提示:局域网socket版,一对多
文章目录
- @[TOC](文章目录)
- 前言
- 一、教程
- 二、webrtc工作流程
- 三、推流端
- 四、拉流
- 五、socket服务
- 六、效果
- 七、备注
- 总结
文章目录
- @[TOC](文章目录)
- 前言
- 一、教程
- 二、webrtc工作流程
- 三、推流端
- 四、拉流
- 五、socket服务
- 六、效果
- 七、备注
- 总结
前言
WebRTC(Web Real-Time Communication)是一种实时通讯技术,允许网络应用或站点在不借助中间媒介的情况下,建立浏览器之间的点对点(Peer-to-Peer)连接,实现视频流、音频流或其他任意数据的传输。WebRTC的核心功能包括音视频的采集、编解码、网络传输和显示等
WebRTC的技术特点
1、实时通信:WebRTC专注于实时通信,包括音频、视频和其他数据传输。
2、点对点通信:WebRTC支持点对点通信,即两个浏览器之间直接建立连接,无需通过中间服务器。
3、多媒体引擎:WebRTC包含一个多媒体引擎,处理音频和视频流,并提供丰富的API和协议。
4、NAT穿越:WebRTC提供机制,使得在NAT(Network Address Translation)和防火墙等网络设备背后进行通信更为容易。
5、TURN服务器:当P2P连接无法建立时,WebRTC会利用TURN服务器进行数据中转,确保通信的稳定性
一、教程
webrtc文档
二、webrtc工作流程
// 推流拉流过程
/*** 推流端获取视频stream* 推流端生成offer * 推流端通过offer设置推流LocalDescription* 推流端发送offer给(拉)流端* (拉)流端接收offer* (拉)流端通过offer设置(拉)流端RemoteDescription* (拉)流端生成answer* (拉)流端通过answer设置(拉)流端LocalDescription* (拉)流端发送answer给推流端* 推流端接收answer设置推流端RemoteDescription* 推流端发送candidate(video,audio各一次)* (拉)流端接收candidate* (拉)流端发送candidate(video,audio各一次)* 推流端接收candidate* **/
三、推流端
一个拉流RTCPeerConnection,对应一个推流RTCPeerConnection
X 个拉流RTCPeerConnection,对应X 个推流RTCPeerConnection
push.html
<!DOCTYPE html>
<html lang="en">
<head><meta charset="UTF-8"><meta name="viewport" content="width=device-width, initial-scale=1.0"><title>推流</title>
</head>
<body><video id="webrtcVideo" autoplay></video><script>const video = document.getElementById('webrtcVideo');// webscoketconst ws = new WebSocket('ws://127.0.0.1:1990'); // 可换成局域网ip地址let videoStream;// 一个拉流RTCPeerConnection对应一个推流RTCPeerConnection,xx个拉流RTCPeerConnection,对应xx个推流RTCPeerConnectionconst pushPool = {};// rtc connectionlet pushRtcCon;// 打开摄像头,video标签播放视频流const getStream = async () => {if(!navigator.mediaDevices||!navigator.mediaDevices.getUserMedia)console.log('不支持:getUserMedia');const stream = await navigator.mediaDevices.getUserMedia({video:true});video.srcObject = stream;videoStream = stream;}getStream();// 开始推流const startPush = (pullId) => {if(!pushPool[pullId])pushPool[pullId] = pushRtcCon = new RTCPeerConnection();// rtc connection 添加trackvideoStream.getVideoTracks().forEach(track => {pushRtcCon.addTrack(track,videoStream);});// 监听icecandidatepushRtcCon.onicecandidate = (event)=>{if(event.candidate)ws.send(JSON.stringify({type:'candidate',candidate:event.candidate,id:pullId}))}// 创建offerpushRtcCon.createOffer().then(offer=>{console.log(offer)// 设置推流LocalDescriptionpushRtcCon.setLocalDescription(offer).then(()=>{ console.log('推流设置LocalDescription成功');});// offer信息发送给拉流ws.send(JSON.stringify({type:'offer',id:pullId,offer}))});}// 开启websocket服务ws.addEventListener('open',()=>{// 初始化推流通道ws.send(JSON.stringify({type:'push_init'}))console.log('websocket连接成功')});// 接收wenbscoket信息ws.addEventListener('message', (event) => {let data = JSON.parse(event.data);console.log(data)// 接收到拉流传来的answer 设置推流RemoteDescriptionif(data.type == 'answer')pushRtcCon.setRemoteDescription(data.answer).then(()=>{ console.log('推流设置RemoteDescription成功');});// 接收拉流candidate 推流rtc connection 添加IceCandidateif(data.type == 'candidate'&&data.candidate)pushRtcCon.addIceCandidate(data.candidate).then(()=>{ console.log('推流添加candidate成功');});// 接收拉流开启消息 开始推流if(data.type == 'pull_start')startPush(data.id);})</script>
