使用js搭建简易的WebRTC实现视频直播

首先需要一个信令服务器，我们使用nodejs来搭建。两个端：发送端和接收端。
我的目录结构如下图：

流程

创建一个文件夹 WebRTC-Test。
进入文件夹中，新建一个node的文件夹。
使用终端并进入node的目录下，使用
```
npm init
```
创建package.json。

新建server.js，复制一下代码

const app = require('express')();
const wsInstance = require('express-ws')(app);const cors = require('cors');
app.use(cors({ origin: 'http://localhost:3000' }));app.ws('/', ws => {ws.on('message', data => {// 未做业务处理，收到消息后直接广播wsInstance.getWss().clients.forEach(server => {if (server !== ws) {server.send(data);console.log(data,)}});});
});console.log("服务启动： http://localhost:8080");
app.listen(8080, '0.0.0.0');

下载信令服务器的依赖。

npm install express;
npm install express-ws;
npm install cors;

使用node server.js启动node的服务。

准备接收方的代码（receive.html）。

<!DOCTYPE html>
<html lang="zh-CN"><head><meta charset="UTF-8"><meta http-equiv="X-UA-Compatible" content="IE=edge"><meta name="viewport" content="width=device-width, initial-scale=1.0"><meta http-equiv="Content-Security-Policy" content="upgrade-insecure-requests"><title>Receiver</title>
</head><body><video autoplay id="remote"></video>
</body>
<script>const remoteVideo = document.querySelector('#remote')const socket = new WebSocket('ws://localhost:8080');socket.onopen = function () {console.log("Socket Success")}let buddy = new RTCPeerConnection()// 如果接收到对方的视频socket.onmessage = function (e) {const { type, sdp, iceCandidate } = JSON.parse(e.data)console.log(type)switch (type) {case "offer":buddy.setRemoteDescription(new RTCSessionDescription({ type, sdp }))buddy.createAnswer().then(answer => {buddy.setLocalDescription(answer)socket.send(JSON.stringify(answer))})break;case "offer_ice":buddy.addIceCandidate(iceCandidate)break;default:break;}}buddy.ontrack = function (e) {remote.srcObject = e.streams[0]}buddy.onicecandidate = function (e) {if (e.candidate) {socket.send(JSON.stringify({type: "answer_ice",iceCandidate: e.candidate}))}}</script></html>

准备发送方的代码（send.html）。

<!DOCTYPE html>
<html lang="zh-CN"><head><meta charset="UTF-8"><meta http-equiv="X-UA-Compatible" content="IE=edge"><meta name="viewport" content="width=device-width, initial-scale=1.0"><meta http-equiv="Content-Security-Policy" content="upgrade-insecure-requests"><title>Send</title>
</head><body><script>const socket = new WebSocket('ws://localhost:8080');socket.onopen = function () {console.log("Socket Success")}let peer = new RTCPeerConnection()navigator.mediaDevices.getUserMedia({video: true,audio: true}).then(stream => {stream.getTracks().forEach(track => {peer.addTrack(track, stream);})peer.createOffer().then(offer => {peer.setLocalDescription(offer);socket.send(JSON.stringify(offer));})})peer.onicecandidate = function (e) {if (e.candidate) {socket.send(JSON.stringify({type: "offer_ice",iceCandidate: e.candidate}))}}// 如果接收到对方的视频socket.onmessage = function (e) {const { type, sdp, iceCandidate } = JSON.parse(e.data)console.log(type)switch (type) {case "answer":peer.setRemoteDescription(new RTCSessionDescription({ type, sdp }))break;case "answer_ice":peer.addIceCandidate(iceCandidate)break;default:break;}}</script>
</body></html>