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

代码

客户端 index.html

html 复制代码

<!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

js 复制代码

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;
		});
	});
});

使用

在文件目录运行命令：node server.mjs
修改信令服务器地址，浏览器打开 index.html
将 index.html 复制到另一台电脑上用浏览器打开
允许使用摄像头和麦克风，两边点击开始按钮即可

参考资料

WebRTC_API
前端使用WebRTC