Java+Netty+WebRTC、语音、视频、屏幕共享【聊天室设计实践】

企业开发 2023-04-07 11:03:31 阅读次数: 0

背景

本文使用webtrc实现了一个简单的语音视频聊天室、支持多人音视频聊天、屏幕共享。

环境配置

音视频功能需要在有Https协议的域名下才能获取到设备信息，

测试环境搭建Https服务参考Windows下Nginx配置SSL实现Https访问（包含openssl证书生成）_殷长庆的博客-CSDN博客

正式环境可以申请一个免费的证书 

复杂网络环境下需要自己搭建turnserver，网络上搜索大多是使用coturn来搭建turn服务 

turn默认监听端口3478，可以使用webrtc.github.io 测试服务是否可用

本文在局域网内测试，不必要部署turn，使用的谷歌的stun:stun.l.google.com:19302

webrtc参考文章

WebRTC技术简介 - 知乎 (zhihu.com)

实现 

服务端 

服务端使用netty构建一个websocket服务，用来完成为音视频传递ICE信息等工作。 

maven配置

<project xmlns="http://maven.apache.org/POM/4.0.0" xmlns:xsi="http://www.w3.org/2001/XMLSchema-instance"
	xsi:schemaLocation="http://maven.apache.org/POM/4.0.0 http://maven.apache.org/xsd/maven-4.0.0.xsd">
	<modelVersion>4.0.0</modelVersion>
	<groupId>com.luck.cc</groupId>
	<artifactId>cc-im</artifactId>
	<version>1.0-SNAPSHOT</version>
	<name>cc-im</name>
	<url>http://maven.apache.org</url>

	<properties>
		<java.home>${env.JAVA_HOME}</java.home>
		<project.build.sourceEncoding>UTF-8</project.build.sourceEncoding>
		<java.version>1.8</java.version>
	</properties>

	<dependencies>
		<dependency>
			<groupId>io.netty</groupId>
			<artifactId>netty-all</artifactId>
			<version>4.1.74.Final</version>
		</dependency>
        <dependency>
            <groupId>cn.hutool</groupId>
            <artifactId>hutool-all</artifactId>
            <version>5.5.7</version>
        </dependency>
	</dependencies>
	<build>
		<plugins>
			<plugin>
				<artifactId>maven-compiler-plugin</artifactId>
				<configuration>
					<source>1.8</source>
					<target>1.8</target>
				</configuration>
			</plugin>
	        <plugin>
	            <artifactId>maven-assembly-plugin</artifactId>
	            <version>3.0.0</version>
	            <configuration>
	                <archive>
	                    <manifest>
	                        <mainClass>com.luck.im.ServerStart</mainClass>
	                    </manifest>
	                </archive>
	                <descriptorRefs>
	                    <descriptorRef>jar-with-dependencies</descriptorRef>
	                </descriptorRefs>
	            </configuration>
	            <executions>
	                <execution>
	                    <id>make-assembly</id>
	                    <phase>package</phase>
	                    <goals>
	                        <goal>single</goal>
	                    </goals>
	                </execution>
	            </executions>
	        </plugin>
		</plugins>
	</build>
</project>

 JAVA代码

 聊天室服务

package com.luck.im;

import java.util.List;

import io.netty.bootstrap.ServerBootstrap;
import io.netty.channel.ChannelFuture;
import io.netty.channel.ChannelHandlerContext;
import io.netty.channel.ChannelInitializer;
import io.netty.channel.ChannelPipeline;
import io.netty.channel.EventLoopGroup;
import io.netty.channel.nio.NioEventLoopGroup;
import io.netty.channel.socket.SocketChannel;
import io.netty.channel.socket.nio.NioServerSocketChannel;
import io.netty.handler.codec.MessageToMessageCodec;
import io.netty.handler.codec.http.HttpServerCodec;
import io.netty.handler.codec.http.websocketx.TextWebSocketFrame;
import io.netty.handler.codec.http.websocketx.WebSocketServerProtocolHandler;

public class ChatSocket {
	private static EventLoopGroup bossGroup = new NioEventLoopGroup();
	private static EventLoopGroup workerGroup = new NioEventLoopGroup();
	private static ChannelFuture channelFuture;

