I've updated my code with the complete signaling exchange. The problem now, is that upon completing the exchange, "socket.on('receivedAnswer')" throws an error. I'm testing all this on my local machine in two browser tabs so I assume I don't need ICE just yet for this to work...
<html>
<head>
<link rel = 'stylesheet' type = 'text/css' href= 'css.css'>
<script src="node_modules/socket.io/node_modules/socket.io-client/socket.io.js"></script>
</head>
<body>
<div id='video_box'>
<video id= 'video' autoplay="true">
</video>
</div>
<div id='video_box2'>
<video id='video2' autoplay="true">
</video>
<div>
<script>
var local_stream;
var baseURL = getBaseURL();
var socketIOPort = 8999;
var socketIOLocation = baseURL + socketIOPort;
var socket = io(socketIOLocation);
var localvid = document.getElementById('video');
var mediaOptions = { audio: false, video: true };
var pc = new mozRTCPeerConnection( {"iceServers": [{"url": "stun:stun.1.google.com:19302" }] });
var pc2 = new mozRTCPeerConnection();
var PeerConnection = window.RTCPeerConnection || window.mozRTCPeerConnection || window.webkitRTCPeerConnection;
var SessionDescription = window.RTCSessionDescription || window.mozRTCSessionDescription || window.webkitRTCSessionDescription;
var offerConstraints = {OfferToReceiveAudio: true, OfferToReceiveVideo: true }
socket.on('receivedAnswer', function(answerSDP){
pc.setRemoteDescription(new mozRTCSessionDescription(answerSDP), function(){
alert('received the answer');
}, error2);
});
socket.on('getPeer1', function(SDP) {
alert('got peer 1 SDP');
pc.setRemoteDescription(new mozRTCSessionDescription(SDP), function() {
pc.createAnswer(function(answerSDP){
pc.setLocalDescription(answerSDP, function() {
socket.emit('answerSDP', answerSDP);
alert('sending answer');
}, error2);
}, error2);
}, error2);
});
function start(){
checkMedia();
navigator.getUserMedia(mediaOptions, getMediaSuccess, error2);
}
function answer(offeredSDP){
offeredSDP = new SessionDescription(offeredSDP);
}
function peer(){
pc.addStream(local_stream);
pc.createOffer(function(SDP){
socket.emit('sendSDPtoServer', SDP);
}, error2, offerConstraints);
}
function getMediaSuccess(stream){
localvid.src = window.URL.createObjectURL(stream);
local_stream = stream;
peer(); ///////////
}
function error2(){
alert('error');
}
function error3(){
alert('error here');
}
function checkMedia() {
if (!navigator.getUserMedia) {
navigator.getUserMedia = navigator.getUserMedia || navigator.webkitGetUserMedia || navigator.mozGetUserMedia || navigator.msGetUserMedia;
}
if (!navigator.getUserMedia){
alert('getUserMedia not supported in this browser.');
}
}
function getBaseURL(){
baseURL = location.protocol + "//" + location.hostname + ":" + location.port;
return baseURL;
}
start();
//alert('script');
</script>
</body>
</html>
Server
var static = require('node-static');
var express = require('express');
var app = express();
var port = 8999;
var http = require('http');
var file= new(static.Server)();
var io = require('socket.io').listen(app.listen(port));
var connectedClients = {};
var peer1 = 0;
var peer2 = 0;
var peer1sdp = 0;
var peer2sdp = 0;
app.set('views', __dirname)
.engine('html', require('ejs').renderFile)
.use(express.static(__dirname + '/public'))
.get('/', function(req, res) {
//res.render('indexcpy.html');
});
io.on('connection', function(socket) {
console.log('a user connected');
socket.on('disconnect', function(){
console.log('user disconnected');
});
socket.on('answerSDP', function(answerSDP){
console.log('ANSWER SENT');
io.to(peer1).emit('receivedAnswer', answerSDP);
});
socket.on('sendSDPtoServer', function(SDP) {
if(peer1 == 0){
console.log('peer 1 ' + socket.id + ' has sent its SDP to server');
peer1 = socket.id;
peer1sdp = SDP;
} else {
console.log('peer 2 ' + socket.id + ' has been sent to the server');
socket.emit('getPeer1', peer1sdp);
}
});
});
the mistake here is your assumption that, pc.addStream(stream) would fire pc.onstream(event) on the same peer, but it would actually be fired on the remote peer.
