;
}
}
// ICE candidates management
function handleIceCandidate(event) {
console.log('handleIceCandidate 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.');
}
}
// Create Offer
function doCall() {
console.log('Creating Offer...');
pc.createOffer(setLocalAndSendMessage, onSignalingError, sdpConstraints);
}
// Signaling error handler
function onSignalingError(error) {
console.log('Failed to create signaling message : ' + error.name);
}
// Create Answer
function doAnswer() {
console.log('Sending answer to peer.');
pc.createAnswer(setLocalAndSendMessage, onSignalingError, sdpConstraints);
}
// Success handler for both createOffer()
// and createAnswer()
function setLocalAndSendMessage(sessionDescription) {
pc.setLocalDescription(sessionDescription);
sendMessage(sessionDescription);
}
// Remote stream handlers...
function handleRemoteStreamAdded(event) {
attachMediaStream(BStream, event.stream);
BStream = event.stream;
}
function handleRemoteStreamRemoved(event) {
console.log('Remote stream removed. Event: ', event);
}
// Clean-up functions...
function hangup() {
console.log('Hanging up.');
stop();
sendMessage('bye');
}
function handleRemoteHangup() {
console.log('Session terminated.');
stop();
isInitiator = false;
}
function stop() {
isStarted = false;
if (sendChannel) sendChannel.close();
if (receiveChannel) receiveChannel.close();
if (pc) pc.close();
pc = null;
sendButton.disabled=true;
}
someone tell me where I'm wrong and how to fix ?
attachMediaStream is not part of WebRTC. It's a shim adapter.js used to expose for setting video.src or video.srcObject (which Chrome still doesn't support, but Canary does).
In any case, you're passing in the wrong arguments, which should be an element and a stream, not two streams. I.e. make it:
attachMediaStream(remoteVideo, event.streams[0]);
or better, use the spec-way that adapter.js now supports on all browsers:
remoteVideo.srcObject = event.streams[0];
Important: The pc.ontrack event contains event.streams (plural), not event.stream! - It's an array since a track may (but rarely does) exist in more than one stream.
If you're not using the latest adapter.js, then note that pc.ontrack is only natively available in Firefox at the moment, so in Chrome you would need the older pc.onaddstream (and its event.stream).
PS: You're currently setting remoteVideo.src to the local video. Don't do that.
PPS: Remove pc_constraints. Really old stuff that will break Chrome.
Here's a demo of ontrack (use https fiddle in Chrome):
var pc1 = new RTCPeerConnection(), pc2 = new RTCPeerConnection();
navigator.mediaDevices.getUserMedia({ video: true, audio: true })
.then(stream => pc1.addStream(video1.srcObject = stream))
.catch(log);
var add = (pc, can) => can && pc.addIceCandidate(can).catch(log);
pc1.onicecandidate = e => add(pc2, e.candidate);
pc2.onicecandidate = e => add(pc1, e.candidate);
pc2.ontrack = e => video2.srcObject = e.streams[0];
pc1.oniceconnectionstatechange = e => log(pc1.iceConnectionState);
pc1.onnegotiationneeded = e =>
pc1.createOffer().then(d => pc1.setLocalDescription(d))
.then(() => pc2.setRemoteDescription(pc1.localDescription))
.then(() => pc2.createAnswer()).then(d => pc2.setLocalDescription(d))
.then(() => pc1.setRemoteDescription(pc2.localDescription))
.catch(log);
var log = msg => div.innerHTML += "<br>" + msg;
<video id="video1" height="120" width="160" autoplay muted></video>
<video id="video2" height="120" width="160" autoplay></video><br>
<div id="div"></div>
<script src="https://webrtc.github.io/adapter/adapter-latest.js"></script>

Webrtc Remote video stream not working

While setting the remoteDescription , I am getting below error in firefox :
DOMException [InvalidStateError: "Cannot set remote offer in state have-local-offer"
code: 11
nsresult: 0x8053000b
location: http://localhost:8080/resources/assets/js/test-online.js:111]
Please find below my test-online.js code
var localVideo;
var remoteVideo;
var peerConnection;
var serverConnection;
var peerConnectionConfig = {'iceServers': [{'url': 'stun:stun.services.mozilla.com'}, {'url': 'stun:stun.l.google.com:19302'}]};
pageReady();
var offerOptions = {
offerToReceiveAudio: 1,
offerToReceiveVideo: 1
};
var sdpConstraints = {'mandatory': {
'OfferToReceiveAudio':true,
'OfferToReceiveVideo':true }};
function pageReady() {
localVideo = document.getElementById('localVideo');
remoteVideo = document.getElementById('remoteVideo');
localVideo.addEventListener('loadedmetadata', function() {
trace('Local video videoWidth: ' + this.videoWidth +
'px, videoHeight: ' + this.videoHeight + 'px');
});
remoteVideo.addEventListener('loadedmetadata', function() {
trace('Remote video videoWidth: ' + this.videoWidth +
'px, videoHeight: ' + this.videoHeight + 'px');
});
remoteVideo.onresize = function() {
trace('Remote video size changed to ' +
remoteVideo.videoWidth + 'x' + remoteVideo.videoHeight);
// We'll use the first onsize callback as an indication that video has started
// playing out.
