基于Chrome、Java、WebSocket、WebRTC实现浏览器视频通话

http://blog.csdn.net/leecho571/article/details/8146525

 

基于Chrome、Java、WebSocket、WebRTC实现浏览器视频通话

分类： Java HTML5 WebSocket WebRTC 2012-11-04 13:04  6598人阅读  评论(14)  收藏  举报

目录(?)[+]

介绍

        最近这段时间折腾了一下 WebRTC，看了网上的 https://apprtc.appspot.com/的例子（可能需要访问），这个例子是部署在Google App Engine上的应用程序，依赖GAE的环境，后台的语言是python，而且还依赖Google App Engine Channel API，所以无法在本地运行，也无法扩展。费了一番功夫研读了例子的python端的源代码，决定用Java实现，Tomcat7之后开始支持WebSocket，打算用WebSocket代替Google App Engine Channel API实现前后台的通讯，在整个例子中Java+WebSocket起到的作用是负责客户端之间的通信，并不负责视频的传输，视频的传输依赖于WebRTC。

实例的特点是：

1. HTML5
2. 不需要任何插件
3. 资源占用不是很大，对服务器的开销比较小，只要客户端建立连接，视频传输完全有浏览器完成
4. 通过JS实现，理论上只要浏览器支持WebSocket，WebRTC就能运行（目前只在Chrome测试通过，Chrome版本24.0.1312.2 dev-m）

实现

对于前端JS代码及用到的对象大家可以访问 http://www.html5rocks.com/en/tutorials/webrtc/basics/查看详细的代码介绍。我在这里只介绍下我改动过的地方，首先建立一个客户端实时获取状态的连接，在GAE的例子上是通过GAE Channel API实现，我在这里用WebSocket实现，代码：

 function openChannel() {
     console.log("Opening channel.");
     socket = new WebSocket(
             "ws://192.168.1.102:8080/RTCApp/websocket?u=${user}");
     socket.onopen = onChannelOpened;
     socket.onmessage = onChannelMessage;
     socket.onclose = onChannelClosed;
 }

建立一个WebSocket连接，并注册相关的事件。这里通过Java实现WebSocket连接：

 package org.rtc.servlet;

 import java.io.IOException;

 import javax.servlet.ServletException;
 import javax.servlet.annotation.WebServlet;
 import javax.servlet.http.HttpServletRequest;
 import javax.servlet.http.HttpServletResponse;

 import org.apache.catalina.websocket.StreamInbound;
 import org.apache.catalina.websocket.WebSocketServlet;
 import org.rtc.websocket.WebRTCMessageInbound;

 @WebServlet(urlPatterns = { "/websocket"})
 public class WebRTCWebSocketServlet extends WebSocketServlet {

     private static final long serialVersionUID = 1L;

     private String user;

     public void doGet(HttpServletRequest request, HttpServletResponse response)
             throws ServletException, IOException {
         this.user = request.getParameter("u");
         super.doGet(request, response);
     }

     @Override
     protected StreamInbound createWebSocketInbound(String subProtocol) {
         return new WebRTCMessageInbound(user);
     }
 }

如果你想实现WebSocket必须得用Tomcat7及以上版本，并且引入：catalina.jar，tomcat-coyote.jar两个JAR包，部署到Tomcat7之后得要去webapps/应用下面去删除这两个AR包否则无法启动，WebSocket访问和普通的访问最大的不同在于继承了WebSocketServlet，关于WebSocket的详细介绍大家可以访问 http://redstarofsleep.iteye.com/blog/1488639，在这里就不再赘述。大家可以看看WebRTCMessageInbound这个类的实现：

 package org.rtc.websocket;

 import java.io.IOException;
 import java.nio.ByteBuffer;
 import java.nio.CharBuffer;

 import org.apache.catalina.websocket.MessageInbound;
 import org.apache.catalina.websocket.WsOutbound;

 public class WebRTCMessageInbound extends MessageInbound {

     private final String user;

     public WebRTCMessageInbound(String user) {
         this.user = user;
     }

     public String getUser(){
         return this.user;
     }

     @Override
     protected void onOpen(WsOutbound outbound) {
         //触发连接事件，在连接池中添加连接
         WebRTCMessageInboundPool.addMessageInbound(this);
     }