Reference
other that that, your webrtc code is incomplete, while your offer SDP reaches the server, it is not sent to the remote peer and also it's answer has to be forwarded to your peer and so on...
You have to make sure that before creating Answer your local stream is added to the peerconnection. It will be more clear if you show where you are generating the answer
Related
I'm trying to make a video call using WebRTC on the local same machine. I can see the remote video on Firefox. But I can't see it on Chrome. When I console logged my code I found that event.candidate is null for both localPeerConnection.onicecandidate and remotePeerConnection.onicecandidate. So, I tested on Chrome again and found out that the connectionStatus is "new" and the iceGatheringState is "complete". On trying out Trickle ICE (https://webrtc.github.io/samples/src/content/peerconnection/trickle-ice/), I found that I wasn't getting any ice options on Chrome, but was for Firefox.
Thank you :)
Here's my code
<!DOCTYPE html>
<html lang="en">
<head>
<meta charset="UTF-8">
<title>WebRTC</title>
</head>
<body>
<video id="local" playsinline muted autoplay controls></video>
<video id="remote" playsinline autoplay controls></video>
<button id="startButton">Start</button>
<button id="callButton">Call</button>
<button id="hangupButton">Hang UP</button>
<script type="text/javascript">
// Stream info
var localStream, localPeerConnection, remotePeerConnection;
// Video tags
var localVideo = document.getElementById("local");
var remoteVideo = document.getElementById("remote");
// Buttons
var startButton = document.getElementById("startButton");
var callButton = document.getElementById("callButton");
var hangupButton = document.getElementById("hangupButton");
startButton.disabled = false;
callButton.disabled = true;
hangupButton.disabled = true;
var servers = null;
startButton.onclick = start;
callButton.onclick = call;
hangupButton.onclick = hangup;
function log(text){
console.log("At time"+(performance.now()/1000).toFixed(3)+"-->\n"+ text);
}
function successCallback(stream){
log("Received local stream");
if(navigator.mediaDevices.getUserMedia){
localVideo.srcObject = stream;
}
else{
console.error("GetUserMediaError",error);
}
localStream = stream;
callButton.disabled = false;
}
var constraints = {audio: true, video: true};
function start(){
log("Requesting local stream");
startButton.disabled = true;
navigator.mediaDevices.getUserMedia(constraints).then(successCallback);
}
function call(){
callButton.disabled = true;
hangupButton.disabled = false;
log("Starting Call");
if(navigator.mediaDevices.getUserMedia){
if(localStream.getVideoTracks().length > 0){
log("Using video device:" + localStream.getVideoTracks()[0].label);
}
if(localStream.getAudioTracks().length > 0){
log("Using audio device:" + localStream.getAudioTracks()[0].label);
}
}
localPeerConnection = new RTCPeerConnection(servers);
log("Created local peer connection object localPeerConnection");
localPeerConnection.onicecandidate = gotLocalIceCandidate;
localPeerConnection.onconnectionstatechange = function(event){
console.log("QWERTY");
}
remotePeerConnection = new RTCPeerConnection(servers);
log("Created remote peer connection object remotePeerConnection");
remotePeerConnection.onicecandidate = gotRemoteIceCandidate;
if (remotePeerConnection.addTrack !== undefined) {
remotePeerConnection.ontrack = ev => {
ev.streams.forEach(stream => doAddStream(stream));
}
} else {
