WebRTC学习实现视频

工作中遇到要视频直播的需求，前提是不能依赖Flash前端，于是就找到了WebRtc的相关资料．

什么GetUserMedia,RTCPeerConnection,DataChannel我不多说．

 

简单讲就是谷歌把实时通信层打包进浏览器．而这一套实时通信层又是来源于电信通信领域．

所以浏览器两端交互需要依赖一个叫做信令服务器的东西，来协助两端完成连接．

简单说下流程

以Ａ呼叫Ｂ为例

Ａ呼叫Ｂ

１．告知Server，我要找Ｂ

２．Server查一下有没有Ｂ，有就传达，没有就不传达，至于结果需不需要告知Ａ，那全看心情了．反正这东西是自己实现的．ＷebRTC里就提了一嘴这东西，没具体规范．

３．Ｂ得到Ａ的呼叫（准确说叫Offer）

４．Ｂ解析Offer,回应(Answer)到Server，Server回给A 

５．Ａ收到Answer，解析．

６．这时，Ａ和Ｂ就算勾搭上了．剩下的事情就交给他们自己办了．

 

 

 

Offer 和　Answer　都属于SDP．具体规范http://datatracker.ietf.org/doc/draft-nandakumar-rtcweb-sdp/01/

 

解析这些是交给浏览器做的．

先看Server端的Java实现

package org.rtc.sip;

import java.io.IOException;

import javax.websocket.OnClose;
import javax.websocket.OnError;
import javax.websocket.OnMessage;
import javax.websocket.OnOpen;
import javax.websocket.Session;
import javax.websocket.server.ServerEndpoint;

import com.google.gson.Gson;

@ServerEndpoint(value = "/sipserver")
public class SignalWorker {

	private Gson gson = new Gson();
	private Session session;

	public Session getSession() {
		return session;
	}

	public void setSession(Session session) {
		this.session = session;
	}

	public SignalWorker() {
		
	}

	@OnClose
	public void onClose(Session session) {
	
	}

	@OnError
	public void onError(Throwable throwable) {

	}

	@OnMessage
	public void onMessage(String message, Session session) {
		System.out.println("收到消息：" + message);
		SIPObject sip = gson.fromJson(message, SIPObject.class);
		try {
			if(sip.getAction().equals("login")){
				SIPSessionManager.addNewSession(sip.getFrom(), this);
			}else if(sip.getAction().equals("offer")){
				SignalWorker sker=SIPSessionManager.getSignalWorker(sip.getTo());
					if(sker!=null){
						sker.getSession().getBasicRemote().sendText(message);
					}
					else{
						miss();
					}
			}else if(sip.getAction().equals("answer")){
				SignalWorker sker=SIPSessionManager.getSignalWorker(sip.getTo());
				if(sker!=null){
					sker.getSession().getBasicRemote().sendText(message);
				}
				else{
						miss();
				}
			}else if(sip.getAction().equals("candidate")){
				SignalWorker sker=SIPSessionManager.getSignalWorker(sip.getTo());
				if(sker!=null){
					sker.getSession().getBasicRemote().sendText(message);
				}
			}
		} catch (IOException e) {
			e.printStackTrace();
		}
	}

	@OnOpen
	public void onOpen(Session session) {
		this.session = session;
	}
	public void miss(){
		SIPObject resp=new SIPObject();
		resp.setAction("miss");
		try {
			session.getBasicRemote().sendText(gson.toJson(resp));
		} catch (IOException e) {
			e.printStackTrace();
		}
	}

	public void send(String msg) {
		System.out.println(msg);
		session.getAsyncRemote().sendText(msg);
	}
}

 代码逻辑很简单，就是Ａ->Ｂ,B->A，

 call

 上面HTML代码逻辑也很简单．

两端Baidu和Google先要登陆在SIPServer上．

Google呼叫baidu是在本地完成了摄像头视频获取后发出的Offer.

整个WebRtc交互用一个ＪＳ库屏蔽掉了复杂的代码．

代码只简单实现了功能，未完善．贴出来分享．

/**
 * 
 */
// define override
var RTCPeerConnection = window.RTCPeerConnection || window.mozRTCPeerConnection
		|| window.webkitRTCPeerConnection;
navigator.getUserMedia = navigator.webkitGetUserMedia
		|| navigator.mozGetUserMedia;
//ICE SERVER
var iceServer = {
        "iceServers": [{
            "url": "stun:stun.l.google.com:19302"
        }]
};
//LOG
function log(msg,level){
	console.log(new Date().toLocaleString()+" "+msg);
}



function RTCSuit(opt) {
	this.sipws=opt.sipws;//信令服务器地址
	this.id=opt.id||new Date().getTime();//身份信息
	this.to=opt.to;
	this.conn=null;//websocket对象,connect时初始化
	
