vue实现多人视频
Vue 实现多人视频的基础架构
使用 Vue 结合 WebRTC 技术实现多人视频通话需要搭建信令服务器和处理媒体流交换。典型方案采用 Socket.IO 进行信令传输,配合简单的 STUN/TURN 服务器解决 NAT 穿透问题。
核心依赖安装
安装必要的 npm 包:
npm install socket.io-client webrtc-adapter vue-webrtc
或使用更底层的 API:
npm install peerjs simple-peer
信令服务器配置
创建信令服务(Node.js 示例):
const io = require('socket.io')(3000);
io.on('connection', socket => {
socket.on('join', roomId => {
socket.join(roomId);
socket.to(roomId).emit('user-connected', socket.id);
});
socket.on('signal', (toId, signal) => {
io.to(toId).emit('signal', socket.id, signal);
});
});
前端实现步骤
初始化 WebRTC 连接:
// 在Vue组件中
data() {
return {
localStream: null,
peers: {},
socket: io('http://localhost:3000')
}
},
methods: {
async startCall(roomId) {
this.localStream = await navigator.mediaDevices.getUserMedia({ video: true, audio: true });
this.socket.emit('join', roomId);
this.socket.on('user-connected', userId => {
const peer = new SimplePeer({ initiator: true, stream: this.localStream });
peer.on('signal', signal => this.socket.emit('signal', userId, signal));
peer.on('stream', stream => this.$refs[`remoteVideo-${userId}`].srcObject = stream);
this.peers[userId] = peer;
});
this.socket.on('signal', (fromId, signal) => {
this.peers[fromId].signal(signal);
});
}
}
视频元素渲染
模板部分实现:
<template>
<div>
<video ref="localVideo" autoplay muted></video>
<video v-for="(peer, id) in peers" :key="id" :ref="`remoteVideo-${id}`" autoplay></video>
</div>
</template>
高级功能扩展
实现屏幕共享:
async shareScreen() {
const screenStream = await navigator.mediaDevices.getDisplayMedia();
Object.values(this.peers).forEach(peer => {
peer.replaceTrack(
this.localStream.getVideoTracks()[0],
screenStream.getVideoTracks()[0],
this.localStream
);
});
}
错误处理与优化
添加 ICE 重启逻辑:
peer.on('ice', () => {
peer.signal({ type: 'iceRestart' });
});
使用 TURN 服务器配置:
const peer = new SimplePeer({
config: {
iceServers: [
{ urls: 'stun:stun.l.google.com:19302' },
{
urls: 'turn:your-turn-server.com',
username: 'user',
credential: 'pass'
}
]
}
});
性能注意事项
-
限制视频分辨率:根据参与人数动态调整
constraints{ video: { width: { ideal: 640 }, height: { ideal: 480 } } } -
实现带宽自适应:使用
RTCRtpSender.setParameters()const sender = peer.getSenders()[0]; const parameters = sender.getParameters(); parameters.encodings[0].maxBitrate = 100000; // 100kbps sender.setParameters(parameters); -
添加离席检测:
socket.on('user-disconnected', userId => { this.peers[userId].destroy(); delete this.peers[userId]; });