remotePeerConnection.onaddstream = ev => {
doAddStream(ev.stream);
}
}
localStream.getTracks().forEach((track)=>{
localPeerConnection.addTrack(track, localStream);
});
// localPeerConnection.addStream(localStream);
log("Added localStream to localPeerConnection");
localPeerConnection.createOffer(gotLocalDescription, onSignalingError);
}
function gotLocalDescription(description){
localPeerConnection.setLocalDescription(description);
log("Offer from localPeerConnection: "+ description.sdp);
remotePeerConnection.setRemoteDescription(description);
remotePeerConnection.createAnswer(gotRemoteDescription, onSignalingError);
}
function gotRemoteDescription(description){
remotePeerConnection.setLocalDescription(description);
log("Answer from remotePeerConnection:"+description.sdp);
localPeerConnection.setRemoteDescription(description);
}
function hangup(){
log("Ending call");
localPeerConnection.close();
remotePeerConnection.close();
localPeerConnection = null;
remotePeerConnection = null;
hangupButton.disabled = true;
callButton.disabled = false;
}
function onSignalingError(error){
log("Failed to create signaling message: "+ error.name);
}
function gotLocalIceCandidate(event){
if(event.candidate){
remotePeerConnection.addIceCandidate(new RTCIceCandidate(event.candidate));
log("Local ICE candidate: \n" + event.candidate.candidate);
}
}
function gotRemoteIceCandidate(event){
if(event.candidate){
localPeerConnection.addIceCandidate(new RTCIceCandidate(event.candidate));
log("Remote ICE candidate: \n" +event.candidate.candidate);
}
}
function doAddStream(stream){
remoteVideo.srcObject = stream;
log("Received remote stream Do add Stream");
}
</script>
</body>
</html>
[1]: https://i.stack.imgur.com/gjNi4.png
I try to run poc which created in angular 8 using webrtc-adpater. Problem is that it is not working in firefox and safari, but working in chrome.
What I observe connection is never connecting or connected in firefox.
I import webrtc in code so anything else we need to handle for firefox and safari.
Basically from Wowza I trying to consume streaming.
Below code sample which working in chrome but not in Firefox and safari:
const ICE_SERVERS: RTCIceServer[] = [
{ urls: ['stun:stun.example.com', 'stun:stun-1.example.com'] },
{ urls: 'stun:stun.l.google.com:19302' }
];
setupSignalingServer(ele) {
const self = this;
window.RTCPeerConnection = window.RTCPeerConnection ||
window.mozRTCPeerConnection ||
window.webkitRTCPeerConnection;
let signalingConnection = new WebSocket('wss://local.geofabricdev.com/webrtc-session.json');
signalingConnection.binaryType = 'arraybuffer';
signalingConnection.onopen = function (res) {
console.log('connection open');
signalingConnection.onerror = self.errorHandler;
// window.RTCPeerConnection = self.getRTCPeerConnection();
const peerConnection = new RTCPeerConnection(PEER_CONNECTION_CONFIG);
peerConnection.addEventListener('connectionstatechange', event => {
console.log(event);
if (peerConnection.connectionState === 'connected') {
console.log('connected');
}
});
peerConnection.addEventListener('icecandidate', event => {
console.log('peerconnection state5:' + peerConnection.connectionState);
if (event.candidate) {
// signalingChannel.send({'new-ice-candidate': event.candidate});
}
});
peerConnection.addEventListener('track', async event => {
console.log('gotRemoteTrack: kind:' + event.track.kind + ' stream:' + event.streams[0]);
const video = self.createVideo();
