音视频通话功能完成（P2P）
已知问题： 文件无法发送
This commit is contained in:
@@ -1,7 +1,8 @@
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import { ref, reactive } from 'vue'
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import { reactive, shallowRef } from 'vue'
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import * as systemApi from '@/api/modules/system'
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import * as messageApi from '@/api/modules/message'
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import { wsManager } from '@/api/websocket'
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import { useToastStore } from '@/stores/toast'
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import type { ChatMessage } from '@/types/api'
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import type { CallStatus } from '@/types/message'
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@@ -14,9 +15,13 @@ export interface CallState {
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id: string | null
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muted: boolean
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camOff: boolean
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duration: number
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startTime: number | null
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}
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export function useWebRTC(userId: string) {
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export function useWebRTC(userId: string, onIncomingCall?: (senderUserId: string) => void) {
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const toastStore = useToastStore()
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const call = reactive<CallState>({
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active: false,
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minimized: false,
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@@ -26,91 +31,134 @@ export function useWebRTC(userId: string) {
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id: null,
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muted: false,
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camOff: false,
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duration: 0,
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startTime: null,
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})
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const localVideo = ref<HTMLVideoElement | null>(null)
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const remoteVideo = ref<HTMLVideoElement | null>(null)
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// 使用 shallowRef 存储 MediaStream，避免 Vue 进行深层代理导致的性能问题
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const localStream = shallowRef<MediaStream | null>(null)
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const remoteStream = shallowRef<MediaStream | null>(null)
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let durationTimer: number | null = null
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let pc: RTCPeerConnection | null = null
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let localStream: MediaStream | null = null
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const pendingCandidates: RTCIceCandidate[] = []
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let currentReceiverUserId = ''
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/**
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* 初始化媒体流
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*/
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async function initMedia(videoEnabled: boolean): Promise<void> {
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try {
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const constraints = { video: videoEnabled, audio: true }
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const stream = await navigator.mediaDevices.getUserMedia(constraints)
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localStream = stream
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if (videoEnabled && localVideo.value) {
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localVideo.value.srcObject = stream
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// 停止之前的流
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if (localStream.value) {
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localStream.value.getTracks().forEach(t => t.stop())
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}
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const constraints: MediaStreamConstraints = {
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video: videoEnabled ? {
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width: { ideal: 1280 },
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height: { ideal: 720 },
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facingMode: 'user', // 优先使用前置摄像头
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} : false,
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audio: {
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echoCancellation: true,
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noiseSuppression: true,
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autoGainControl: true,
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},
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}
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const stream = await navigator.mediaDevices.getUserMedia(constraints)
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localStream.value = stream // 赋值给响应式对象，视图会自动更新
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console.log('[WebRTC] Local media stream obtained')
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} catch (error) {
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console.error('Failed to get media:', error)
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throw new Error('无法获取设备权限或设备不支持')
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throw new Error('无法获取摄像头或麦克风权限')
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}
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}
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/**
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* 获取ICE服务器配置
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* 获取并增强 ICE Servers 配置
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*/
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async function getIceServers() {
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const servers: RTCIceServer[] = []
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try {
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const servers = await systemApi.getIceServers(userId)
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if (servers && servers.length > 0) {
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return servers.map((s) => ({
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const apiServers = await systemApi.getIceServers(userId)
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if (apiServers && apiServers.length > 0) {
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servers.push(...apiServers.map((s: any) => ({
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urls: s.urls,
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username: s.username,
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credential: s.credential,
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}))
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})))
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}
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} catch (error) {
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console.error('Failed to fetch ICE servers:', error)
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console.warn('Failed to fetch ICE servers from API')
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}
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// Fallback: 使用默认STUN
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return [{ urls: 'stun:stun.l.google.com:19302' }]
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// 添加公共 STUN 服务器作为兜底
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servers.push({ urls: 'stun:stun.l.google.com:19302' })
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servers.push({ urls: 'stun:global.stun.twilio.com:3478' })
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return servers
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}
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/**
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* 创建PeerConnection
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* 创建 PeerConnection
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*/
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async function createPC(): Promise<void> {
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const iceServers = await getIceServers()
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iceServers.push({ urls: 'stun:stun.l.google.com:19302' })
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pc = new RTCPeerConnection({ iceServers })
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// 关闭旧连接
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if (pc) pc.close()
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if (localStream) {
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localStream.getTracks().forEach((track) => {
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pc!.addTrack(track, localStream!)