	/**
	 * 启动服务代理
	 * 
	 * @throws Exception
	 */
	public static void startServer() throws Exception {
		try {
			ServerBootstrap b = new ServerBootstrap();
			b.group(bossGroup, workerGroup).channel(NioServerSocketChannel.class)
					.childHandler(new ChannelInitializer<SocketChannel>() {
						@Override
						public void initChannel(SocketChannel ch) throws Exception {
							ChannelPipeline pipeline = ch.pipeline();
							pipeline.addLast(new HttpServerCodec());
							pipeline.addLast(
									new WebSocketServerProtocolHandler("/myim", null, true, Integer.MAX_VALUE, false));
							pipeline.addLast(new MessageToMessageCodec<TextWebSocketFrame, String>() {
								@Override
								protected void decode(ChannelHandlerContext ctx, TextWebSocketFrame frame,
										List<Object> list) throws Exception {
									list.add(frame.text());
								}

								@Override
								protected void encode(ChannelHandlerContext ctx, String msg, List<Object> list)
										throws Exception {
									list.add(new TextWebSocketFrame(msg));
								}
							});
							pipeline.addLast(new ChatHandler());
						}
					});
			channelFuture = b.bind(8321).sync();

			channelFuture.channel().closeFuture().sync();
		} finally {
			shutdown();
			// 服务器已关闭
		}
	}

	public static void shutdown() {
		if (channelFuture != null) {
			channelFuture.channel().close().syncUninterruptibly();
		}
		if ((bossGroup != null) && (!bossGroup.isShutdown())) {
			bossGroup.shutdownGracefully();
		}
		if ((workerGroup != null) && (!workerGroup.isShutdown())) {
			workerGroup.shutdownGracefully();
		}
	}

}

聊天室业务 

package com.luck.im;

import java.util.Map;
import java.util.concurrent.ConcurrentHashMap;

import cn.hutool.json.JSONObject;
import cn.hutool.json.JSONUtil;
import io.netty.channel.Channel;
import io.netty.channel.ChannelHandlerContext;
import io.netty.channel.SimpleChannelInboundHandler;
import io.netty.util.AttributeKey;
import io.netty.util.internal.StringUtil;

public class ChatHandler extends SimpleChannelInboundHandler<String> {

	/** 用户集合 */
	private static Map<String, Channel> umap = new ConcurrentHashMap<>();

	/** channel绑定自己的用户ID */
	public static final AttributeKey<String> UID = AttributeKey.newInstance("uid");

	@Override
	public void channelRead0(ChannelHandlerContext ctx, String msg) {
		JSONObject parseObj = JSONUtil.parseObj(msg);
		Integer type = parseObj.getInt("t");
		String uid = parseObj.getStr("uid");
		String tid = parseObj.getStr("tid");
		switch (type) {
		case 0:
			// 心跳
			break;
		case 1:
			// 用户加入聊天室
			umap.put(uid, ctx.channel());
			ctx.channel().attr(UID).set(uid);
			umap.forEach((x, y) -> {
				if (!x.equals(uid)) {
					JSONObject json = new JSONObject();
					json.set("t", 2);
					json.set("uid", uid);
					json.set("type", "join");
					y.writeAndFlush(json.toString());
				}
			});
			break;
		case 2:
			Channel uc = umap.get(tid);
			if (null != uc) {
				uc.writeAndFlush(msg);
			}
			break;
		case 9:
			// 用户退出聊天室
			umap.remove(uid);
			leave(ctx, uid);
			ctx.close();
			break;
		default:
			break;
		}
	}