// const remoteVideo1 = document.querySelector('#remoteVideo') as HTMLVideoElement;
try {
video.srcObject = event.streams[0];
// remoteVideo1.srcObject = event.streams[0];
} catch (error) {
video.src = window.URL.createObjectURL(event.streams[0]);
}
});
signalingConnection.addEventListener('message', async message => {
console.dir(message);
// self.getSignalMessageCallback(message);
console.log('wsConnection.onmessage: ' + message.data);
const signal = JSON.parse(message.data);
if(signal.status !== 200) {
const video = self.createVideo();
video.poster = '../assets/streaming-error.png';
}
const streamInfoResponse = signal['streamInfo'];
let g = [];
// self.socketStream.some((f) => {
g = self.streams.filter((str) => signal['streamInfo'].streamName === str.streamName);
// if(g.length > 0) {
// return true;
// }
// });
// });
if (streamInfoResponse !== undefined) {
g[0]['sessionId'] = streamInfoResponse.sessionId;
}
console.log('Received signal');
const msgCommand = signal['command'];
console.log(msgCommand);
if (signal.sdp) {
peerConnection.setRemoteDescription(new RTCSessionDescription(signal.sdp));
if (signal.sdp) {
const description = await peerConnection.createAnswer();
await peerConnection.setLocalDescription(description);
signalingConnection.send('{"direction":"play", "command":"sendResponse", "streamInfo":' +
JSON.stringify(g[0]) + ', "sdp":' + JSON.stringify(description) + ',"userData":' + JSON.stringify(self.userData) + '}');
}
// })
// .catch(self.errorHandler);
} else if (signal.iceCandidates) {
console.log('ice: ' + JSON.stringify(signal.iceCandidates));
peerConnection.addIceCandidate(new RTCIceCandidate(signal.iceCandidates[0])).catch(self.errorHandler);
}
if ('sendResponse'.localeCompare(msgCommand) === 0) {
if (signalingConnection != null) {
signalingConnection.close();
}
signalingConnection = null;
}
// };
});
signalingConnection.send('{"direction":"play", "command":"getOffer", "streamInfo":' +
JSON.stringify(ele) + ', "userData":' + JSON.stringify(self.userData) + '}');
};
signalingConnection.onclose = function (r) {
console.log('close');
};
}
private createVideo() {
const video = document.createElement('video');
video.classList.add('cameraPanel');
video.autoplay = true;
video.preload = 'true';
video.muted = true;
video.width = 300;
const remoteVideo = document.querySelector('#videoContainer') as HTMLElement;
remoteVideo.appendChild(video);
return video;
}
I've created a demo to show this, it seems that when I show a visualization the recorded audio gets really crunchy. Is there any way I can fix this? This works in chrome and Firefox, but safari is the only offender.
<style>
html, body {
margin: 0!important;
padding: 0!important;
overflow: hidden!important;
width: 100%;
}
</style>
<title>Audio Recording | RecordRTC</title>
<h1>Simple Audio Recording using RecordRTC</h1>
<meta name="viewport" content="width=device-width, initial-scale=1.0, minimum-scale=1.0">
<br>
<button id="btn-start-recording">Start Recording</button>
<button id="btn-stop-recording" disabled>Stop Recording</button>
<label><input id="visualizer" type="checkbox" checked/>Show Visualizer</label>
<hr>
<canvas id="audio-canvas"></canvas>
<div><audio controls autoplay></audio></div>
<script src="https://cdn.webrtc-experiment.com/RecordRTC.js"></script>
<script>
var audio = document.querySelector('audio');
function captureMicrophone(callback) {
if(microphone) {
callback(microphone);
return;
}
if(typeof navigator.mediaDevices === 'undefined' || !navigator.mediaDevices.getUserMedia) {
