webRTC ReferenceError: webkitRTCPeerConnection is not defined - webrtc

I am study about learning WebRTC book and create a demo 4 chapter. I am gating an error in console:
ReferenceError: webkitRTCPeerConnection is not defined
and not understand what can I confi the "iceServers":
Here is my javascript code
function hasUserMedia(){
navigator.getUserMedia = navigator.getUserMedia || navigator.webkitGetUserMedia || navigator.mozGetUserMedia || navigator.msGetUserMedia;
return !!navigator.getUserMedia;
}
function hasRTCPeerConnection() {
window.RTCPeerConnection = window.RTCPeerConnection || window.webkitRTCPeerConnection || window.mozRTCPeerConnection;
return !!window.RTCPeerConnection;
}
//This function will create our RTCPeerConnection objects, set up the SDP offer and response, and find the ICE candidates for both peers. page 48
function startPeerConnection(stream) {
var configuration = {
// Uncomment this code to add custom iceServers
"iceServers": [{ "url": "stun:stun.1.google.com:19302" }]
};
yourConnection = new webkitRTCPeerConnection(configuration);
theirConnection = new webkitRTCPeerConnection(configuration);
// Setup stream listening
yourConnection.addStream(stream);
theirConnection.onaddstream = function (e) {
theirVideo.src = window.URL.createObjectURL(e.stream);
};
// Setup ice handling
yourConnection.onicecandidate = function (event) {
if (event.candidate){
theirConnection.addIceCandidate(new RTCIceCandidate(event.candidate));
}
};
theirConnection.onicecandidate = function (event) {
if (event.candidate) {
yourConnection.addIceCandidate(new RTCIceCandidate(event.candidate));
}
};
// Begin the offer
yourConnection.createOffer(function (offer) {
yourConnection.setLocalDescription(offer);
theirConnection.setRemoteDescription(offer);
theirConnection.createAnswer(function (offer) {
theirConnection.setLocalDescription(offer);
yourConnection.setRemoteDescription(offer);
});
});
}
var yourVideo = document.querySelector("#yours"),
theirVideo = document.querySelector("#theirs"),
yourConnection, theirConnection;
if (hasUserMedia()) {
navigator.getUserMedia({ video: true, audio: false }, function (stream) {
yourVideo.src = window.URL.createObjectURL(stream);
if (hasRTCPeerConnection()) {
startPeerConnection(stream);
} else {
alert("Sorry, your browser does not support WebRTC.");
}
}, function (error) {
console.log(error);
}
);
} else {
alert("Sorry, your browser does not support WebRTC.");
}
and it output like this..
Please let me know why my video not working properly? Please help me to do this
learning WebRTC

Change:
yourConnection = new webkitRTCPeerConnection(configuration);
into:
yourConnection = new RTCPeerConnection(configuration);
as webkitRTCPeerConnection is for Chrome browsers, and the code already defines window.RTCPeerConnection in hasRTCPeerConnection so that it works for most browsers (inclusing Firefox that you are using).
[EDIT]
Your logic is not correct in this program. You are creating both connections like this:
yourConnection = new webkitRTCPeerConnection(configuration);
theirConnection = new webkitRTCPeerConnection(configuration);
This is not logical. Your program is one peer of a 2-peers connection. You need to setup your connection only. Also, you need some kind of messaging server to transmit SDP messages between the two peers. This is not the role of the ICE server.
Your ICE configuration is fine. You are using a public Google STUN server to handle the streaming and the public IP discovery necessary for establishing the WebRTC connection.

Related

switch camera during call peer js webrtc?