if (startTime) {
var elapsedTime = window.performance.now() - startTime;
trace('Setup time: ' + elapsedTime.toFixed(3) + 'ms');
startTime = null;
}
};
serverConnection = new SockJS("/onlineHandler");
serverConnection.onopen = function() {
console.log("Opening server connection");
};
serverConnection.onmessage = gotMessageFromServer;
serverConnection.onclose = function() {
console.log("Closing server connection");
};
//serverConnection.onmessage = gotMessageFromServer;
var constraints = {
video: true,
audio: true,
};
navigator.mediaDevices.getUserMedia(constraints)
.then(getUserMediaSuccess)
.catch(function(e) {
alert('getUserMedia() error: ' + e.name);
});
}
function getUserMediaSuccess(stream) {
trace('Received local stream');
localVideo.srcObject = stream;
localStream = stream;
}
function start(isCaller) {
trace('Starting call');
startTime = window.performance.now();
var videoTracks = localStream.getVideoTracks();
var audioTracks = localStream.getAudioTracks();
if (videoTracks.length > 0) {
trace('Using video device: ' + videoTracks[0].label);
}
if (audioTracks.length > 0) {
trace('Using audio device: ' + audioTracks[0].label);
}
peerConnection = new RTCPeerConnection(peerConnectionConfig);
peerConnection.onicecandidate = gotIceCandidate;
peerConnection.oniceconnectionstatechange = onIceStateChange;
peerConnection.onaddStream = gotRemoteStream;
peerConnection.addStream(localStream);
if(isCaller) {
peerConnection.createOffer(gotDescription, errorHandler , offerOptions);
}
}
function gotMessageFromServer(message) {
/* if(!peerConnection) start(false);
var signal = JSON.parse(message.data);
// console.log("Got Message from server :" + message.data);
if(signal.sdp) {;
console.log("hi in sdp" + message.data);
peerConnection.setRemoteDescription(new RTCSessionDescription(signal.sdp), function() {
console.log("Creating answer :");
if (peerConnection.remoteDescription.type == 'offer')
peerConnection.createAnswer(gotDescription, errorHandler);
}, errorHandler);
} else if(signal.ice) {
peerConnection.addIceCandidate(new RTCIceCandidate(signal.ice));
}*/
var signal = JSON.parse(message.data);
if (signal.type === 'offer') {
peerConnection.setRemoteDescription(new RTCSessionDescription(signal),doAnswer,errorHandler);
} else if (signal.type === 'answer') {
peerConnection.setRemoteDescription(new RTCSessionDescription(signal),doNothing, errorHandler);
} else if (signal.type === 'candidate') {
var candidate = new RTCIceCandidate({
sdpMLineIndex:signal.label,
candidate: signal.candidate
});
peerConnection.addIceCandidate(candidate);
} else if (signal === 'bye' && isStarted) {
handleRemoteHangup();
}
}
function doNothing(){
}
function doAnswer() {
console.log('Sending answer to peer.');
peerConnection.createAnswer(gotDescription, errorHandler, sdpConstraints);
}
function handleRemoteHangup() {
// console.log('Session terminated.');
// stop();
// isInitiator = false;
}
function gotIceCandidate(event) {
if(event.candidate != null) {
var message ={
type: 'candidate',
label: event.candidate.sdpMLineIndex,
id: event.candidate.sdpMid,
candidate: event.candidate.candidate};
// serverConnection.send(JSON.stringify({'ice': event.candidate}));
serverConnection.send(JSON.stringify(message));
}
}
function onIceStateChange(event) {
if (peerConnection) {
trace(' ICE state: ' + peerConnection.iceConnectionState);
console.log('ICE state change event: ', event);
}
}
function gotDescription(description) {
// trace('Offer from peerConnection\n' + description.sdp);
description.sdp = preferOpus(description.sdp);
// pc.setLocalDescription(description);
console.log('setLocalAndSendMessage sending message' , description);
// trace('peerConnection setLocalDescription start');
peerConnection.setLocalDescription(
description,
function () {
serverConnection.send(JSON.stringify(description));
},
onSetSessionDescriptionError
);
}
function preferOpus(sdp) {
var sdpLines = sdp.split('\r\n');
var mLineIndex;
// Search for m line.
