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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