we are using peer js webrtc for video call. Everything is working fine just the problem is i am not able to switch camera during call. I have done some work where i can switch camera in local during call but its doesnt effect on remote area.
here is my code
$('select').on('change', function (e) {
navigator.mediaDevices.enumerateDevices().then(function (devices) {
var valueSelected = $("#myselect option:selected").val();
alert(valueSelected);
//var myselect = 0;
if (valueSelected == "0") {
var cameras = [];
devices.forEach(function (device) {
'videoinput' === device.kind && cameras.push(device.deviceId);
});
var constraints = { video: { deviceId: { exact: cameras[0] } } };
navigator.mediaDevices.getUserMedia(constraints).then(function (stream) {
window.localStream = stream;
myapp.setMyVideo(window.localStream)
//if (callback)
// callback();
}, function (err) {
console.log("The following error occurred: " + err.name);
alert('Unable to call ' + err.name)
});
}
else {
var cameras = [];
devices.forEach(function (device) {
'videoinput' === device.kind && cameras.push(device.deviceId);
});
var constraints = { video: { deviceId: { exact: cameras[1] } } };
navigator.mediaDevices.getUserMedia(constraints).then(function (stream) {
window.localStream = stream;
myapp.setMyVideo(window.localStream)
//if (callback)
// callback();
}, function (err) {
console.log("The following error occurred: " + err.name);
alert('Unable to call ' + err.name)
});
}
//var myselect = $("#myselect option:selected").val();
});
});
The recommended way to change stream when a peer-to-peer connection is established is to use replaceTrack function that does not require ICE renegotiation:
RTCRtpSender.replaceTrack
The documentation says:
Among the use cases for replaceTrack() is the common need to switch between the rear- and front-facing cameras on a phone. With replaceTrack(), you can simply have a track object for each camera and switch between the two as needed. See the example...

webrtc and peerjs: how to play a stream without starting his own stream?

I am using peerjs
Media call
var getUserMedia = navigator.getUserMedia || navigator.webkitGetUserMedia || navigator.mozGetUserMedia;
getUserMedia({video: true, audio: true}, function(stream) {
var call = peer.call('another-peers-id', stream);
call.on('stream', function(remoteStream) {
// Show stream in some video/canvas element.
});
}, function(err) {
console.log('Failed to get local stream' ,err);
});
Answer
var getUserMedia = navigator.getUserMedia || navigator.webkitGetUserMedia || navigator.mozGetUserMedia;
peer.on('call', function(call) {
getUserMedia({video: true, audio: true}, function(stream) {
call.answer(stream); // Answer the call with an A/V stream.
call.on('stream', function(remoteStream) {
// Show stream in some video/canvas element.
});
}, function(err) {
console.log('Failed to get local stream' ,err);
});
});
Problem is: to get the remoteStream, I need to show my own stream
var call = peer.call('another-peers-id', stream);
How can I play someone else stream without having to show my own stream ?
This is NOT a bug, you just got confused how WebRTC API works. We first create a function variable named getUserMedia which will be mapped to native webrtc method provided by browser (for example in case the browser is Chrome then method navigator.getUserMedia will be assigned to this variable and rest will be null. Similarly if browser is Firefox then native method navigator.mozGetUserMedia is assigned to this variable and rest will be null and so on).
Once appropriate native method is assigned to the variable getUserMedia, we then call this method with appropriate arguments. This first argument is type of call we want to create (i.e. media constraints), e.g. audio only, audio + video and so on. The second argument is a callback in case of success (i.e. when browser successfully able to access and connect to local mic and/or camera etc.). An example of this callback function would be,
function(stream) {
var video = document.querySelector('video');
video.srcObject = stream;
video.onloadedmetadata = function(e) {
// Do something with the video here.
};
}
There is a third argument for getUserMedia method which is optional and missing in your example code, this argument is a callback to another method which will kick-in in case getUserMedia method fails. There can be many reasons for failure, e.g. user disallowed the browser to acccess local mic or camera etc. Using this call back method you can inform user about call failure issues.
Here is full example code (Ref. https://developer.mozilla.org/en-US/docs/Web/API/Navigator/getUserMedia)
var getUserMedia = navigator.getUserMedia ||
navigator.webkitGetUserMedia ||
navigator.mozGetUserMedia;
if (getUserMedia) {
getUserMedia({ audio: true, video: { width: 1280, height: 720 } },
function(stream) {
var video = document.querySelector('video');
video.srcObject = stream;
video.onloadedmetadata = function(e) {
video.play();
};
},
function(err) {
console.log("The following error occurred: " + err.name);
}
);
} else {
console.log("getUserMedia not supported");
}
If you do not want to send your own audio and/or video, then simply set it to false in media constraints variable.
Hope this helps.