for (var i = 0; i < sdpLines.length; i++) {
if (sdpLines[i].search('m=audio') !== -1) {
mLineIndex = i;
break;
}
}
if (mLineIndex === null) {
return sdp;
}
// If Opus is available, set it as the default in m line.
for (i = 0; i < sdpLines.length; i++) {
if (sdpLines[i].search('opus/48000') !== -1) {
var opusPayload = extractSdp(sdpLines[i], /:(\d+) opus\/48000/i);
if (opusPayload) {
sdpLines[mLineIndex] = setDefaultCodec(sdpLines[mLineIndex], opusPayload);
}
break;
}
}
// Remove CN in m line and sdp.
sdpLines = removeCN(sdpLines, mLineIndex);
sdp = sdpLines.join('\r\n');
return sdp;
}
function extractSdp(sdpLine, pattern) {
var result = sdpLine.match(pattern);
return result && result.length === 2 ? result[1] : null;
}
// Set the selected codec to the first in m line.
function setDefaultCodec(mLine, payload) {
var elements = mLine.split(' ');
var newLine = [];
var index = 0;
for (var i = 0; i < elements.length; i++) {
if (index === 3) { // Format of media starts from the fourth.
newLine[index++] = payload; // Put target payload to the first.
}
if (elements[i] !== payload) {
newLine[index++] = elements[i];
}
}
return newLine.join(' ');
}
// Strip CN from sdp before CN constraints is ready.
function removeCN(sdpLines, mLineIndex) {
var mLineElements = sdpLines[mLineIndex].split(' ');
// Scan from end for the convenience of removing an item.
for (var i = sdpLines.length-1; i >= 0; i--) {
var payload = extractSdp(sdpLines[i], /a=rtpmap:(\d+) CN\/\d+/i);
if (payload) {
var cnPos = mLineElements.indexOf(payload);
if (cnPos !== -1) {
// Remove CN payload from m line.
