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webrtc学习笔记一 (视频流)

阅读更多
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
<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
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