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pc = new RTCPeerConnection({
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iceServers,
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iceCandidatePoolSize: 10
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})
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// 添加本地轨道
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if (localStream.value) {
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localStream.value.getTracks().forEach((track) => {
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pc!.addTrack(track, localStream.value!)
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})
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} else {
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console.warn('[WebRTC] No local stream to add to PC')
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}
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// 监听远程轨道
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pc.ontrack = (e) => {
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if (remoteVideo.value) {
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remoteVideo.value.srcObject = e.streams[0]
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console.log('[WebRTC] Received remote track', e.streams)
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if (e.streams && e.streams[0]) {
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remoteStream.value = e.streams[0] // 赋值给响应式对象
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}
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}
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// 监听 ICE 候选
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pc.onicecandidate = (e) => {
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if (e.candidate && call.id) {
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sendSignal('candidate', e.candidate)
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}
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}
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// 监听连接状态
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pc.onconnectionstatechange = () => {
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console.log('[WebRTC] Connection state:', pc?.connectionState)
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if (pc?.connectionState === 'disconnected' || pc?.connectionState === 'failed') {
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toastStore.error('通话连接中断')
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}
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}
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}
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let currentReceiverUserId = ''
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/**
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* 发送信令
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* 发送信令 (统一封装)
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*/
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function sendSignal(status: CallStatus, data?: any, receiverUserId?: string) {
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if (!call.id) return
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const targetUserId = receiverUserId || currentReceiverUserId
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if (!targetUserId) return
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const payload = {
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sender_client_id: wsManager.getClientId() || '',
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receiver_user_id: receiverUserId || currentReceiverUserId,
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receiver_user_id: targetUserId,
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room_id: '',
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message_type: 6,
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message_type: 6, // Signaling Message
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content: JSON.stringify(data || {}),
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call_id: call.id,
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call_status: status,
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@@ -120,9 +168,33 @@ export function useWebRTC(userId: string) {
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messageApi.sendMessage(payload).catch(console.error)
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}
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/**
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* 开始通话
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*/
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// --- 计时器逻辑 ---
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function startCallTimer() {
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stopCallTimer()
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call.startTime = Date.now()
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call.duration = 0
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durationTimer = window.setInterval(() => {
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if (call.startTime) call.duration = Math.floor((Date.now() - call.startTime) / 1000)
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}, 1000)
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}
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function stopCallTimer() {
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if (durationTimer) {
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clearInterval(durationTimer)
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durationTimer = null
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}
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call.startTime = null
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call.duration = 0
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}
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function formatDuration(seconds: number): string {
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const m = Math.floor(seconds / 60).toString().padStart(2, '0')
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const s = (seconds % 60).toString().padStart(2, '0')
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return `${m}:${s}`
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}
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// --- 通话控制 ---
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async function startCall(type: 'audio' | 'video', receiverUserId: string) {
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currentReceiverUserId = receiverUserId
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call.type = type
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@@ -130,160 +202,160 @@ export function useWebRTC(userId: string) {
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call.active = true
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call.minimized = false
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call.status = 'outgoing'
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call.statusText = '等待对方接听...'
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call.statusText = '正在呼叫...'
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call.duration = 0
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remoteStream.value = null // 重置远程流
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try {
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await initMedia(type === 'video')
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await createPC()
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sendSignal('invite', undefined, receiverUserId)
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} catch (error: any) {
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alert(error.message || '无法启动通话')
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toastStore.error(error.message || '无法启动通话')
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endCall()
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}
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}
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/**
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* 接听通话
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*/
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async function acceptCall(senderUserId?: string) {
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if (senderUserId) {
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currentReceiverUserId = senderUserId
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}
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if (senderUserId) currentReceiverUserId = senderUserId
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call.status = 'connected'
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call.statusText = '连接中...'
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call.statusText = '正在连接...'