	@Override
	public void channelInactive(ChannelHandlerContext ctx) throws Exception {
		String uid = ctx.channel().attr(UID).get();
		if (StringUtil.isNullOrEmpty(uid)) {
			super.channelInactive(ctx);
			return;
		}
		ctx.channel().attr(UID).set(null);
		umap.remove(uid);
		leave(ctx, uid);
		super.channelInactive(ctx);
	}

	/**
	 * 用户退出
	 * 
	 * @param ctx
	 * @param uid
	 */
	private void leave(ChannelHandlerContext ctx, String uid) {
		umap.forEach((x, y) -> {
			if (!x.equals(uid)) {
				// 把数据转到用户服务
				JSONObject json = new JSONObject();
				json.set("t", 9);
				json.set("uid", uid);
				y.writeAndFlush(json.toString());
			}
		});
	}

	@Override
	public void exceptionCaught(ChannelHandlerContext ctx, Throwable cause) throws Exception {
		cause.printStackTrace();
		ctx.close();
	}
}

启动类

package com.luck.im;

public class ServerStart {
	public static void main(String[] args) throws Exception {
		// 启动聊天室
		ChatSocket.startServer();
	}
}

前端

网页主要使用了adapter-latest.js，下载地址webrtc.github.io

github访问不了可以用webrtc/adapter-latest.js-Javascript文档类资源-CSDN文库 

index.html 

<!DOCTYPE html>
<html>
<head>
    <meta charset="UTF-8">
    <title>聊天室</title>
	<style>video{width:100px;height:100px}</style>
</head>
<body>
<video id="localVideo" autoplay playsinline></video>
<video id="screenVideo" autoplay playsinline></video>
<div id="videos"></div>
<div id="screenVideos"></div>
<div>
<button onclick="startScreen()">开启屏幕共享</button>
<button onclick="closeScreen()">关闭屏幕共享</button>
<button onclick="startVideo()">开启视频</button>
<button onclick="closeVideo()">关闭视频</button>
<button onclick="startAudio()">开启语音</button>
<button onclick="closeAudio()">关闭语音</button>
<button onclick="leave()">退出</button>
</div>
</body>
<script src="https://webrtc.github.io/adapter/adapter-latest.js"></script>
<script>
function getUid(id){
	return id?id:uid;
}
// 开启屏幕共享
function startScreen(id){
	id=getUid(id);
	if(id!=uid){
		sendMsg(id,{type:'startScreen'})
		return false;
	}
	if(!screenVideo.srcObject){
		let options = {audio: false, video: true};
		navigator.mediaDevices.getDisplayMedia(options)
		.then(stream => {
			screenVideo.srcObject = stream;
			for(let i in remotes){
				onmessage({uid:i,t:2,type:'s_join'});
			}
			stream.getVideoTracks()[0].addEventListener('ended', () => {
				closeScreen();
			});
		}) 
	}
}
function selfCloseScreen(ot){
	screenVideo.srcObject.getVideoTracks()[0].stop()
	for(let i in remotes){
		sendMsg(i,{type:'closeScreen',ot:ot})
	}
	screenVideo.srcObject=null;
}
// 关闭屏幕共享
function closeScreen(id,ot){
	id=getUid(id);
	ot=(ot?ot:1);
	if(id!=uid){
		if(ot==1&&remotes[id].screenVideo){
			remotes[id].screenVideo.srcObject=null;
		}else{
			sendMsg(id,{type:'closeScreen',ot:2})
		}
		return false;
	}
	if(screenVideo.srcObject&&ot==1){
		selfCloseScreen(ot)
	}
}
// 开启视频
function startVideo(id){
	id=getUid(id);
	if(id!=uid){
		sendMsg(id,{type:'startVideo'})
		return false;
	}
	let v = localVideo.srcObject.getVideoTracks();
	if(v&&v.length>0&&!v[0].enabled){
		v[0].enabled=true;
	}
}
// 关闭视频
function closeVideo(id){
	id=getUid(id);
	if(id!=uid){
		sendMsg(id,{type:'closeVideo'})
		return false;
	}
	let v = localVideo.srcObject.getVideoTracks();
	if(v&&v.length>0&&v[0].enabled){
		v[0].enabled=false;
	}
}
// 开启语音
function startAudio(id){
	id=getUid(id);
	if(id!=uid){
		sendMsg(id,{type:'startAudio'})
		return false;
	}
	let v = localVideo.srcObject.getAudioTracks();
	if(v&&v.length>0&&!v[0].enabled){
		v[0].enabled=true;
	}
}
// 关闭语音
function closeAudio(id){
	id=getUid(id);
	if(id!=uid){
		sendMsg(id,{type:'closeAudio'})
		return false;
	}
	let v = localVideo.srcObject.getAudioTracks();
	if(v&&v.length>0&&v[0].enabled){
