Java+WebSocket+WebRTC实现视频通话实例

最新推荐文章于 2024-05-23 16:25:54 发布

mengzhengjie

最新推荐文章于 2024-05-23 16:25:54 发布

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http://www.tuicool.com/articles/qmE3ii

Java+WebSocket+WebRTC实现视频通话实例

时间 2012-11-04 21:04:55 洗尽铅华

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

主题 WebSocket Java

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（可能需要翻墙访问），这个例子是部署在Google App Engine上的应用程序，依赖与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：

HTML5
不需要任何插件
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
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<String,WebRTCMessageInbound > connections = new HashMap<String,WebRTCMessageInbound>();
  
  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

其他

源码问题，由于各方面的原因源码暂时不能公布出来，主要是现在源码比较乱，而且还有很多BUG需要整理。

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。

mengzhengjie

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Java+WebSocket+WebRTC实现视频通话实例

http://www.tuicool.com/articles/qmE3iiJava+WebSocket+WebRTC实现视频通话实例时间 2012-11-04 21:04:55 洗尽铅华原文 http://blog.csdn.net/leecho571/article/details/8146525主题 WebSocket Java介绍
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