alert('This browser does not supports WebRTC getUserMedia API.');
if(!!navigator.getUserMedia) {
alert('This browser seems supporting deprecated getUserMedia API.');
}
}
navigator.mediaDevices.getUserMedia({
audio: {
echoCancellation: false
}
}).then(function(mic) {
callback(mic);
}).catch(function(error) {
alert('Unable to capture your microphone. Please check console logs.');
console.error(error);
});
}
function replaceAudio(src) {
var newAudio = document.createElement('audio');
newAudio.controls = true;
if(src) {
newAudio.src = src;
}
var parentNode = audio.parentNode;
parentNode.innerHTML = '';
parentNode.appendChild(newAudio);
audio = newAudio;
}
function stopRecordingCallback() {
replaceAudio(URL.createObjectURL(recorder.getBlob()));
btnStartRecording.disabled = false;
setTimeout(function() {
if(!audio.paused) return;
setTimeout(function() {
if(!audio.paused) return;
audio.play();
}, 1000);
audio.play();
}, 300);
audio.play();
if(microphone) {
microphone.stop();
microphone = null;
}
}
var recorder; // globally accessible
var microphone;
var btnStartRecording = document.getElementById('btn-start-recording');
var btnStopRecording = document.getElementById('btn-stop-recording');
var canvas = document.getElementById('audio-canvas');
var visualizerCheckbox = document.getElementById('visualizer');
btnStartRecording.onclick = function() {
this.disabled = true;
this.style.border = '';
this.style.fontSize = '';
if (!microphone) {
captureMicrophone(function(mic) {
microphone = mic;
replaceAudio();
audio.muted = true;
audio.play();
btnStartRecording.disabled = false;
btnStartRecording.style.border = '1px solid red';
btnStartRecording.style.fontSize = '150%';
alert('Please click startRecording button again. First time we tried to access your microphone. Now we will record it.');
});
return;
}
replaceAudio();
audio.muted = true;
audio.play();
if(visualizerCheckbox.checked){
initScope(canvas);
}
var options = {
type: 'audio',
numberOfAudioChannels: 2,
checkForInactiveTracks: true,
bufferSize: 16384,
sampleRate: 48000
};
if(recorder) {
recorder.destroy();
recorder = null;
}
recorder = RecordRTC(microphone, options);
recorder.startRecording();
btnStopRecording.disabled = false;
};
btnStopRecording.onclick = function() {
this.disabled = true;
recorder.stopRecording(stopRecordingCallback);
};
function initScope(audioScopeCanvas) {
if (audioScopeCanvas) {
var analyser;
var mic;
var javascriptNode;
var canvasContext;
audioContext = new AudioContext(); // NEW!!
analyser = audioContext.createAnalyser();
mic = audioContext.createMediaStreamSource(microphone);
javascriptNode = audioContext.createScriptProcessor(2048, 1, 1);
analyser.smoothingTimeConstant = 0.3;
analyser.fftSize = 1024;
mic.connect(analyser);
analyser.connect(javascriptNode);
javascriptNode.connect(audioContext.destination);
//canvasContext = $("#canvas")[0].getContext("2d");
canvasContext = audioScopeCanvas.getContext("2d");
javascriptNode.onaudioprocess = function () {
var array = new Uint8Array(analyser.frequencyBinCount);
analyser.getByteFrequencyData(array);
var values = 0;
var length = array.length;
for (var i = 0; i < length; i++) {
values += array[i];
}
var average = values / length;
canvasContext.clearRect(0, 0, 60, 130);
canvasContext.fillStyle = '#00ff00';
canvasContext.fillRect(0, 130 - average, 25, 130);
};
}
}
</script>
you can play with it here:
Http://andrew.colchagoff.com/safari-record.html
I'm using record rtc.