ICE failed, PeerJS call started but no video/audio through

I am trying to get a simple video chat working with PeerJS. I want to send audio between Firefox on a pc and Firefox on Android. I can get the call running (call.on) and I can view local video but for some reason the media just doesn't come through to the other user. Currently I am getting the error:
ICE failed, see about:webrtc for more details
I have a server which in its simple version is as such:
var ip = require('ip');
var PeerServer = require('peer').PeerServer;
var port = 9000;
var server = new PeerServer({port: port, allow_discovery: true});
Then I have two clients, one for the pc that makes the call:
var SERVER_IP = window.location.hostname;
var SERVER_PORT = 9000;
var localStream = "";
var peerID = "pc"
var peerConnectionID = "and"
var remoteVideo = document.querySelector('#rremote-video');
var localVideo = document.querySelector('#llocal-video');
var peer = new Peer(peerID, {host: SERVER_IP, port: SERVER_PORT});
var conn = peer.connect(peerConnectionID);
var getUserMedia = navigator.mediaDevices.getUserMedia({ video: true, audio: true })
.then(stream => localVideo.srcObject = stream)
.then(stream => localStream = stream)
.catch(e => console.log(e.name + ": "+ e.message));
waitForElement();
function waitForElement(){
if(localStream != ""){
conn.on('open', function(){
conn.send('hi from PC!');
});
peer.on('connection', function(conn) {
conn.on('data', function(data){
console.log(data);
});
});
console.log("we have a stream: "+localStream);
var call = peer.call(peerConnectionID, localStream);
console.log("Calling "+peerConnectionID);
call.on('stream', function(remotestream) {
console.log("Call on.");
remoteVideo.srcObject = remotestream;
});
}
else{
setTimeout(function(){
waitForElement();
},750);
}
}
And the one that answers the call is:
var SERVER_IP = window.location.hostname;
var SERVER_PORT = 9000;
var localStream = "";
var peerID = "and"
var peerConnectionID = "pc"
var remoteVideo = document.querySelector('#rremote-video');
var localVideo = document.querySelector('#llocal-video');
var remoteAudio = document.querySelector('#remote-audio');
var localAudio = document.querySelector('#local-audio');
var peer = new Peer(peerID, {host: SERVER_IP, port: SERVER_PORT});
var conn = peer.connect(peerConnectionID);
var getUserMedia = navigator.mediaDevices.getUserMedia({ video: true, audio: true })
.then(stream => localAudio.srcObject = stream)
.then(stream => localVideo.srcObject = stream)
.then(stream => localStream = stream)
.catch(e => console.log(e.name + ": "+ e.message));
waitForElement();
function waitForElement(){
if(localStream != ""){
conn.on('open', function(){
conn.send('hi from android!');
});
peer.on('connection', function(conn) {
conn.on('data', function(data){
console.log(data);
});
});
peer.on('call', function(call) {
console.log("Picking up call.");
call.answer(localStream);
call.on('stream', function(remotestream) {
console.log("Call on.");
remoteVideo.srcObject = remotestream;
});
});
}
else{
setTimeout(function(){
waitForElement();
},750);
}
}
I think it is some little tweak that I'm getting wrong, I have mainly followed instructions on PeerJS website: http://peerjs.com/ Please if anyone can see something that needs to change, any help is welcome!
Are you using https? Making calls to non-local machines is no longer allowed by the browsers.
To test this out, run both sets of code on your local machine. If you can do that connection, it means your code is ok.
To do a remote connection you will unfortunately need https. This means you will also need your own peerjs server (to run as https).
The other option is to use port forwarding to make one of the machines think it is talking to the localhost
It sounds like your ICE Candidates cannot communicate one to each other. You will have to use a STUN server and, if it still doesnt work, you will need a TURN server.
From PeerJS Documentation:
var peer = new Peer({
config: {'iceServers': [
{ url: 'stun:stun.l.google.com:19302' },
{ url: 'turn:homeo#turn.bistri.com:80', credential: 'homeo' }
]} /* Sample servers, please use appropriate ones */
});
This link will provide you a method to deploy your own TURN server.