mLineElements.splice(cnPos, 1);
}
// Remove CN line in sdp
sdpLines.splice(i, 1);
}
}
sdpLines[mLineIndex] = mLineElements.join(' ');
return sdpLines;
}
function onSetSessionDescriptionError(error) {
trace('Failed to set session description: ' + error.toString());
}
function gotRemoteStream(event) {
remoteVideo.srcObject = event.stream;
trace('Received remote stream');
}
function errorHandler(error) {
console.log(error);
}
And my html code is below :
<%# taglib uri="http://java.sun.com/jsp/jstl/core" prefix="c"%>
<%# taglib uri="http://www.springframework.org/tags/form" prefix="form"%>
<html lang="en">
<head>
<link
href="//netdna.bootstrapcdn.com/font-awesome/4.0.3/css/font-awesome.css"
rel="stylesheet">
<!-- Meta tag to not followed by search engine. -->
<meta name="robots" content="noindex,nofollow,nosnippet,noodp,noarchive">
<meta name="keywords" content="JavaScript, WebRTC" />
<meta name="description" content="WebRTC codelab" />
<meta name="viewport" content="width=device-width,initial-scale=1,minimum-scale=1,maximum-scale=1">
<style>
video {
filter: hue-rotate(180deg) saturate(200%);
-moz-filter: hue-rotate(180deg) saturate(200%);
-webkit-filter: hue-rotate(180deg) saturate(200%);
/*-webkit-filter: grayscale(0.3) hue-rotate(360deg) saturate(10) opacity(0.7) sepia(0.5); */
}
</style>
</head>
<body>
<div id='videos'>
<video id='localVideo' autoplay muted></video>
<video id='remoteVideo' autoplay></video>
</div>
<input type="button" id="start" onclick="start(true)" value="Start Video"></input>
<script src="//cdn.jsdelivr.net/sockjs/1.0.0/sockjs.min.js"></script>
<script
src="${pageContext.request.contextPath}/resources/assets/js/jquery-2.1.1.min.js"></script>
<script
src="${pageContext.request.contextPath}/resources/assets/js/bootstrap.min.js"></script>
<script src ="${pageContext.request.contextPath}/resources/assets/js/adapter-0.2.10.js"></script>
<script src="${pageContext.request.contextPath}/resources/assets/js/test-online.js"></script>
</body>
</html>
I am not able to understand what I am doing wrong here.I am still a novice in webrtc field juts want to run this basic thing in my code.
The offer-answer exchange is a state machine, and some methods are disallowed in certain states.
Calling setLocalDescription (or setRemoteDescription) changes the signaling state to "have-local-offer" (or "have-remote-offer").
At that point, it is the application's job to bring the state back to stable or closed, as described in the spec.
For instance, it is the application's responsibility to handle glare (which is where both sides send an offer at the same time).
There is also a bug in Firefox that it doesn't allow you to call createOffer or setLocalDescription again once in have-local-offer and vice versa for the answer (the little hoops in the state diagram in the spec linked above). But it doesn't sound from your error message like you're hitting that.

ReferenceError:room is not defined, RTCpeerconnection not working. Clients do not connect

I am making a WebRTC video chat application and it was working before i started to add or subtract more code and in the process i deleted or changed the order in a way that now i am getting this error. Sadly i don't have a backup code and it has consumed so much of my time.
You will need socket.io and node-static packages installed on your node server.
Okay i managed to solve the issue with the error but now clients are not connecting with each other, it seems two clients are unable to exchange messages via the server.
My server.js code is below
var static = require('node-static');
var http = require('http');
var file = new(static.Server)();
var app = http.createServer(function (req, res) {
file.serve(req, res);
}).listen(8080, '127.0.0.1');
var io = require('socket.io').listen(app);
io.sockets.on('connection', function (socket){
socket.emit('emit(): client ' + socket.id + ' joined room ' + room);
socket.broadcast.emit('broadcast(): client ' + socket.id + ' joined room ' + room);
function log(){
var array = [">>> Message from server: "];
for (var i = 0; i < arguments.length; i++) {
array.push(arguments[i]);
}
socket.emit('log', array);
}
socket.on('message', function (message) {
log('Got message: ', message);
// For a real app, should be room only (not broadcast)
socket.broadcast.emit('message', message);
});
socket.on('create or join', function (room) {
var numClients = io.sockets.clients(room).length;
log('Room ' + room + ' has ' + numClients + ' client(s)');
log('Request to create or join room', room);
if (numClients == 0){
socket.join(room);
socket.emit('created', room);
} else if (numClients == 1) {
io.sockets.in(room).emit('join', room);
socket.join(room);
socket.emit('joined', room);
} else { // max two clients
socket.emit('full', room);
}
});
});
my application.js file is as follows
'use strict';
var isChannelReady;
var isInitiator = false;
var isStarted = false;
var localStream;
var pc;
var remoteStream;
var pc_config = webrtcDetectedBrowser === 'firefox' ?