I am trying to record audio from the browser and stream it as raw audio (PCM) to my node server where I want to save to it in .mp3 format. I am using node-lame module on my server for creating an mp3 file from the PCM audio stream. The problem here is that the mp3 file is always high pitched and playing at a fast speed. I have tried both sending the data from the browser(client side) as int16 and as float32 and setting the appropriate lame.Encoder for them :
stream.pipe(new lame.Encoder({channels:2, bitDepth: 32, float:true,})) //float32
.pipe(fs.createWriteStream(path.resolve(__dirname, 'demo.mp3')))
stream.pipe(new lame.Encoder({channels:2, bitDepth: 16, sampleRate:44100,})) // int16
.pipe(fs.createWriteStream(path.resolve(__dirname, 'demo.mp3')))
Here is the code for the client side
(function(window) {
var client = new BinaryClient('ws://localhost:9001');
client.on('open', function() {
window.Stream = client.createStream();
if (!navigator.getUserMedia)
navigator.getUserMedia = navigator.getUserMedia || navigator.webkitGetUserMedia ||
navigator.mozGetUserMedia || navigator.msGetUserMedia;
if (navigator.getUserMedia) {
navigator.getUserMedia({audio:true}, success, function(e) {
alert('Error capturing audio.');
});
} else alert('getUserMedia not supported in this browser.');
var recording = false;
window.startRecording = function() {
recording = true;
}
window.stopRecording = function() {
recording = false;
window.Stream.end();
}
function success(e) {
audioContext = window.AudioContext || window.webkitAudioContext;
context = new audioContext();
// the sample rate is in context.sampleRate
audioInput = context.createMediaStreamSource(e);
var bufferSize = 2048;
recorder = context.createScriptProcessor(bufferSize, 1, 1);
recorder.onaudioprocess = function(e){
if(!recording) return;
console.log ('recording');
var left = e.inputBuffer.getChannelData(0);
window.Stream.write(left); //trying it with float32
}
audioInput.connect(recorder)
recorder.connect(context.destination);
}
function convertoFloat32ToInt16(buffer) {
var l = buffer.length;
var buf = new Int16Array(l)
while (l--) {
buf[l] = buffer[l]*0xFFFF; //convert to 16 bit
}
return buf.buffer
}
});
})(this);
Here is the node app
var express = require('express');
var BinaryServer = require('binaryjs').BinaryServer;
var fs = require('fs');
var lame = require("lame");
var path = require('path');
var KalmanFilter = require('kalmanjs').default;
var buffer = [];
var port = 3700;
var outFile = 'demo.mp3';
var app = express();
app.set('views', __dirname + '/tpl');
app.set('view engine', 'jade');
app.engine('jade', require('jade').__express);
app.use(express.static(__dirname + '/public'))
app.get('/', function(req, res){
res.render('index');
});
app.listen(port);
console.log('server open on port ' + port);
binaryServer = BinaryServer({port: 9001});
binaryServer.on('connection', function(client) {
console.log('new connection');
client.on('stream', function(stream, meta) {
console.log('new stream');
stream.pipe(new lame.Encoder({channels:2, bitDepth: 32, float:true}))
.pipe(fs.createWriteStream(path.resolve(__dirname, 'demo.mp3')))
.on('close',function () {
console.log('done?');
})
stream.on('end', function() {
console.log('wrote to file ' + outFile);
});
});
});
Both of these snippets are taken from https://github.com/gabrielpoca/browser-pcm-stream except for the mp3 part.
The default sampling rate in the browser is 48000hz, while you are encoding your mp3 files with a sampling rate of 44100hz. You either need to resample your PCM data or set the recording sampling rate to 44100hz.
<!DOCTYPE html>
<html>
<head>
<title>WebRTC Reference App</title>
<meta http-equiv="X-UA-Compatible" content="chrome=1"/>
<link rel="canonical" href="https://apprtc.appspot.com/?r=96443121"/>
<link rel="stylesheet" href="css/main.css" />
<script type="text/javascript" src="_ah/channel/jsapi.js"></script>
<!--<script type="text/javascript" src="/_ah/channel/jsapi"></script>-->
<script src="js/main.js"></script>
<!-- Load the polyfill to switch-hit between Chrome and Firefox -->
<script src="js/adapter.js"></script>
</head>
<body >
<script type="text/javascript">
var errorMessages = [];
var channelToken = 'AHRlWroGHj6YqWMdr7JeFULjqux9vQzrpCOnFM7zD91GLHc4PVW1WDZJI08ptgM_XMj5M22InerI5Lw5QN1rS3rHu2Dlnx3g95ILmzupe2OFB4OLCbZGbAQ';
var me = '49847744';
var roomKey = '96443121';
var roomLink = 'https://apprtc.appspot.com/?r=96443121';
var initiator = 0;