     @Override
     protected void onClose(int status) {
         //触发关闭事件，在连接池中移除连接
         WebRTCMessageInboundPool.removeMessageInbound(this);
     }

     @Override
     protected void onBinaryMessage(ByteBuffer message) throws IOException {
         throw new UnsupportedOperationException(
                 "Binary message not supported.");
     }

     @Override
     protected void onTextMessage(CharBuffer message) throws IOException {

     }
 }

WebRTCMessageInbound继承了MessageInbound，并绑定了两个事件，关键的在于连接事件，将连接存放在连接池中，等客户端A发起发送信息的时候将客户端B的连接取出来发送数据，看看WebRTCMessageInboundPool这个类：

 package org.rtc.websocket;

 import java.io.IOException;
 import java.nio.CharBuffer;
 import java.util.HashMap;
 import java.util.Map;

 public class WebRTCMessageInboundPool {

     private static final Map connections = new HashMap();

     public static void addMessageInbound(WebRTCMessageInbound inbound){
         //添加连接
         System.out.println("user : " + inbound.getUser() + " join..");
         connections.put(inbound.getUser(), inbound);
     }

     public static void removeMessageInbound(WebRTCMessageInbound inbound){
         //移除连接
         connections.remove(inbound.getUser());
     }

     public static void sendMessage(String user,String message){
         try {
             //向特定的用户发送数据
             System.out.println("send message to user : " + user + " message content : " + message);
             WebRTCMessageInbound inbound = connections.get(user);
             if(inbound != null){
                 inbound.getWsOutbound().writeTextMessage(CharBuffer.wrap(message));
             }
         } catch (IOException e) {
             e.printStackTrace();
         }
     }
 }

WebRTCMessageInboundPool这个类中最重要的是sendMessage方法，向特定的用户发送数据。

大家可以看看这段代码：

 function openChannel() {
     console.log("Opening channel.");
     socket = new WebSocket(
             "ws://192.168.1.102:8080/RTCApp/websocket?u=${user}");
     socket.onopen = onChannelOpened;
     socket.onmessage = onChannelMessage;
     socket.onclose = onChannelClosed;
 }

${user}是怎么来的呢？其实在进入这个页面之前是有段处理的：

 package org.rtc.servlet;

 import java.io.IOException;
 import java.util.UUID;

 import javax.servlet.ServletException;
 import javax.servlet.annotation.WebServlet;
 import javax.servlet.http.HttpServlet;
 import javax.servlet.http.HttpServletRequest;
 import javax.servlet.http.HttpServletResponse;

 import org.apache.commons.lang.StringUtils;
 import org.rtc.room.WebRTCRoomManager;

 @WebServlet(urlPatterns = {"/room"})
 public class WebRTCRoomServlet extends HttpServlet {

     private static final long serialVersionUID = 1L;

     public void doGet(HttpServletRequest request, HttpServletResponse response)
             throws ServletException, IOException {
         this.doPost(request, response);
     }

     public void doPost(HttpServletRequest request, HttpServletResponse response)
             throws ServletException, IOException {
         String r = request.getParameter("r");
         if(StringUtils.isEmpty(r)){
             //如果房间为空，则生成一个新的房间号
             r = String.valueOf(System.currentTimeMillis());
             response.sendRedirect("room?r=" + r);
         }else{
             Integer initiator = 1;
             String user = UUID.randomUUID().toString().replace("-", "");//生成一个用户ID串
             if(!WebRTCRoomManager.haveUser(r)){//第一次进入可能是没有人的，所以就要等待连接，如果有人进入了带这个房间好的页面就会发起视频通话的连接
                 initiator = 0;//如果房间没有人则不发送连接的请求
             }
             WebRTCRoomManager.addUser(r, user);//向房间中添加一个用户
             String basePath = request.getScheme()+"://"+request.getServerName()+":"+request.getServerPort() +  request.getContextPath() +"/";
             String roomLink = basePath + "room?r=" + r;
             String roomKey = r;//设置一些变量
             request.setAttribute("initiator", initiator);
             request.setAttribute("roomLink", roomLink);
             request.setAttribute("roomKey", roomKey);
             request.setAttribute("user", user);
             request.getRequestDispatcher("index.jsp").forward(request, response);
         }
     }
 }