		v[0].enabled=false;
	}
}
// 存储通信方信息 
const remotes = {}
// 本地视频预览 
const localVideo = document.querySelector('#localVideo')
// 视频列表区域 
const videos = document.querySelector('#videos')
// 同屏视频预览 
const screenVideo = document.querySelector('#screenVideo')
// 同屏视频列表区域 
const screenVideos = document.querySelector('#screenVideos')
// 用户ID
var uid = new Date().getTime() + '';
var ws = new WebSocket('wss://127.0.0.1/myim');
ws.onopen = function() {
	heartBeat();
	onopen();
}
// 心跳保持ws连接
function heartBeat(){
	setInterval(()=>{
		ws.send(JSON.stringify({ t: 0 }))
	},3000);
}
function onopen() {
	navigator.mediaDevices
	.getUserMedia({
		audio: true, // 本地测试防止回声 
		video: true
	})
	.then(stream => {
		localVideo.srcObject = stream;
		ws.send(JSON.stringify({ t: 1, uid: uid }));
		ws.onmessage = function(event) {
			onmessage(JSON.parse(event.data));
		}
	}) 
}
async function onmessage(message) {
	if(message.t==9){
		onleave(message.uid);
	}
	if(message.t==2&&message.type)
	switch (message.type) {
		case 'join': {
			// 有新的人加入就重新设置会话，重新与新加入的人建立新会话 
			createRTC(message.uid,localVideo.srcObject,1)
			const pc = remotes[message.uid].pc
			const offer = await pc.createOffer()
			pc.setLocalDescription(offer)
			sendMsg(message.uid, { type: 'offer', offer })
			if(screenVideo.srcObject){
				onmessage({uid:message.uid,t:2,type:'s_join'});
			}
			break
		}
		case 'offer': {
			createRTC(message.uid,localVideo.srcObject,1)
			const pc = remotes[message.uid].pc
			pc.setRemoteDescription(new RTCSessionDescription(message.offer))
			const answer = await pc.createAnswer()
			pc.setLocalDescription(answer)
			sendMsg(message.uid, { type: 'answer', answer })
			break
		}
		case 'answer': {
			const pc = remotes[message.uid].pc
			pc.setRemoteDescription(new RTCSessionDescription(message.answer))
			break
		}
		case 'candidate': {
			const pc = remotes[message.uid].pc
			pc.addIceCandidate(new RTCIceCandidate(message.candidate))
			break
		}
		case 's_join': {
			createRTC(message.uid,screenVideo.srcObject,2)
			const pc = remotes[message.uid].s_pc
			const offer = await pc.createOffer()
			pc.setLocalDescription(offer)
			sendMsg(message.uid, { type: 's_offer', offer })
			break
		}
		case 's_offer': {
			createRTC(message.uid,screenVideo.srcObject,2)
			const pc = remotes[message.uid].s_pc
			pc.setRemoteDescription(new RTCSessionDescription(message.offer))
			const answer = await pc.createAnswer()
			pc.setLocalDescription(answer)
			sendMsg(message.uid, { type: 's_answer', answer })
			break
		}
		case 's_answer': {
			const pc = remotes[message.uid].s_pc
			pc.setRemoteDescription(new RTCSessionDescription(message.answer))
			break
		}
		case 's_candidate': {
			const pc = remotes[message.uid].s_pc
			pc.addIceCandidate(new RTCIceCandidate(message.candidate))
			break
		}
		case 'startScreen': {
			startScreen()
			break
		}
		case 'closeScreen': {
			const ot = message.ot
			if(ot==1){
				closeScreen(message.uid,1)
			}else{
				closeScreen(uid,1)
			}
			break
		}
		case 'startVideo': {
			startVideo()
			break
		}
		case 'closeVideo': {
			closeVideo()
			break
		}
		case 'startAudio': {
			startAudio()
			break
		}
		case 'closeAudio': {
			closeAudio()
			break
		}
		default:
			console.log(message)
			break
	}
}
function removeScreenVideo(id){
	if(remotes[id].s_pc){
		remotes[id].s_pc.close()
		if(remotes[id].screenVideo)
		screenVideos.removeChild(remotes[id].screenVideo)
	}
}
function onleave(id) {
	if (remotes[id]) {
		remotes[id].pc.close()
		videos.removeChild(remotes[id].video)
		removeScreenVideo(id)
		delete remotes[id]
	}
}
function leave() {
	ws.send(JSON.stringify({ t: 9, uid: uid }));
}