var pcConfig = {"iceServers": [{"url": "stun:stun.l.google.com:19302"}]};
var pcConstraints = {"optional": [{"DtlsSrtpKeyAgreement": true}]};
var offerConstraints = {"optional": [], "mandatory": {}};
var mediaConstraints = {"audio": true, "video": true};
var turnUrl = 'https://computeengineondemand.appspot.com/turn?username=49847744&key=4080218913';
var stereo = false;
var audio_send_codec = '';
var audio_receive_codec = 'opus/48000';
setTimeout(initialize, 1);
</script>`enter code here`
<div id="maindiv">
<div id="container" ondblclick="enterFullScreen()">
<div id="card">
<div id="local">
<video id="localVideo" autoplay muted="true"/>
</div>
<div id="remote">
<video id="remoteVideo" autoplay>
</video>
<div id="mini">
<video id="miniVideo" autoplay muted="true"/>
</div>
</div>
</div>
</div>
<div class="buttons">
<button id="startButton">Start</button>
<button id="callButton">Call</button>
<button id="hangupButton">Hang Up</button>
</div>
</div>
<script>
var localStream, localPeerConnection, remotePeerConnection;
var localVideo = document.getElementById("localVideo");
var remoteVideo = document.getElementById("remoteVideo");
var startButton = document.getElementById("startButton");
var callButton = document.getElementById("callButton");
var hangupButton = document.getElementById("hangupButton");
startButton.disabled = false;
callButton.disabled = true;
hangupButton.disabled = true;
startButton.onclick = start;
callButton.onclick = call;
hangupButton.onclick = hangup;
function trace(text) {
console.log((performance.now() / 1000).toFixed(3) + ": " + text);
}
function gotStream(stream){
trace("Received local stream");
localVideo.src = URL.createObjectURL(stream);
localStream = stream;
callButton.disabled = false;
}
function start() {
trace("Requesting local stream");
startButton.disabled = true;
navigator.getUserMedia = navigator.getUserMedia ||
navigator.webkitGetUserMedia || navigator.mozGetUserMedia;
navigator.getUserMedia({audio:true, video:true}, gotStream,
function(error) {
trace("navigator.getUserMedia error: ", error);
});
}
function call() {
callButton.disabled = true;
hangupButton.disabled = false;
trace("Starting call");
if (localStream.getVideoTracks().length > 0) {
trace('Using video device: ' + localStream.getVideoTracks()[0].label);
}
if (localStream.getAudioTracks().length > 0) {
trace('Using audio device: ' + localStream.getAudioTracks()[0].label);
}
var servers = null;
localPeerConnection = new webkitRTCPeerConnection(servers);
trace("Created local peer connection object localPeerConnection");
localPeerConnection.onicecandidate = gotLocalIceCandidate;
remotePeerConnection = new webkitRTCPeerConnection(servers);
trace("Created remote peer connection object remotePeerConnection");
remotePeerConnection.onicecandidate = gotRemoteIceCandidate;
remotePeerConnection.onaddstream = gotRemoteStream;
localPeerConnection.addStream(localStream);
trace("Added localStream to localPeerConnection");
localPeerConnection.createOffer(gotLocalDescription);
}
function gotLocalDescription(description){
localPeerConnection.setLocalDescription(description);
trace("Offer from localPeerConnection: \n" + description.sdp);
remotePeerConnection.setRemoteDescription(description);
remotePeerConnection.createAnswer(gotRemoteDescription);
}
function gotRemoteDescription(description){
remotePeerConnection.setLocalDescription(description);
trace("Answer from remotePeerConnection: \n" + description.sdp);
localPeerConnection.setRemoteDescription(description);
}
function hangup() {
trace("Ending call");
localPeerConnection.close();
remotePeerConnection.close();
localPeerConnection = null;
remotePeerConnection = null;
hangupButton.disabled = true;
callButton.disabled = false;
}
function gotRemoteStream(event){
remoteVideo.src = URL.createObjectURL(event.stream);
trace("Received remote stream");
}
function gotLocalIceCandidate(event){
if (event.candidate) {
remotePeerConnection.addIceCandidate(new RTCIceCandidate(event.candidate));
trace("Local ICE candidate: \n" + event.candidate.candidate);
}
}
function gotRemoteIceCandidate(event){
if (event.candidate) {
localPeerConnection.addIceCandidate(new RTCIceCandidate(event.candidate));
trace("Remote ICE candidate: \n " + event.candidate.candidate);
}
}
</script>
</body>
<footer id="status">`enter code here`
</footer>
<div id="infoDiv"></div>
</html>
here in the code above how to connect two different pc's and do a video call. I had taken it from webrtc.org.
