WebRTC实现音视频通话的代码

WebRTC 是一种用于实时通信的开放 Web 技术，支持音视频通话、文件传输等功能。以下是使用 WebRTC 实现音视频通话的代码示例：

1. 获取本地摄像头和麦克风

navigator.mediaDevices.getUserMedia({
  audio: true,
  video: true
})
.then(stream => {
  // stream 是获取到的本地音视频流
})
.catch(error => {
  console.error(error);
});

2. 创建 RTCPeerConnection 对象

const configuration = {
  iceServers: [
    { urls: 'stun:stun.l.google.com:19302' },
    { urls: 'stun:stun1.l.google.com:19302' },
    { urls: 'stun:stun2.l.google.com:19302' }
  ]
};

const peerConnection = new RTCPeerConnection(configuration);

3. 添加本地流到 RTCPeerConnection 对象

stream.getTracks().forEach(track => {
  peerConnection.addTrack(track, stream);
});

4. 创建 Offer SDP

peerConnection.createOffer().then(offer => {
  return peerConnection.setLocalDescription(offer);
})
.then(() => {
  // 获取 Offer SDP
  const offerSdp = peerConnection.localDescription.sdp;
})
.catch(error => {
  console.error(error);
});

5. 发送 Offer SDP 给远端

// 使用 WebSocket 发送 Offer SDP 到远端
ws.send(JSON.stringify({
  type: 'offer',
  sdp: offerSdp
}));

6. 接收远端的 Answer SDP

ws.onmessage = event => {
  const message = JSON.parse(event.data);
  if (message.type === 'answer') {
    const answerSdp = message.sdp;
    peerConnection.setRemoteDescription(new RTCSessionDescription({
      type: 'answer',
      sdp: answerSdp
    }));
  }
};

7. 创建 Answer SDP

peerConnection.ontrack = event => {
  // 获取远端音视频流
  const remoteStream = event.streams[0];
};

peerConnection.createAnswer().then(answer => {
  return peerConnection.setLocalDescription(answer);
})
.then(() => {
  // 获取 Answer SDP
  const answerSdp = peerConnection.localDescription.sdp;
})
.catch(error => {
  console.error(error);
});

8. 发送 Answer SDP 给远端

// 使用 WebSocket 发送 Answer SDP 到远端
ws.send(JSON.stringify({
  type: 'answer',
  sdp: answerSdp
}));

9. 接收 ICE 候选项

ws.onmessage = event => {
  const message = JSON.parse(event.data);
  if (message.type === 'candidate') {
    const candidate = new RTCIceCandidate({
      sdpMLineIndex: message.label,
      candidate: message.candidate
    });
    peerConnection.addIceCandidate(candidate);
  }
};

10. 发送 ICE 候选项给远端

peerConnection.onicecandidate = event => {
  if (event.candidate) {
    // 使用 WebSocket 发送 ICE 候选项到远端
    ws.send(JSON.stringify({
      type: 'candidate',
      label: event.candidate.sdpMLineIndex,
      id: event.candidate.sdpMid,
      candidate: event.candidate.candidate
    }));
  }
};

以上是使用 WebRTC 实现音视频通话的基本代码示例，具体实现还需要根据业务需求进行调整。