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网页版WebRTC多人聊天Demo

时间:2015-07-01 20:36:00      阅读:198      评论:0      收藏:0      [点我收藏+]

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网页版WebRTC多人聊天Demo
本文基于Codelabstep7,在其基础上作简单修改,使其支持多人视频通讯,本文暂时只支持星状结构三人聊天,多人聊天可以在基础上扩展,原理相同。
.源码分析
该工程包括三个文件:server.jsmain.jsindex.html
1.server.js
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);
后台服务代码,负责异步消息通讯。当有新用户加入房间时,向客户端发送消息,客户端接收到消息后作相应的处理。
2.index.html
网站主页,包括两块视频区域和文本区域。

<!DOCTYPE html>
<html>
<head>

<meta name=‘keywords‘ content=‘WebRTC, HTML5, JavaScript‘ />
<meta name=‘description‘ content=‘WebRTC Reference App‘ />
<meta name=‘viewport‘ content=‘width=device-width,initial-scale=1,minimum-scale=1,maximum-scale=1‘>

<base target=‘_blank‘>

<title>WebRTC client</title>

<link rel=‘stylesheet‘ href=‘css/main.css‘ /> 

</head>

<body>

<div id=‘container‘ class=‘main‘ >

  <div id=‘videos‘ class=‘videos‘>
    <video id=‘localVideo‘ class=‘localVideo‘ autoplay muted></video>
    <video id=‘remoteVideo‘ class=‘remoteVideo‘ autoplay></video>
  </div>
 
  <div id=‘textareas‘>
        <textarea id="dataChannelSend" disabled placeholder="Press Start, enter some text, then press Send."></textarea>
        <textarea id="dataChannelReceive" disabled></textarea>
    </div>

  <button id="sendButton" disabled>Send</button>

</div>

<script src=‘/socket.io/socket.io.js‘></script>
<script src=‘js/lib/adapter.js‘></script>
<script src=‘js/main.js‘></script>

</body>
</html>
3.main.js
核心代码区域,包括房间的创建,RTCPeerConnection创建和两点间的视频通话。
3.1消息处理
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(‘message‘, function (message){
  console.log(‘Received message:‘, message);
  if (message === ‘got user media‘) {
      maybeStart();
  } else if (message.type === ‘offer‘) {
    if (!isInitiator && !isStarted) {
      maybeStart();
    }
    pc.setRemoteDescription(new RTCSessionDescription(message));
    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();
  }
});

3.2peerconnection创建和通讯
function createPeerConnection() {
  try {
    pc = new RTCPeerConnection(pc_config, pc_constraints);
    pc.onicecandidate = handleIceCandidate;
    console.log(‘Created RTCPeerConnnection with:\n‘ +
      ‘  config: \‘‘ + JSON.stringify(pc_config) + ‘\‘;\n‘ +
      ‘  constraints: \‘‘ + JSON.stringify(pc_constraints) + ‘\‘.‘);
  } catch (e) {
    console.log(‘Failed to create PeerConnection, exception: ‘ + e.message);
    alert(‘Cannot create RTCPeerConnection object.‘);
      return;
  }
  pc.onaddstream = handleRemoteStreamAdded;
  pc.onremovestream = handleRemoteStreamRemoved;

  if (isInitiator) {
    try {
      // Reliable Data Channels not yet supported in Chrome
      sendChannel = pc.createDataChannel("sendDataChannel",
        {reliable: false});
      sendChannel.onmessage = handleMessage;
      trace(‘Created send data channel‘);
    } catch (e) {
      alert(‘Failed to create data channel. ‘ +
            ‘You need Chrome M25 or later with RtpDataChannel enabled‘);
      trace(‘createDataChannel() failed with exception: ‘ + e.message);
    }
    sendChannel.onopen = handleSendChannelStateChange;
    sendChannel.onclose = handleSendChannelStateChange;
    console.log(‘....................this is  a initiator = true....................‘);
  } else {
    pc.ondatachannel = gotReceiveChannel;
    console.log(‘....................this is not a initiator = false....................‘);
  }
}

3.3 视频源的输出展现

function handleRemoteStreamAdded(event) {
  console.log(‘Remote stream added.‘);
 // reattachMediaStream(miniVideo, localVideo);
  attachMediaStream(remoteVideo, event.stream);
  remoteStream = event.stream;
//  waitForRemoteVideo();
}

.
简单工作流程介绍与修改思路

1. 工作过程如下:

1.1.浏览器A访问主页,允许访问摄像头音频设备,server接收到‘create or join‘消息,计算此时连接到服务器的客户端数量,此时数量为0,则向客户端发送‘created‘消息。
1.2.浏览器A接收到‘created‘消息,将isInitiator设为true,该值为true表示该客户断是peerconnection的发起者。
1.3.浏览器B访问主页,允许访问摄像头音频设备,server接收到‘create or join‘消息,计算此时连接到服务器的客户端数量,此时数量为1,则向客户端发送joinjoined消息。
1.4.浏览器A和浏览器B都接收到joinjoined消息,设置isChannelReady=true,表示此时准备好建立连接。浏览器A发起peerconnection连接doCall,浏览器B回应peerconnection连接doAnswer,AB建立P2P连接。
1.5.AB分别将来自本地和远端的视频stream显示在页面上。
注意:浏览器A和浏览器B都接受来自server相同的消息,而两者在接收到相同的消息后的处理却不一样(main.js代码是一样的),一个是发起者,一个是应答者。可以使用状态机来理解,程序所处状态不一样,虽然接收到相同的命令,但可以做出不同的处理(通过isInitiator变量区分不同的状态)。

2.三人聊天室的实现
简单起见,我们暂时先实现三人视频通讯,使用星状结构。下面是修改思路:
a.AB以及建立连接,此时如C加入,可以将AC建立连接,同时保持AB之前的连接。此时,A能看到BC,而BC只能看到A
b.如果A B C三者需要互相看到,则需要AB的视频传给C,并将C的视频传给B

本文暂时只实现AB通讯,AC通讯,BC之间不能通讯。下面是具体的代码修改步骤:
2.1server.js

if (numClients == 0){
            socket.join(room);
            socket.emit(‘created‘, room);
        } else if (numClients <=2 ) { //第三个用户加入后仍然发送join joined消息
            io.sockets.in(room).emit(‘join‘, room);
            socket.join(room);
            socket.emit(‘joined‘, room);
        } else { // max two clients
            socket.emit(‘full‘, room);
        }

2.2index.html
可以采用动态方式添加,这里简单起见直接增加一路视频实现块。
<div id=‘videos‘ class=‘videos‘>
    <video id=‘localVideo‘ class=‘localVideo‘ autoplay muted></video> //本地视频 A
  </div>
  <div >
    <video id=‘remoteVideo‘ class=‘remoteVideo‘ autoplay></video>// remote视频B
  </div>
  <div >
    <video id=‘remoteVideo2‘ class=‘remoteVideo2‘ autoplay></video> //remote视频c
  </div>

2.3 main.js
   a.增加一个全局变量isPeerEstablished
用来表示该客户端是否已经创建了PeerConnectionisPeerEstablishedisInitiator两者可以区分发起者和应答者,因为具有超过2个客户端,所以必须使用isPeerEstablished来选择尚未创建连接的客户端作为应答者。
     var isPeerEstablished=false;
   b.处理message机制修改
在判断条件里面加入(!isPeerEstablished||isInitiator)表示尚未创建链接C和发起者A才会执行peerconnection。保证新加入者CA创建链接,同时保持AB的连接。
socket.on(‘message‘, function (message){
  console.log(‘Received message:‘, message);
  if (message === ‘got user media‘&&(!isPeerEstablished||isInitiator)) {
    maybeStart();
  } else if (message.type === ‘offer‘&&(!isPeerEstablished||isInitiator)) {    
    if (!isInitiator && !isStarted) {     
      maybeStart();
    }
    pc.setRemoteDescription(new RTCSessionDescription(message));
    doAnswer();
  } else if (message.type === ‘answer‘ && isStarted&&(!isPeerEstablished||isInitiator)) {
    pc.setRemoteDescription(new RTCSessionDescription(message));
  } else if (message.type === ‘candidate‘ && isStarted&&(!isPeerEstablished||isInitiator)) {
    var candidate = new RTCIceCandidate({sdpMLineIndex:message.label,
      candidate:message.candidate});
    pc.addIceCandidate(candidate);
  } else if (message === ‘bye‘ && isStarted) {      
    handleRemoteHangup();
  }
});

c.视频流展现

如果isInitiatorisPeerEstablished都为true,说明此时AB已经建立链接。此时,应该将新的视频流显示在remoteVideo2中。其他情况将视频流展示在remoteVideo中。
function handleRemoteStreamAdded(event) {
  console.log(‘Remote stream added.‘);
  
 // reattachMediaStream(miniVideo, localVideo);
if(isInitiator&&isPeerEstablished){
  attachMediaStream(remoteVideo2, event.stream);
  remoteStream2 = event.stream;
}else{
  attachMediaStream(remoteVideo, event.stream);
  remoteStream = event.stream;
}
isPeerEstablished=true;
//  waitForRemoteVideo();
}

d.其他两处修改

var remoteVideo2 = document.querySelector(‘#remoteVideo2‘);

......


function handleRemoteHangup() {
  console.log(‘Session terminated.‘);
  stop();
  //isInitiator = false;  //总是保持A的发起者角色


}
三人聊天效果图:

技术分享

网页版WebRTC多人聊天Demo

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原文地址:http://www.cnblogs.com/cther/p/4614391.html

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