这个是进入房间前的处理，然而客户端是怎么发起视频通话的呢？

 function initialize() {
             console.log("Initializing; room=${roomKey}.");
             card = document.getElementById("card");
             localVideo = document.getElementById("localVideo");
             miniVideo = document.getElementById("miniVideo");
             remoteVideo = document.getElementById("remoteVideo");
             resetStatus();
             openChannel();
             getUserMedia();
         }

         function getUserMedia() {
             try {
                 navigator.webkitGetUserMedia({
                     'audio' : true,
                     'video' : true
                 }, onUserMediaSuccess, onUserMediaError);
                 console.log("Requested access to local media with new syntax.");
             } catch (e) {
                 try {
                     navigator.webkitGetUserMedia("video,audio",
                             onUserMediaSuccess, onUserMediaError);
                     console
                             .log("Requested access to local media with old syntax.");
                 } catch (e) {
                     alert("webkitGetUserMedia() failed. Is the MediaStream flag enabled in about:flags?");
                     console.log("webkitGetUserMedia failed with exception: "
                             + e.message);
                 }
             }
         }

         function onUserMediaSuccess(stream) {
             console.log("User has granted access to local media.");
             var url = webkitURL.createObjectURL(stream);
             localVideo.style.opacity = 1;
             localVideo.src = url;
             localStream = stream;
             // Caller creates PeerConnection.
             if (initiator)
                 maybeStart();
         }

         function maybeStart() {
             if (!started && localStream && channelReady) {
                 setStatus("Connecting...");
                 console.log("Creating PeerConnection.");
                 createPeerConnection();
                 console.log("Adding local stream.");
                 pc.addStream(localStream);
                 started = true;
                 // Caller initiates offer to peer.
                 if (initiator)
                     doCall();
             }
         }

         function doCall() {
             console.log("Sending offer to peer.");
             if (isRTCPeerConnection) {
                 pc.createOffer(setLocalAndSendMessage, null, mediaConstraints);
             } else {
                 var offer = pc.createOffer(mediaConstraints);
                 pc.setLocalDescription(pc.SDP_OFFER, offer);
                 sendMessage({
                     type : 'offer',
                     sdp : offer.toSdp()
                 });
                 pc.startIce();
             }
         }

         function setLocalAndSendMessage(sessionDescription) {
             pc.setLocalDescription(sessionDescription);
             sendMessage(sessionDescription);
         }

         function sendMessage(message) {
             var msgString = JSON.stringify(message);
             console.log('发出信息 : ' + msgString);
             path = 'message?r=${roomKey}' + '&u=${user}';
             var xhr = new XMLHttpRequest();
             xhr.open('POST', path, true);
             xhr.send(msgString);
         }

页面加载完之后会调用initialize方法，initialize方法中调用了getUserMedia方法，这个方法是通过本地摄像头获取视频的方法，在成功获取视频之后发送连接请求，并在客户端建立连接管道，最后通过sendMessage向另外一个客户端发送连接的请求，参数为当前通话的房间号和当前登陆人，下图是连接产生的日志：

 package org.rtc.servlet;

 import java.io.BufferedReader;
 import java.io.IOException;
 import java.io.InputStreamReader;

 import javax.servlet.ServletException;
 import javax.servlet.ServletInputStream;
 import javax.servlet.annotation.WebServlet;
 import javax.servlet.http.HttpServlet;
 import javax.servlet.http.HttpServletRequest;
 import javax.servlet.http.HttpServletResponse;

 import net.sf.json.JSONObject;

 import org.rtc.room.WebRTCRoomManager;
 import org.rtc.websocket.WebRTCMessageInboundPool;