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try {
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await initMedia(call.type === 'video')
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await createPC()
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const offer = await pc!.createOffer()
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await pc!.setLocalDescription(offer)
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sendSignal('accepted', undefined, senderUserId)
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sendSignal('offer', offer, senderUserId)
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// 发送 accepted，等待发起方创建 Offer
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sendSignal('accepted', undefined, currentReceiverUserId)
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} catch (error) {
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console.error('Failed to accept call:', error)
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endCall()
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}
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}
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/**
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* 结束通话
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*/
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function endCall() {
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sendSignal('hangup')
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closeCall()
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}
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/**
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* 关闭通话
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*/
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function closeCall() {
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call.active = false
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call.status = 'idle'
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call.statusText = ''
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call.id = null
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stopCallTimer()
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if (pc) {
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pc.close()
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pc = null
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}
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if (localStream) {
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localStream.getTracks().forEach((track) => track.stop())
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localStream = null
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}
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if (localVideo.value) {
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localVideo.value.srcObject = null
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}
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if (remoteVideo.value) {
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remoteVideo.value.srcObject = null
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// 停止本地流
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if (localStream.value) {
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localStream.value.getTracks().forEach(t => t.stop())
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localStream.value = null
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}
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remoteStream.value = null
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}
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/**
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* 处理信令消息
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*/
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// --- 信令处理 ---
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async function handleSignaling(message: ChatMessage) {
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const signal = message.call_status as CallStatus
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const extra = typeof message.extra === 'string' ? JSON.parse(message.extra || '{}') : message.extra
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const content = message.content ? JSON.parse(message.content) : {}
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const extra = message.extra ? (typeof message.extra === 'string' ? JSON.parse(message.extra) : message.extra) : {}
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let callType = 'video'
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if (extra && extra.type) {
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callType = extra.type
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}
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// 更新通话类型
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if (extra.type) call.type = extra.type
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if (signal === 'invite') {
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currentReceiverUserId = message.sender_user_id
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call.id = message.call_id || Date.now().toString()
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call.type = callType as 'audio' | 'video'
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call.id = message.call_id
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call.active = true
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call.minimized = false
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call.status = 'incoming'
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call.statusText = `对方邀请您进行${callType === 'video' ? '视频' : '语音'}通话...`
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call.statusText = `邀请你进行${call.type === 'video' ? '视频' : '语音'}通话`
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if (onIncomingCall) onIncomingCall(message.sender_user_id)
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} else if (signal === 'accepted') {
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// 对方已接受，作为发起方，创建 Offer
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call.status = 'connected'
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call.statusText = '通话中'
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startCallTimer()
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const offer = await pc!.createOffer()
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await pc!.setLocalDescription(offer)
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sendSignal('offer', offer, message.sender_user_id)
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sendSignal('offer', offer)
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} else if (signal === 'offer') {
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const desc = JSON.parse(message.content)
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// 接收 Offer，创建 Answer
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if (!pc) {
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await initMedia(call.type === 'video')
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await createPC()
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}
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await pc!.setRemoteDescription(desc)
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await pc!.setRemoteDescription(content)
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// 处理缓冲的 Candidates
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processPendingCandidates()
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const answer = await pc!.createAnswer()
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await pc!.setLocalDescription(answer)
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sendSignal('answer', answer, message.sender_user_id)
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call.status = 'connected'
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call.statusText = '通话中'
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startCallTimer()
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} else if (signal === 'answer') {
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await pc!.setRemoteDescription(JSON.parse(message.content))
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// 接收 Answer
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await pc!.setRemoteDescription(content)
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processPendingCandidates()
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} else if (signal === 'candidate') {
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await pc!.addIceCandidate(JSON.parse(message.content))
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if (pc && pc.remoteDescription) {
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await pc.addIceCandidate(content)
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} else {
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pendingCandidates.push(content)
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}
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} else if (['hangup', 'ended', 'answered_elsewhere'].includes(signal)) {
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closeCall()
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if (signal === 'answered_elsewhere') {
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alert('已在其他设备接听')
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if (signal === 'answered_elsewhere') toastStore.info('已在其他设备接听')
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}
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}
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async function processPendingCandidates() {
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while (pendingCandidates.length > 0) {
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const candidate = pendingCandidates.shift()
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if (candidate && pc) {
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await pc.addIceCandidate(candidate)
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}
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}
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}
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/**
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* 切换静音
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*/
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// --- 媒体控制 ---
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function toggleMute() {
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call.muted = !call.muted
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if (localStream) {
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localStream.getAudioTracks()[0].enabled = !call.muted
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if (localStream.value) {
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localStream.value.getAudioTracks().forEach(t => t.enabled = !call.muted)
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}
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}
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/**
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* 切换摄像头
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*/
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function toggleCamera() {
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call.camOff = !call.camOff
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if (localStream) {
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const videoTrack = localStream.getVideoTracks()[0]
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if (videoTrack) {
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videoTrack.enabled = !call.camOff
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}
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if (localStream.value) {
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localStream.value.getVideoTracks().forEach(t => t.enabled = !call.camOff)
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}
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}
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return {
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call,
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localVideo,
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remoteVideo,
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localStream, // 暴露流
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remoteStream, // 暴露流
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startCall,
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acceptCall,
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endCall,
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handleSignaling,
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toggleMute,
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toggleCamera,
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formatDuration,
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}
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}
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