本意是将此内容成为一个社区Wiki帖子，以便开发人员能够获取简单而实用的示例，实现浏览器之间（p2p）JSON消息通信，同时保持最新状态。
WebRTC DataChannels是实验性技术，并且仍处于草案阶段。目前看来，网络上存在大量过时的WebRTC示例，尤其是对于试图学习RTCDataChannel API的开发者而言，情况更加复杂。
简单而功能齐全的1页示例，可在支持WebRTC的兼容浏览器上运行，似乎很难找到。例如，一些示例省略了信令实现，另一些仅适用于单个浏览器（如Chrome-Chrome），许多由于API最近的更改已过时，其他则过于复杂，从而创建了一个启动障碍。这里有一个使用HTML5 WebSockets进行信令和node.js后端的工作示例
信令技术：WebSockets
客户端：纯html/javascript
服务器：node.js，ws
最近测试于：Firefox 40.0.2，Chrome 44.0.2403.157 m，Opera 31.0.1889.174
<html>
<head>
</head>
<body>
<p id='msg'>Click the following in different browser windows</p>
<button type='button' onclick='init(false)'>I AM Answerer Peer (click first)</button>
<button type='button' onclick='init(true)'>I AM Offerer Peer</button>
<script>
(function() {
var offererId = 'Gandalf', // note: client id conflicts can happen
answererId = 'Saruman', // no websocket cleanup code exists
ourId, peerId,
RTC_IS_MOZILLA = !!window.mozRTCPeerConnection,
RTCPeerConnection = window.RTCPeerConnection || window.webkitRTCPeerConnection || window.mozRTCPeerConnection || window.msRTCPeerConnection,
RTCSessionDescription = window.RTCSessionDescription || window.mozRTCSessionDescription || window.msRTCSessionDescription,
RTCIceCandidate = window.RTCIceCandidate || window.mozRTCIceCandidate || window.msRTCIceCandidate,
rtcpeerconn = new RTCPeerConnection(
{iceServers: [{ 'url': 'stun:stun.services.mozilla.com'}, {'url': 'stun:stun.l.google.com:19302'}]},
{optional: [{RtpDataChannels: false}]}
),
rtcdatachannel,
websocket = new WebSocket('ws://' + window.location.hostname + ':8000'),
comready, onerror;
window.init = function(weAreOfferer) {
ourId = weAreOfferer ? offererId : answererId;
peerId = weAreOfferer ? answererId : offererId;
websocket.send(JSON.stringify({
inst: 'init',
id: ourId
}));
if(weAreOfferer) {
rtcdatachannel = rtcpeerconn.createDataChannel(offererId+answererId);
rtcdatachannel.onopen = comready;
rtcdatachannel.onerror = onerror;
rtcpeerconn.createOffer(function(offer) {
rtcpeerconn.setLocalDescription(offer, function() {
var output = offer.toJSON();
if(typeof output === 'string') output = JSON.parse(output); // normalize: RTCSessionDescription.toJSON returns a json str in FF, but json obj in Chrome
websocket.send(JSON.stringify({
inst: 'send',
peerId: peerId,
message: output
}));
}, onerror);
}, onerror);
}
};
rtcpeerconn.ondatachannel = function(event) {
rtcdatachannel = event.channel;
rtcdatachannel.onopen = comready;
rtcdatachannel.onerror = onerror;
};
websocket.onmessage = function(input) {
var message = JSON.parse(input.data);
if(message.type && message.type === 'offer') {
var offer = new RTCSessionDescription(message);
rtcpeerconn.setRemoteDescription(offer, function() {
rtcpeerconn.createAnswer(function(answer) {
rtcpeerconn.setLocalDescription(answer, function() {
var output = answer.toJSON();
if(typeof output === 'string') output = JSON.parse(output); // normalize: RTCSessionDescription.toJSON returns a json str in FF, but json obj in Chrome
websocket.send(JSON.stringify({
inst: 'send',
peerId: peerId,
message: output
}));
}, onerror);
}, onerror);
}, onerror);
} else if(message.type && message.type === 'answer') {
var answer = new RTCSessionDescription(message);
rtcpeerconn.setRemoteDescription(answer, function() {/* handler required but we have nothing to do */}, onerror);
} else if(rtcpeerconn.remoteDescription) {
// ignore ice candidates until remote description is set
rtcpeerconn.addIceCandidate(new RTCIceCandidate(message.candidate));
}
};
rtcpeerconn.onicecandidate = function (event) {
if (!event || !event.candidate) return;
websocket.send(JSON.stringify({
inst: 'send',
peerId: peerId,
message: {candidate: event.candidate}
}));
};
/** called when RTC signaling is complete and RTCDataChannel is ready */
comready = function() {
rtcdatachannel.send('hello world!');
rtcdatachannel.onmessage = function(event) {
document.getElementById('msg').innerHTML = 'RTCDataChannel peer ' + peerId + ' says: ' + event.data;
}
};
/** global error function */
onerror = websocket.onerror = function(e) {
console.log('====== WEBRTC ERROR ======', arguments);
document.getElementById('msg').innerHTML = '====== WEBRTC ERROR ======<br>' + e;
throw new Error(e);
};
})();
</script>
</body>
</html>服务器端代码：
var server = require('http').createServer(),
express = require('express'),
app = express(),
WebSocketServer = require('ws').Server,
wss = new WebSocketServer({ server: server, port: 8000 });
app.use(express.static(__dirname + '/static')); // client code goes in static directory
var clientMap = {};
wss.on('connection', function (ws) {
ws.on('message', function (inputStr) {
var input = JSON.parse(inputStr);
if(input.inst == 'init') {
clientMap[input.id] = ws;
} else if(input.inst == 'send') {
clientMap[input.peerId].send(JSON.stringify(input.message));
}
});
});
server.on('request', app);
server.listen(80, YOUR_HOSTNAME_OR_IP_HERE, function () { console.log('Listening on ' + server.address().port) });