 @WebServlet(urlPatterns = {"/message"})
 public class WebRTCMessageServlet extends HttpServlet {

     private static final long serialVersionUID = 1L;

     public void doGet(HttpServletRequest request, HttpServletResponse response)
             throws ServletException, IOException {
         super.doPost(request, response);
     }

     public void doPost(HttpServletRequest request, HttpServletResponse response)
             throws ServletException, IOException {
         String r = request.getParameter("r");//房间号
         String u = request.getParameter("u");//通话人
         BufferedReader br = new BufferedReader(new InputStreamReader((ServletInputStream)request.getInputStream()));
         String line = null;
         StringBuilder sb = new StringBuilder();
         while((line = br.readLine())!=null){
             sb.append(line); //获取输入流，主要是视频定位的信息
         }

         String message = sb.toString();
         JSONObject json = JSONObject.fromObject(message);
         if (json != null) {
             String type = json.getString("type");
             if ("bye".equals(type)) {//客户端退出视频聊天
                 System.out.println("user :" + u + " exit..");
                 WebRTCRoomManager.removeUser(r, u);
             }
         }
         String otherUser = WebRTCRoomManager.getOtherUser(r, u);//获取通话的对象
         if (u.equals(otherUser)) {
             message = message.replace("\"offer\"", "\"answer\"");
             message = message.replace("a=crypto:0 AES_CM_128_HMAC_SHA1_32",
                     "a=xrypto:0 AES_CM_128_HMAC_SHA1_32");
             message = message.replace("a=ice-options:google-ice\\r\\n", "");
         }
         //向对方发送连接数据
         WebRTCMessageInboundPool.sendMessage(otherUser, message);
     }
 }

就这样通过WebSokcet向客户端发送连接数据，然后客户端根据接收到的数据进行视频接收：

 function onChannelMessage(message) {
             console.log('收到信息 : ' + message.data);
             if (isRTCPeerConnection)
                 processSignalingMessage(message.data);//建立视频连接
             else
                 processSignalingMessage00(message.data);
         }

         function processSignalingMessage(message) {
             var msg = JSON.parse(message);

             if (msg.type === 'offer') {
                 // Callee creates PeerConnection
                 if (!initiator && !started)
                     maybeStart();

                 // We only know JSEP version after createPeerConnection().
                 if (isRTCPeerConnection)
                     pc.setRemoteDescription(new RTCSessionDescription(msg));
                 else
                     pc.setRemoteDescription(pc.SDP_OFFER,
                             new SessionDescription(msg.sdp));

                 doAnswer();
             } else if (msg.type === 'answer' && started) {
                 pc.setRemoteDescription(new RTCSessionDescription(msg));
             } else if (msg.type === 'candidate' && started) {
                 var candidate = new RTCIceCandidate({
                     sdpMLineIndex : msg.label,
                     candidate : msg.candidate
                 });
                 pc.addIceCandidate(candidate);
             } else if (msg.type === 'bye' && started) {
                 onRemoteHangup();
             }
         }

就这样通过Java、WebSocket、WebRTC就实现了在浏览器上的视频通话。

请教

还有一个就自己的一个疑问，我定义的WebSocket失效时间是20秒，时间太短了。希望大家指教一下如何设置WebSocket的失效时间。

截图

演示地址

你可以和你的朋友一起进入 http://blog.csdn.net/leecho571/article/details/8207102 ，感受下Ext结合WebSocket、WebRTC构建的即时通讯

建议大家将chrome升级至最新版本 http://www.google.cn/intl/zh-CN/chrome/browser/eula.html?extra=devchannel&platform=win

源码下载

http://download.csdn.net/detail/leecho571/5117399

大家可以按照这种思路去自己实现，建议大家最好用Chrome浏览器进行测试。

大家可以进群：197331959进行交流。