DOMException: Failed to set remote offer sdp: Called in wrong state: STATE_SENTOFFER

I am trying to make a video calling web application using webRTC. I am using angularjs and express.io
I am getting this error:
DOMException: Failed to set remote offer sdp: Called in wrong state: STATE_SENTOFFER
Some of my code is:
// in controller (socket is already defined in controller)
var videolocal = document.getElementById('videolocal');
var videoremote = document.getElementById('videoremote');
var streamlocal = null;
var pc = null;
window.URL = window.URL || window.webkitURL;
navigator.getUserMedia = navigator.getUserMedia || navigator.webkitGetUserMedia || navigator.mozGetUserMedia || navigator.msGetUserMedia;
window.RTCPeerConnection = window.RTCPeerConnection || window.mozRTCPeerConnection || window.webkitRTCPeerConnection;
var configuration = {'iceServers': [
// {'url': 'stun:stun.services.mozilla.com'},
{'url': 'stun:stun.l.google.com:19302'}
]};
// run start(true) to initiate a call
$scope.start = function() {
console.log('start');
// get the local stream, show it in the local video element and send it
navigator.getUserMedia({ "audio": true, "video": true }, function (stream) {
videolocal.src = URL.createObjectURL(stream);
pc = new RTCPeerConnection(configuration);
pc.addStream(stream);
// once remote stream arrives, show it in the remote video element
pc.onaddstream = function (evt) {
console.log('onaddstream');
videoremote.src = URL.createObjectURL(evt.stream);
};
// send any ice candidates to the other peer
pc.onicecandidate = function (evt) {
console.log('onicecandidate');
if(evt.candidate){
socket.emit('video_call',{user:2, type: 'candidate', candidate: evt.candidate});
}
};
// create an offer
pc.createOffer(function (offer) {
socket.emit('video_call', {user:2, type: "offer", offer: offer});
pc.setLocalDescription(offer);
}, function (error) {
alert("Error when creating an offer");
});
}, function () {alert('error in start')});
}
$scope.start();
socket.on('video_call', function (data) {
console.log(data);
//when somebody sends us an offer
function handleOffer(offer) {
// this line is giving error
pc.setRemoteDescription(new RTCSessionDescription(offer), function(){alert('success')}, function(e){ console.log(e); alert(e)});
//create an answer to an offer
pc.createAnswer(function (answer) {
pc.setLocalDescription(answer);
socket.emit('video_call', {user:2, type: "answer", answer: answer});
}, function (error) {
console.log(error);
alert("Error when creating an answer");
});
};
//when we got an answer from a remote user
function handleAnswer(answer) {
pc.setRemoteDescription(new RTCSessionDescription(answer));
};
//when we got an ice candidate from a remote user
function handleCandidate(candidate) {
pc.addIceCandidate(new RTCIceCandidate(candidate));
};
switch(data['type']) {
case "offer":
handleOffer(data["offer"]);
break;
case "answer":
handleAnswer(data['answer']);
break;
//when a remote peer sends an ice candidate to us
case "candidate":
handleCandidate(data['candidate']);
break;
default:
break;
}
});
On server:
// this function is called on video_call event
video_call: function (data) {
var id = data.user;
// if user is active
// users is dict of users (user_id as key)
if(Object.keys(users).indexOf(id.toString()) > -1){
// for each device of the user
users[id].forEach(function(user_socket){
console.log(data);
user_socket.emit('video_call', data);
});
}
}
Please can anyone tell what is wrong with this code. The local stream is capturing properly. I am using chromium browser.
Data on server:
I think the problem is that in your handleOffer() function you need to create another PeerConnection and call setRemoteDescription() on that pc.
var remote_pc = new RTCPeerConnection(configuration)
remote_pc.setRemoteDescription(new RTCSessionDescription(offer), ...) {
remote_pc.createAnswer()
}
This is what I have in my code.
EDIT: In the official link you can go to chapter 11.7 and check the steps after 15 (when the offer is sent and the other peer receives it).