{'iceServers':[{'url':'stun:23.21.150.121'}]} : // number IP
{'iceServers': [{'url': 'stun:stun.l.google.com:19302'}]};
var pc_constraints = {'optional': [{'DtlsSrtpKeyAgreement': true}]};
// Set up audio and video regardless of what devices are present.
var sdpConstraints = {'mandatory': {
'OfferToReceiveAudio':true,
'OfferToReceiveVideo':true }};
var room = location.pathname.substring(1);
if (room === '') {
room = window.prompt('Enter room name:');
room = '';
}
var socket = io.connect();
if (room !== '') {
console.log('Create or join room', room);
socket.emit('create or join', room);
}
socket.on('created', function (room){
console.log('Created room ' + room);
isInitiator = true;
});
socket.on('full', function (room){
console.log('Room ' + room + ' is full');
});
socket.on('join', function (room){
console.log('Another peer made a request to join room ' + room);
console.log('This peer is the initiator of room ' + room + '!');
isChannelReady = true;
});
socket.on('joined', function (room){
console.log('This peer has joined room ' + room);
isChannelReady = true;
});
socket.on('log', function (array){
console.log.apply(console, array);
});
////////////////////////////////////////////////
function sendMessage(message){
console.log('Client sending message: ', message);
// if (typeof message === 'object') {
// message = JSON.stringify(message);
// }
socket.emit('message', message);
}
socket.on('message', function (message){
console.log('Client received message:', message);
if (message === 'got user media') {
maybeStart();
} else if (message.type === 'offer') {
if (!isInitiator && !isStarted) {
maybeStart();
}
pc.setRemoteDescription(new RTCSessionDescription(message));
doAnswer();
} else if (message.type === 'answer' && isStarted) {
pc.setRemoteDescription(new RTCSessionDescription(message));
} else if (message.type === 'candidate' && isStarted) {
var candidate = new RTCIceCandidate({
sdpMLineIndex: message.label,
candidate: message.candidate
});
pc.addIceCandidate(candidate);
} else if (message === 'bye' && isStarted) {
handleRemoteHangup();
}
});
////////////////////////////////////////////////////
function handleUserMedia(stream) {
console.log('Adding local stream.');
localVideo.src = window.URL.createObjectURL(stream);
localStream = stream;
sendMessage('got user media');
if (isInitiator) {
maybeStart();
}
}
function handleUserMediaError(error){
console.log('getUserMedia error: ', error);
}
var constraints = {video: true, audio:true};
getUserMedia(constraints, handleUserMedia, handleUserMediaError);
console.log('Getting user media with constraints', constraints);
function maybeStart() {
if (!isStarted && typeof localStream != 'undefined' && isChannelReady) {
createPeerConnection();
pc.addStream(localStream);
isStarted = true;
console.log('isInitiator', isInitiator);
if (isInitiator) {
doCall();
}
}
}
window.onbeforeunload = function(e){
sendMessage('bye');
}
/////////////////////////////////////////////////////////
function createPeerConnection() {
try {
pc = new RTCPeerConnection(null);
pc.onicecandidate = handleIceCandidate;
pc.onaddstream = handleRemoteStreamAdded;
pc.onremovestream = handleRemoteStreamRemoved;
console.log('Created RTCPeerConnnection');
} catch (e) {
console.log('Failed to create PeerConnection, exception: ' + e.message);
alert('Cannot create RTCPeerConnection object.');
return;
}
}
function handleIceCandidate(event) {
console.log('handleIceCandidate 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.');
}
}
function handleRemoteStreamAdded(event) {
console.log('Remote stream added.');
remoteVideo.src = window.URL.createObjectURL(event.stream);
remoteStream = event.stream;
}
function handleCreateOfferError(event){
console.log('createOffer() error: ', e);
}
function doCall() {
console.log('Sending offer to peer');
pc.createOffer(setLocalAndSendMessage, handleCreateOfferError);
}
function doAnswer() {
console.log('Sending answer to peer.');
pc.createAnswer(setLocalAndSendMessage, null, sdpConstraints);
}
function setLocalAndSendMessage(sessionDescription) {
// Set Opus as the preferred codec in SDP if Opus is present.
