前端WebRTC局域网1V1视频通话

基本概念

WebRTC（Web Real-Time Communications）

网络实时通讯，它允许网络应用或者站点，在不借助中间媒介的情况下，建立点对点（Peer-to-Peer）的连接，实现视频流和音频流或者其他任意数据的传输

NAT（Network Address Translation）

网络地址转换协议，用来给私网设备映射一个公网的 IP 地址

STUN（Session Traversal Utilities for NAT）

会话穿透，通过NAT找到公网地址进行P2P通信

TURN（Traversal Using Relay around NAT）

中继转发，当STUN不可用时，通过TURN转发音视频数据，显然这样是开销最大的
开源STUN&TURN服务器：coturn

ICE（Interactive Connectivity Establishment）

交互式连接建立，即网络信息
candidate：候选，优先级为：局域网、STUN、TURN

SDP（Session Description Protocol）

会话描述协议，即媒体信息，不是音视频流，在WebRTC中分为offer和answer

Signaling Server

信令服务器，用来交换ICE和SDP信息，WebRTC未做规定，自己选择实现技术，比如WebSocket

局域网视频通信

局域网不需要STUN/TURN服务器，只需信令服务器，这里使用Node.js ws库实现

效果

在这里插入图片描述

代码

客户端 index.html

<!DOCTYPE html>
<html><head><meta http-equiv="Content-Type" content="text/html; charset=utf-8" /><meta name="viewport" content="width=device-width, initial-scale=1" /><title>WebRTC Demo</title><style>button {margin: 1rem;}video {width: 300px;}</style>
</head><body><div><div>信令服务器地址：<input id="inputServer" value="ws://192.168.205.165:8888" /><button onclick="start()">开始</button></div><video id="localVideo" autoplay muted></video><video id="remoteVideo" autoplay muted></video></div><script>const inputServer = document.querySelector("#inputServer");const remoteVideo = document.querySelector("#remoteVideo");const localVideo = document.querySelector("#localVideo");let peerConn;let webSocket;let localStream;// 打开本地摄像头navigator.mediaDevices.getUserMedia({ audio: true, video: true }).then((mediaStream) => {localStream = mediaStream;localVideo.srcObject = mediaStream;}).catch((err) => {console.error(err);});// 创建WebRTC连接peerConn = new RTCPeerConnection();peerConn.addEventListener('icecandidate', (event) => {if (event.candidate) {webSocket.send(JSON.stringify({type: "ice",candidate: event.candidate}));}});peerConn.addEventListener("track", (event) => {remoteVideo.srcObject = event.streams[0];});function start() {// 连接信令服务器webSocket = new WebSocket(inputServer.value);webSocket.addEventListener('open', () => {webSocket.send(JSON.stringify({type: "join"}));});// 收到服务端消息webSocket.addEventListener('message', (event) => {const msg = JSON.parse(event.data);console.log(msg);switch (msg.type) {case "sendOffer":peerConn.addTrack(localStream.getVideoTracks()[0], localStream);peerConn.createOffer({ offerToReceiveAudio: true, offerToReceiveVideo: true }).then((offer) => {peerConn.setLocalDescription(offer).then(() => {webSocket.send(JSON.stringify(offer));})});break;case "offer":peerConn.addTrack(localStream.getVideoTracks()[0], localStream);peerConn.setRemoteDescription(msg).then(() => {peerConn.createAnswer().then((answer) => {peerConn.setLocalDescription(answer).then(() => {webSocket.send(JSON.stringify(answer));})})});break;case "answer":peerConn.setRemoteDescription(msg);break;case "ice":peerConn.addIceCandidate(msg.candidate);break;default:}});webSocket.addEventListener('close', () => {console.log("websocket close");});}</script>
</body></html>

服务端 server.mjs

import { WebSocketServer } from 'ws';const wss = new WebSocketServer({ port: 8888 });let clients = []; // 已连接的客户端wss.on('connection', function connection(ws) {ws.on('message', function message(rawData) {const data = rawData.toString();const obj = JSON.parse(data);console.log("type", obj.type);switch (obj.type) {case "join":if (clients.length < 2) {clients.push(ws);if (clients.length === 2) {clients[0].send(JSON.stringify({ type: "sendOffer" }));}} else {console.log("room is full");}break;case "offer":clients[1].send(data);break;case "answer":clients[0].send(data);break;case "ice":clients.forEach((item) => {if (item !== ws) {item.send(data);}})break;default:}});ws.on('error', (err) => console.error("error:", err));ws.on('close', (code) => {console.log("ws close", code);clients = clients.filter((item) => {if (item === ws) {item = null;return false;}return true;});});
});