I am getting two videos on the same PC. I want to connect two different PC's and do a video call. the example for this can be seen on vline.com where the link is autogenerated and on just needs to connect to the given link and they can do a video call.
please help
You will need a signaling server to relay all messages between the two peers. You could for example do this in Javascript using node.js:
var WebSocketServer = require('websocket').server;
var http = require('http');
var clients = [];
var server = http.createServer(function(request, response) {
// process HTTP request. Since we're writing just WebSockets server
// we don't have to implement anything.
});
server.listen(80, function() {
console.log((new Date()) + " Server is listening on port 80");
});
// create the server
wsServer = new WebSocketServer({
httpServer: server
});
function sendCallback(err) {
if (err) console.error("send() error: " + err);
}
// This callback function is called every time someone
// tries to connect to the WebSocket server
wsServer.on('request', function(request) {
console.log((new Date()) + ' Connection from origin ' + request.origin + '.');
var connection = request.accept(null, request.origin);
console.log(' Connection ' + connection.remoteAddress);
clients.push(connection);
// This is the most important callback for us, we'll handle
// all messages from users here.
connection.on('message', function(message) {
if (message.type === 'utf8') {
// process WebSocket message
console.log((new Date()) + ' Received Message ' + message.utf8Data);
// broadcast message to all connected clients
clients.forEach(function (outputConnection) {
if (outputConnection != connection) {
outputConnection.send(message.utf8Data, sendCallback);
}
});
}
});
connection.on('close', function(connection) {
// close user connection
console.log((new Date()) + " Peer disconnected.");
});
});
The peers need to send all signaling (offers, answers and candidates) via this signaling server (socket.send()). peer example below:
function connect() {
if (!started && localStream && channelReady) {
createPeerConnection();
started = true;
peerConn.createOffer(setLocalAndSendMessage, errorCallback, mediaConstraints);
} else {
alert("Local stream may not be running yet, or problem with signaling");
}
}
// send SDP via websocket connection
function setLocalAndSendMessage(sessionDescription) {
peerConn.setLocalDescription(sessionDescription);
socket.send(JSON.stringify(sessionDescription));
}
// Signaling messages end up here
function onMessage(evt) {
var descr = JSON.parse(evt.data);
if (descr.type === 'offer') {
if (!started) {
createPeerConnection();
started = true;
}
peerConn.setRemoteDescription(new RTCSessionDescription(descr));
peerConn.createAnswer(setLocalAndSendMessage, errorCallback, mediaConstraints);
} else if (descr.type === 'answer'){
peerConn.setRemoteDescription(new RTCSessionDescription(descr));
} else if (descr.type === 'candidate'){
var candidate = new RTCIceCandidate({sdpMLineIndex: descr.sdpMLineIndex, sdpMid: descr.sdpMid, candidate: descr.candidate});
peerConn.addIceCandidate(candidate);
}
}
//Creates a new RTCPeerConnection
function createPeerConnection() {
peerConn = new webkitRTCPeerConnection({"iceServers": []});
// send any ice candidates to the other peer
peerConn.onicecandidate = function(evt) {
if (event.candidate) {
socket.send(JSON.stringify({
type: "candidate",
sdpMLineIndex: evt.candidate.sdpMLineIndex,
sdpMid: evt.candidate.sdpMid,
candidate: evt.candidate.candidate}));
} else {
console.log("End of candidates.");
}
};
peerConn.addStream(localStream);
peerConn.addEventListener("addstream", onRemoteStreamAdded, false);
}
function onRemoteStreamAdded(event) {
//peerConn.removeEventListener("addstream", onRemoteStreamAdded, false);
remotevidArr.src = window.URL.createObjectURL(event.stream);
}