sessionDescription.sdp = preferOpus(sessionDescription.sdp);
pc.setLocalDescription(sessionDescription);
console.log('setLocalAndSendMessage sending message' , sessionDescription);
sendMessage(sessionDescription);
}
function handleRemoteStreamAdded(event) {
console.log('Remote stream added.');
remoteVideo.src = window.URL.createObjectURL(event.stream);
remoteStream = event.stream;
}
function handleRemoteStreamRemoved(event) {
console.log('Remote stream removed. Event: ', event);
}
function hangup() {
console.log('Hanging up.');
stop();
sendMessage('bye');
}
function handleRemoteHangup() {
// console.log('Session terminated.');
// stop();
// isInitiator = false;
}
function stop() {
isStarted = false;
// isAudioMuted = false;
// isVideoMuted = false;
pc.close();
pc = null;
}
///////////////////////////////////////////
// Set Opus as the default audio codec if it's present.
function preferOpus(sdp) {
var sdpLines = sdp.split('\r\n');
var mLineIndex = null;
// Search for m line.
for (var i = 0; i < sdpLines.length; i++) {
if (sdpLines[i].search('m=audio') !== -1) {
mLineIndex = i;
break;
}
}
if (mLineIndex === null) {
return sdp;
}
// If Opus is available, set it as the default in m line.
for (i = 0; i < sdpLines.length; i++) {
if (sdpLines[i].search('opus/48000') !== -1) {
var opusPayload = extractSdp(sdpLines[i], /:(\d+) opus\/48000/i);
if (opusPayload) {
sdpLines[mLineIndex] = setDefaultCodec(sdpLines[mLineIndex], opusPayload);
}
break;
}
}
// Remove CN in m line and sdp.
sdpLines = removeCN(sdpLines, mLineIndex);
sdp = sdpLines.join('\r\n');
return sdp;
}
function extractSdp(sdpLine, pattern) {
var result = sdpLine.match(pattern);
return result && result.length === 2 ? result[1] : null;
}
// Set the selected codec to the first in m line.
function setDefaultCodec(mLine, payload) {
var elements = mLine.split(' ');
var newLine = [];
var index = 0;
for (var i = 0; i < elements.length; i++) {
if (index === 3) { // Format of media starts from the fourth.
newLine[index++] = payload; // Put target payload to the first.
}
if (elements[i] !== payload) {
newLine[index++] = elements[i];
}
}
return newLine.join(' ');
}
// Strip CN from sdp before CN constraints is ready.
function removeCN(sdpLines, mLineIndex) {
var mLineElements = sdpLines[mLineIndex].split(' ');
// Scan from end for the convenience of removing an item.
for (var i = sdpLines.length-1; i >= 0; i--) {
var payload = extractSdp(sdpLines[i], /a=rtpmap:(\d+) CN\/\d+/i);
if (payload) {
var cnPos = mLineElements.indexOf(payload);
if (cnPos !== -1) {
// Remove CN payload from m line.
mLineElements.splice(cnPos, 1);
}
// Remove CN line in sdp
sdpLines.splice(i, 1);
}
}
sdpLines[mLineIndex] = mLineElements.join(' ');
return sdpLines;
}
Index.html
<!DOCTYPE html>
<html>
<head>
<meta name='keywords' content='WebRTC, HTML5, JavaScript' />
<meta name='description' content='WebRTC Reference App' />
<meta name='viewport' content='width=device-width,initial-scale=1,minimum-scale=1,maximum-scale=1'>
<base target='_blank'>
<title>WebRTC client</title>
<link rel='stylesheet' href='css/style.css' />
</head>
<body id='body'>
<p style=" font-size:24px" align="center">WebRTC Video Share</p>
<div id='container'>
<div>
<video id='localVideo' autoplay muted></video>
<video id='remoteVideo' autoplay></video>
</div>
</div>
<script src='/socket.io/socket.io.js'></script>
<script src='js/lib/adapter.js'></script>
<script src='js/main.js'></script>
</body>
</html>
In your server.js code:
var io = require('socket.io').listen(app);
io.sockets.on('connection', function (socket){
socket.emit('emit(): client ' + socket.id + ' joined room ' + room);
at the end you refer to a variable room, but you haven't created it yet.

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