WebRTC using promises - Remote Video not seen at either end

I had earlier posted some questions on this problem. At that time I had two separate programs for caller and receiver. I was also using old-fashioned callback API. Thanks to help from #jib on that post, I was able to understand the need for some fundamental changes. I rewrote the program to make it an integrated one for both caller and receiver and have used the WebRTC promises API. My problem is that I am not getting remote video from either end. One part I understand but do not know the solution: The receiver does not create SDPs for Video in the first place, only for audio. The caller part does create SDPS for Video and audio but on the receiver end there is no event generated for remote stream.
I have checked, through console logs, that the core functions work. Offer SDP is created, sent out, received, answer SDP created, sent out, received, etc. Candidates get exchanged and added too. But the .onaddstream event handler is never triggered. Local video is shown but that is trivial.
I have spent a LOT of time on this. I simply need to get that exciting feeling of seeing remote video on both ends which has kept me going. ANY HELP WILL BE SINCERELY APPRECIATED.
<script>
$(document).ready(function () {
var iceCandidates = [], countIceCandidates=0;
var socket = io.connect();
socket.on('connect',function() { console.log("Socket connected"); });
var pc = new RTCPeerConnection({"iceServers":[{"url":"stun:stun.l.google.com:19302"}]});
//If remote video stream comes in, display it in DIV vid2
pc.onaddStream = function (event) {
stream = event.stream;
var video = $('#vid2');
video.attr('src', URL.createObjectURL(stream));
video.onloadedmetadata = function(e) { video.play(); }
}
//Display media in both Caller and Receiver
navigator.mediaDevices.getUserMedia({ audio: true, video: true })
.then(function(stream) {
var video = $('#vid1');
video.attr('src', URL.createObjectURL(stream));
video.onloadedmetadata = function(e) { video.play(); };
pc.addStream(stream);
})
.catch(function(err) { console.log(err);});
//INITIATE CALL
$('#call').click(function() {
pc.createOffer({ offerToReceiveVideo: true, offerToReceiveAudio: true })
.then(function(offer) {
localSessionDescription = new RTCSessionDescription(offer);
pc.setLocalDescription(localSessionDescription)
.then (function() { socket.emit('sdpOffer',localSessionDescription); })
.catch(function(err) { console.log("Error in setLocalDescription"); console.log(err); })
.catch(function(err) { console.log("Error in createOffer"); console.log(err); })
});
})
pc.onicecandidate = function (event) {
socket.emit('candidate',event.candidate);
};
socket.on('candidate',function (data) {
if (data != null) {
pc.addIceCandidate(new RTCIceCandidate(data))
.then(function() { console.log("peer candidate added");})
.catch(function(err) {console.log(err); console.log("Error during peer candidate addition");});
}
});
socket.on('disconnect',function() { alert("Disconnected"); });
function error(err) {
console.log("The following error occurred: " + err.name);
}
socket.on('sdpAnswer',function(data) {
sdpAnswer = new RTCSessionDescription(data.sdpAnswer);
pc.setRemoteDescription(sdpAnswer)
.then(function() { console.log("Answer SDP Set:"); console.log(sdpAnswer); })
.catch(function(err) { console.log("Error enountered when setting remote SDP Answer"); console.log(err)});
});
socket.on('sdpOffer', function(data) {
sdpOffer = new RTCSessionDescription(data.sdpOffer);
pc.setRemoteDescription(sdpOffer)
.then(function() { console.log("Remote SDP set in receiver");
pc.createAnswer()
.then(function(sdpAnswer) {
localSessionDescription = new RTCSessionDescription(sdpAnswer);
socket.emit('sdpAnswer',localSessionDescription);
pc.setLocalDescription(localSessionDescription)
.then(function(){
console.log("Local SDP Description set in receiver:");
})
.catch(function(err) { console.log("Error enountered when setting local SDP in receiver"); console.log(err)});
})
.catch(function(err) { console.log("Error enountered when creating answer SDP in receiver"); console.log(err)});
});
});
}); //End of document.ready function
</script>
ON THE SERVER SIDE (RELEVANT CODE ONLY). I have included here just in case there are any datatype related issues - object types, etc. getting changed when sent thru the server.
io.sockets.on('connection', function(socket) {
socket.on('sdpOffer', function(data) {
sdpOffer = data.sdp;
socket.broadcast.emit('sdpOffer',{"sdpOffer":data});
});
socket.on('sdpAnswer', function(data) {
sdpAnswer = data.sdp;
socket.broadcast.emit('sdpAnswer',{"sdpAnswer":data});
});
socket.on('candidate', function(data) {
socket.broadcast.emit('candidate',data);
});
});
Rename pc.onaddStream to pc.onaddstream.