</body>
</html>
四、拉流
pull.html
<!DOCTYPE html>
<html lang="en">
<head><meta charset="UTF-8"><meta name="viewport" content="width=device-width, initial-scale=1.0"><title>Document</title>
</head>
<body><video id="pullVideo" autoplay preload muted></video><div id="pullBtn">拉流</div><script>const pullBtn = document.getElementById('pullBtn');// 开始拉流const startPll = () =>{let ws = new WebSocket('ws://127.0.0.1:1990'); // 可换成局域网ip地址const pullVideo = document.getElementById('pullVideo');let pullStrem;// 拉流rtc connectionconst pullRtcCon = new RTCPeerConnection();const pullID = new Date().getTime()+'io'+Math.round(Math.random()*10000);// 拉流监听icecandidatepullRtcCon.onicecandidate = (event)=>{// 接收到icecandidate 发送candidate给推流端if(event.candidate)ws.send(JSON.stringify({type:'candidate',candidate:event.candidate,num:1,id:pullID}))}// 监听trackpullRtcCon.addEventListener('track' ,(event) => {pullStrem = event.streams[0];pullVideo.srcObject = event.streams[0];})// 打开webscoketws.addEventListener('open',async ()=>{await ws.send(JSON.stringify({type:'pull_init',id:pullID}));// 通知推流端,开始推流ws.send(JSON.stringify({type:'pull_start',id:pullID}));console.log('websocket连接成功')});// 监听webscoket消息ws.addEventListener('message',(event)=>{let data = JSON.parse(event.data);// 接收到推流端offerconsole.log(data,'????')if(data.type == 'offer'){// 设置拉流端 RemoteDescriptionpullRtcCon.setRemoteDescription(data.offer).then(()=>{console.log('拉流设置RemoteDescription成功')// 创建answerpullRtcCon.createAnswer(data.offer).then((answer)=>{// 设置拉流的LocalDescriptionpullRtcCon.setLocalDescription(answer).then(()=>{console.log('拉流设置LocalDescription成功')});// 发送answer到推流端ws.send(JSON.stringify({type:'answer',answer,id:pullID}))});});}// 接收推流端candidate 拉流端添加IceCandidateif(data.type == 'candidate')pullRtcCon.addIceCandidate(data.candidate).then(()=>{ console.log('拉流添加candidate成功');});})}// 拉流按钮点击事件pullBtn.addEventListener('click',startPll)</script>
</body>
</html>
五、socket服务
安装依赖
npm init
npm install nodejs-websocket -S
index.js
const ws = require('nodejs-websocket');
const port = '1990';// 推流通道 拉流通道
let wsPush,wsPull,pullPool={};
const server = ws.createServer((connection)=>{// websocket 连接接收数据connection.on('text',(msg)=>{let data = JSON.parse(msg);// 初始化推流websocketif(data.type == 'push_init')wsPush = connection;// 初始化拉流websocketif(data.type == 'pull_init')if(!pullPool[data.id]) pullPool[data.id] = connection;// 接收推流消息 发送给拉流if(connection == wsPush&&pullPool[data.id])pullPool[data.id].send(msg);// 接收拉流消息 发送给推流for(let key in pullPool){if(connection == pullPool[key]&&wsPush)wsPush.send(msg);}})// websocket 关闭connection.on('close',()=>{wsPush = null;wsPull = null;console.log('通道关闭')})// websocket 报错connection.on('err',(err)=>{wsPush = null;wsPull = null;console.log('通道报错:'+err)})
})
server.listen(port,console.log('ws启动成功,127.0.0.1:'+port));
六、效果
推流端
拉流端(点击拉流按钮)
七、备注
1、socket地址可换成局域网IP地址访问
2、pull来流请求地址可换成局域网IP地址访问
总结
踩坑路漫漫长@~@