	this.sessionId=null;
	this.peercn=new RTCPeerConnection(opt.ice||iceServer);
	this.peercn.rtc=this;
	//init peer
	this.peercn.onicecandidate = RTCSuit.prototype.onIceCandidate;
	this.peercn.onconnecting = RTCSuit.prototype.onPeerConnecting;
	this.peercn.onopen = RTCSuit.prototype.onPeerOpen;
	this.peercn.onaddstream = RTCSuit.prototype.onRemotePeerStreamAdd;
	this.peercn.onremovestream = RTCSuit.prototype.onPeerStreamRemove;
	
	this.localNode=null;
	this.remoteNode=null;
	return this;
}

//视频媒体交互
RTCSuit.prototype.bindLocalMedia=function(id, enableAudio, enableVideo,callback)
{
	var rtc=this;
	navigator.getUserMedia({
		"audio" : enableAudio,
		"video" : enableVideo
	}, function(stream) {//success
		var videoNode=document.getElementById(id);
		rtc.localNode=videoNode;
		videoNode.autoplay=true;
		videoNode.src= URL.createObjectURL(stream);
		rtc.peercn.addStream(stream);
		//触发事件,通知视频流绑定成功
		if(typeof(callback)=="function"){
			callback(true,stream);	
		}
		log("bind local media success");
	},function(error){//error
		if(typeof(callback)=="function"){
			callback(false,null);	
		}
	});
};

RTCSuit.prototype.bindRemoteMedia=function(id){
	var videoNode=document.getElementById(id);
	videoNode.autoplay=true;
	this.remoteNode=videoNode;
};
//PEER交互
RTCSuit.prototype.sendOffer=function(){
	var rtc=this;
	if(!rtc.conn.readyState){
		setTimeout(function(){rtc.sendOffer()},300);return;
	}
	this.peercn.createOffer(function(desc){//success
		rtc.peercn.setLocalDescription(desc);
		rtc.send({action:"offer",sdpWrap:desc});
	},function(error){
		rtc.onSendOfferError(error);
	});
};
RTCSuit.prototype.sendAnswer=function(){
	var rtc=this;
	if(!rtc.conn.readyState){
		setTimeout(function(){rtc.sendOffer()},300);return;
	}
	this.peercn.createAnswer(function(desc){//success
		rtc.peercn.setLocalDescription(desc);
		rtc.send({action:"answer",sdpWrap:desc});
	},function(error){
	});
}

RTCSuit.prototype.onAddStreamToPeer=function(){
	log("get here");
};
RTCSuit.prototype.onIceCandidate=function(event){
	var rtc=this.rtc;
	if (event.candidate) {
		rtc.send({action:"candidate",sdpWrap:{
			type : "candidate",
			label : event.candidate.sdpMLineIndex,
			id : event.candidate.sdpMid,
			candidate : event.candidate.candidate
		}});
	} else {
		console.log("End of candidates.");
	}
};
RTCSuit.prototype.onPeerConnecting=function(){
	log(" peer connecting");
};
RTCSuit.prototype.onPeerOpen=function(){
	log(" peer open");
};
RTCSuit.prototype.onRemotePeerStreamAdd=function(event){
	var url = webkitURL.createObjectURL(event.stream);
	this.rtc.remoteNode.src=url;
};
RTCSuit.prototype.onPeerStreamRemove=function(){
	
};

//信令交互
RTCSuit.prototype.send=function(json){
	var rtc=this;
	json.from=rtc.id;
	json.to=rtc.to;
	var smsg=JSON.stringify(json);
	if(rtc.conn.readyState)
	rtc.conn.send(smsg);
	log(rtc.id+" 发送"+smsg);
};
RTCSuit.prototype.connectSip=function(sipws,callback){
	sipws=sipws||this.sipws;
	var rtc=this;
	this.conn=new WebSocket(sipws);
	this.conn.onopen=function(){
		rtc.send({id:rtc.id,action:"login"});
		log("sip server connected");
	};
	//core msg process
	this.conn.onmessage=function(e){
		log(rtc.id+" 收到消息："+e.data);
		var msg=e.data;
		var json=eval("("+msg+")");
		if(typeof(rtc["on"+json.action]) == "function"){
			rtc["on"+json.action](json);
		}else{
			log("error response:"+msg);
		}
	};
};

RTCSuit.prototype.onoffer=function(json){
	this.peercn.setRemoteDescription(new RTCSessionDescription(json.sdpWrap));
	this.sendAnswer();
	log("recv offer")
};
RTCSuit.prototype.onanswer=function(json){
	this.peercn.setRemoteDescription(new RTCSessionDescription(json.sdpWrap));
	log("recv  answer");
};
RTCSuit.prototype.oncandidate=function(json){
		var msg=json.sdpWrap;
		var candidate = new RTCIceCandidate({
		sdpMLineIndex : msg.label,
		candidate : msg.candidate
		});
	this.peercn.addIceCandidate(candidate);
	
};

 

最后代码奉上．．．．环境 tomcat8.0 java 1.8