webrtc学习笔记一 （视频流）
程序员文章站
2022-07-08 16:46:03
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google官方的
socket.io的源码
https://bitbucket.org/webrtc/codelab/downloads
http://dl.iteye.com/topics/download/88405497-3fd1-3e34-adba-004583638559
最简单的WebRTC示例
http://www.blogjava.net/linli/archive/2014/10/21/418910.html
webrtc.html
nodejs的server
server.js
在局域网可以忽略google的stun
一台电脑先打开
192.168.137.27:3000
另一台打开
192.168.137.27:3000#true
iceServer设置成空也行
官方参考
http://www.html5rocks.com/en/tutorials/webrtc/basics/
http://www.html5rocks.com/en/tutorials/getusermedia/intro/
http://blog.csdn.net/liaowenfeng/article/details/18407837
例子
https://apprtc.appspot.com/
例子
http://www.simpl.info/rtcpeerconnection/
git：
https://github.com/andyet/SimpleWebRTC
https://github.com/webRTC-io/webRTC.io
socket.io的源码
https://bitbucket.org/webrtc/codelab/downloads
http://dl.iteye.com/topics/download/88405497-3fd1-3e34-adba-004583638559
最简单的WebRTC示例
http://www.blogjava.net/linli/archive/2014/10/21/418910.html
webrtc.html
<html>
<body>
Local: <br>
<video id="localVideo" autoplay></video><br>
Remote: <br>
<video id="remoteVideo" autoplay></video>
<script>
// 仅仅用于控制哪一端的浏览器发起offer，#号后面有值的一方发起
var isCaller = window.location.href.split('#')[1];
// 与信令服务器的WebSocket连接
var socket = new WebSocket("ws://192.168.137.27:3000");
// stun和turn服务器
var iceServer = {
"iceServers": [{
"url": "stun:stun.l.google.com:19302"
}, {
"url": "turn:numb.viagenie.ca",
"username": "haoningabc@163.com",
"credential": "12345"
}]
};
// 创建PeerConnection实例 (参数为null则没有iceserver，即使没有stunserver和turnserver，仍可在局域网下通讯)
var pc = new webkitRTCPeerConnection(iceServer);
// 发送ICE候选到其他客户端
pc.onicecandidate = function(event){
if (event.candidate !== null) {
socket.send(JSON.stringify({
"event": "_ice_candidate",
"data": {
"candidate": event.candidate
}
}));
}
};
// 如果检测到媒体流连接到本地，将其绑定到一个video标签上输出
pc.onaddstream = function(event){
//document.getElementById('remoteVideo').src = URL.createObjectURL(event.stream);
document.getElementById('remoteVideo').srcObject = event.stream;
};
// 发送offer和answer的函数，发送本地session描述
var sendOfferFn = function(desc){
pc.setLocalDescription(desc);
socket.send(JSON.stringify({
"event": "_offer",
"data": {
"sdp": desc
}
}));
},
sendAnswerFn = function(desc){
pc.setLocalDescription(desc);
socket.send(JSON.stringify({
"event": "_answer",
"data": {
"sdp": desc
}
}));
};
// 获取本地音频和视频流
navigator.webkitGetUserMedia({
"audio": true,
"video": true
}, function(stream){
//绑定本地媒体流到video标签用于输出
//document.getElementById('localVideo').src = URL.createObjectURL(stream);
document.getElementById('localVideo').srcObject = stream;
//向PeerConnection中加入需要发送的流
pc.addStream(stream);
//如果是发起方则发送一个offer信令
if(isCaller){
pc.createOffer(sendOfferFn, function (error) {
console.log('Failure callback: ' + error);
});
}
}, function(error){
//处理媒体流创建失败错误
console.log('getUserMedia error: ' + error);
});
//处理到来的信令
socket.onmessage = function(event){
var json = JSON.parse(event.data);
console.log('onmessage: ', json);
//如果是一个ICE的候选，则将其加入到PeerConnection中，否则设定对方的session描述为传递过来的描述
if( json.event === "_ice_candidate" ){
pc.addIceCandidate(new RTCIceCandidate(json.data.candidate));
} else {
pc.setRemoteDescription(new RTCSessionDescription(json.data.sdp));
// 如果是一个offer，那么需要回复一个answer
if(json.event === "_offer") {
pc.createAnswer(sendAnswerFn, function (error) {
console.log('Failure callback: ' + error);
});
}
}
};
</script>
</body>
</html>
nodejs的server
server.js
//http://www.blogjava.net/linli/archive/2014/10/21/418910.html
var express = require('express'),
app = express(),
server = require('http').createServer(app);
server.listen(3000);
app.get('/', function(req, res) {
res.sendfile(__dirname + '/webrtc.html');
});
var WebSocketServer = require('ws').Server,
wss = new WebSocketServer({server: server});
// 存储socket的数组，这里只能有2个socket，每次测试需要重启，否则会出错
var wsc = [],
index = 1;
// 有socket连入
wss.on('connection', function(ws) {
console.log('connection');
// 将socket存入数组
wsc.push(ws);
// 记下对方socket在数组中的下标，因为这个测试程序只允许2个socket
// 所以第一个连入的socket存入0，第二个连入的就是存入1
// otherIndex就反着来，第一个socket的otherIndex下标为1，第二个socket的otherIndex下标为0
var otherIndex = index--,
desc = null;
if (otherIndex == 1) {
desc = 'first socket';
} else {
desc = 'second socket';
}
// 转发收到的消息
ws.on('message', function(message) {
var json = JSON.parse(message);
console.log('received (' + desc + '): ', json);
console.log('otherIndex ---(' + otherIndex + '): ');
wsc[otherIndex].send(message, function (error) {
if (error) {
console.log('Send message error (' + desc + '): ', error);
}
});
});
});
在局域网可以忽略google的stun
一台电脑先打开
192.168.137.27:3000
另一台打开
192.168.137.27:3000#true
iceServer设置成空也行
官方参考
http://www.html5rocks.com/en/tutorials/webrtc/basics/
http://www.html5rocks.com/en/tutorials/getusermedia/intro/
http://blog.csdn.net/liaowenfeng/article/details/18407837
例子
https://apprtc.appspot.com/
例子
http://www.simpl.info/rtcpeerconnection/
git：
https://github.com/andyet/SimpleWebRTC
https://github.com/webRTC-io/webRTC.io