// socket发送消息 
function sendMsg(tid, msg) {
	msg.t = 2;
	msg.tid=tid;
	msg.uid=uid;
	ws.send(JSON.stringify(msg))
}
// 创建RTC对象，一个RTC对象只能与一个远端连接 
function createRTC(id,stream,type) {
	const pc = new RTCPeerConnection({
		iceServers: [
			{
				urls: 'stun:stun.l.google.com:19302'
			}
		]
	})

	// 获取本地网络信息，并发送给通信方 
	pc.addEventListener('icecandidate', event => {
		if (event.candidate) {
			// 发送自身的网络信息到通信方 
			sendMsg(id, {
				type: (type==1?'candidate':'s_candidate'),
				candidate: {
					sdpMLineIndex: event.candidate.sdpMLineIndex,
					sdpMid: event.candidate.sdpMid,
					candidate: event.candidate.candidate
				}
			})
		}
	})

	// 有远程视频流时，显示远程视频流 
	pc.addEventListener('track', event => {
		if(type==1){
			if(!remotes[id].video){
				const video = createVideo()
				videos.append(video)
				remotes[id].video=video
			}
			remotes[id].video.srcObject = event.streams[0]
		}else{
			if(!remotes[id].screenVideo){
				const video = createVideo()
				screenVideos.append(video)
				remotes[id].screenVideo=video
			}
			remotes[id].screenVideo.srcObject = event.streams[0]
		}
	})

	// 添加本地视频流到会话中 
	if(stream){
		stream.getTracks().forEach(track => pc.addTrack(track, stream))
	}

	if(!remotes[id]){remotes[id]={}}
	if(type==1){
		remotes[id].pc=pc
	}else{
		remotes[id].s_pc=pc
	}
}
function createVideo(){
	const video = document.createElement('video')
	video.setAttribute('autoplay', true)
	video.setAttribute('playsinline', true)
	return video
}
</script>
</html>

Nginx配置

上面的index.html文件放到D盘根目录下了，然后配置一下websocket

    server {
        listen       443 ssl;
        server_name    mytest.com;
    
        ssl_certificate      lee/lee.crt;
        ssl_certificate_key  lee/lee.key;
    
        ssl_session_cache    shared:SSL:1m;
        ssl_session_timeout  5m;
    
        ssl_ciphers  HIGH:!aNULL:!MD5;
        ssl_prefer_server_ciphers  on;
    
        location / {
            root   d:/;
            index  index.html index.htm index.php;
        }
    
        location /myim {
            proxy_pass http://127.0.0.1:8321/myim;
        }
    }

运行 

java启动

java -jar cc-im.jar

网页访问

https://127.0.0.1/index.html

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