关于 webrtc
提供 WebRTC 点对点连接、信令、数据通道和屏幕共享的实现示例,也涵盖用于群组通话的 LiveKit,以及连接可靠性的生产检查项。
主要功能
- 01点对点连接配置
- 02Socket.IO 信令
- 03数据通道消息传递
- 04屏幕共享
- 05LiveKit 群组通话示例
- 06生产环境可靠性检查
适用场景
- 01构建一对一 WebRTC 通话
- 02通过数据通道交换消息
- 03在通话中共享屏幕
- 04使用 SFU 规划群组通话
安装
复制下面的命令,在终端中运行。
$
npx skills add https:/ / github.com/ claude-dev-suite/ claude-dev-suite --skill webrtc安装后在 Claude Code 中输入/webrtc确认已加载。
详细介绍
WebRTC
对等连接设置
const config: RTCConfiguration = {
iceServers: [
{ urls: 'stun:stun.l.google.com:19302' },
{ urls: 'turn:turn.example.com', username: 'user', credential: 'pass' },
],
};
const pc = new RTCPeerConnection(config);
// Get local media
const stream = await navigator.mediaDevices.getUserMedia({ video: true, audio: true });
stream.getTracks().forEach((track) => pc.addTrack(track, stream));
// Display remote stream
pc.ontrack = (event) => {
remoteVideo.srcObject = event.streams[0];
};
// ICE candidates — send to remote peer via signaling
pc.onicecandidate = (event) => {
if (event.candidate) {
signalingChannel.send({ type: 'ice-candidate', candidate: event.candidate });
}
};
信令(通过 Socket.IO)
// Caller
const offer = await pc.createOffer();
await pc.setLocalDescription(offer);
socket.emit('signal', { type: 'offer', sdp: offer });
// Callee — on receiving offer
socket.on('signal', async (msg) => {
if (msg.type === 'offer') {
await pc.setRemoteDescription(msg.sdp);
const answer = await pc.createAnswer();
await pc.setLocalDescription(answer);
socket.emit('signal', { type: 'answer', sdp: answer });
} else if (msg.type === 'answer') {
await pc.setRemoteDescription(msg.sdp);
} else if (msg.type === 'ice-candidate') {
await pc.addIceCandidate(msg.candidate);
}
});
数据通道
const dataChannel = pc.createDataChannel('chat', { ordered: true });
dataChannel.onopen = () => dataChannel.send('Hello!');
dataChannel.onmessage = (e) => console.log('Received:', e.data);
// Remote side
pc.ondatachannel = (event) => {
const channel = event.channel;
channel.onmessage = (e) => console.log('Received:', e.data);
};
屏幕共享
const screenStream = await navigator.mediaDevices.getDisplayMedia({
video: { cursor: 'always' },
audio: true,
});
const videoTrack = screenStream.getVideoTracks()[0];
const sender = pc.getSenders().find((s) => s.track?.kind === 'video');
await sender?.replaceTrack(videoTrack);
videoTrack.onended = () => {
// User stopped sharing — switch back to camera
const cameraTrack = localStream.getVideoTracks()[0];
sender?.replaceTrack(cameraTrack);
};
LiveKit(SFU — 推荐用于群组通话)
import { Room, RoomEvent } from 'livekit-client';
const room = new Room();
await room.connect(LIVEKIT_URL, accessToken);
await room.localParticipant.enableCameraAndMicrophone();
room.on(RoomEvent.TrackSubscribed, (track, publication, participant) => {
const element = track.attach();
document.getElementById('remote-videos')!.appendChild(element);
});
room.on(RoomEvent.ParticipantDisconnected, (participant) => {
console.log(`${participant.identity} left`);
});
反模式
| 反模式 | 修复方法 |
|---|---|
| 未配置 TURN 服务 | 始终配置 TURN,以便穿透 NAT |
| 群组通话(>4 位用户)使用 P2P | 使用 SFU(LiveKit、mediasoup) |
| 未处理 getUserMedia 错误 | 处理 NotAllowedError、NotFoundError |
| 通过未加密通道传输信令 | 使用 WSS(WebSocket Secure) |
| 未监控连接状态 | 监听 connectionstatechange 事件 |
生产环境检查清单
- 已部署并配置 TURN 服务
- 群组视频(>3 位参与者)使用 SFU
- 拒绝媒体访问时显示备用 UI
- 监控连接质量
- 带宽自适应(simulcast)
- 按需提供录制功能
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