所以我已经建立了一个Web RTC连接，几乎完全理解所涉及的步骤和机制，并在网上使用完整的SSL证书进行测试（所以我可以在chrome上使用它）但它似乎永远不会使用不在同一无线连接/路由器下的计算机。
在我的测试中我记录了生成的ICE候选者并且两个浏览器正在通信但我没有视频输入，除非两个尝试通信的连接处于同一连接之外。
浏览器端设置webRTC：
var socket = io.connect();
var isChannelReady;
var isInitiator = false;
var isStarted = false;
var localStream;
var pc;
var remoteStream;
var turnReady;
var pc_config = {'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 sdpConstraints = {
optional: [],
'mandatory': {
'OfferToReceiveAudio': true,
'OfferToReceiveVideo': true
}
};
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), function(){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.getElementById('localVideo');
var remoteVideo = document.getElementById('remoteVideo');
function handleUserMedia(stream) {
console.log('Adding local stream.');
console.log(localVideo);
localVideo.src = window.URL.createObjectURL(stream);
localStream = stream;
sendMessage('got user media');
if (isInitiator) {
maybeStart();
}
}
function handleUserMediaError(error){
console.log('getUserMedia error: ', error);
}
// window.onload = function () {
// var localVideo = document.getElementById('localVideo');
// var remoteVideo = document.getElementById('mehh');
// }
var constraints = {video: true};
navigator.getUserMedia(constraints, handleUserMedia, handleUserMediaError);
console.log('Getting user media with constraints', constraints);
if (location.hostname != "localhost") {
requestTurn('https://computeengineondemand.appspot.com/turn?username=41784574&key=4080218913');
}
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, function(err){if(err) throw err;}, 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 requestTurn(turn_url) {
var turnExists = false;
for (var i in pc_config.iceServers) {
if (pc_config.iceServers[i].url.substr(0, 5) === 'turn:') {
turnExists = true;
turnReady = true;
break;
}
}
if (!turnExists) {
console.log('Getting TURN server from ', turn_url);
// 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);
pc_config.iceServers.push({
'url': 'turn:' + turnServer.username + '@' + turnServer.turn,
'credential': turnServer.password
});
turnReady = true;
}
};
xhr.open('GET', turn_url, 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 || mLineIndex === undefined) {
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) {
console.log(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;
}
/////////////////////////////////////////////
room = prompt("Enter room name:");
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);
});
////////////////////////////////////////////////
以下是服务器端与webRTC相关的所有内容（使用socket io）：
server.listen(port);
server.on('error', onError);
server.on('listening', onListening);
var io = require('socket.io').listen(server);
io.sockets.on('connection', function (socket){
console.log('connected');
// convenience function to log server messages on the client
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, would be room only (not broadcast)
socket.broadcast.emit('message', message);
});
socket.on('create or join', function (room) {
//io.sockets.clients(room)
//io.of('/').in(room).clients
//io.sockets.adapter.rooms[room]
var numClients;
var ish = io.sockets.adapter.rooms[room];
(ish === undefined) ? numClients = 0 : numClients = ish.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);
}
socket.emit('emit(): client ' + socket.id + ' joined room ' + room);
socket.broadcast.emit('broadcast(): client ' + socket.id + ' joined room ' + room);
});
});
我根本找不到阻止流发送的任何错误......
非常愿意进行私人聊天，因此我可以继续我的项目。