feat: 添加双人对练语音通话功能

- 后端：扩展 SSE 支持 WebRTC 信令消息转发
- 前端：创建 WebRTC 连接管理模块 (webrtc.ts)
- 前端：创建 useVoiceCall 组合式函数
- 前端：在对练房间添加语音通话 UI
- 集成 Web Speech API 实现语音转文字

backend/app/api/v1/practice_room.py

@@ -48,6 +48,13 @@ class JoinRoomRequest(BaseModel):
 class SendMessageRequest(BaseModel):
     """发送消息请求"""
     content: str = Field(..., description="消息内容")
+    source: Optional[str] = Field("text", description="消息来源: text/voice")
+
+
+class WebRTCSignalRequest(BaseModel):
+    """WebRTC 信令请求"""
+    signal_type: str = Field(..., description="信令类型: voice_offer/voice_answer/ice_candidate/voice_start/voice_end")
+    payload: dict = Field(..., description="信令数据（SDP/ICE候选等）")
 
 
 class RoomResponse(BaseModel):
@@ -399,6 +406,60 @@ async def send_message(
     }
 
 
+@router.post("/{room_code}/signal", summary="发送WebRTC信令")
+async def send_signal(
+    room_code: str,
+    request: WebRTCSignalRequest,
+    db: AsyncSession = Depends(get_db),
+    current_user: User = Depends(get_current_user)
+):
+    """
+    发送 WebRTC 信令消息
+    
+    信令类型：
+    - voice_start: 发起语音通话
+    - voice_offer: SDP Offer
+    - voice_answer: SDP Answer
+    - ice_candidate: ICE 候选
+    - voice_end: 结束语音通话
+    """
+    service = PracticeRoomService(db)
+    
+    # 获取房间
+    room = await service.get_room_by_code(room_code.upper())
+    if not room:
+        raise HTTPException(status_code=404, detail="房间不存在")
+    
+    # 检查用户是否在房间中
+    user_role = room.get_user_role(current_user.id)
+    if not user_role:
+        raise HTTPException(status_code=403, detail="您不是房间参与者")
+    
+    # 验证信令类型
+    valid_signal_types = ["voice_start", "voice_offer", "voice_answer", "ice_candidate", "voice_end"]
+    if request.signal_type not in valid_signal_types:
+        raise HTTPException(status_code=400, detail=f"无效的信令类型，必须是: {', '.join(valid_signal_types)}")
+    
+    # 发送信令消息（作为系统消息存储，用于 SSE 推送）
+    message = await service.send_message(
+        room_id=room.id,
+        user_id=current_user.id,
+        content=None,  # 信令消息不需要文本内容
+        role_name=None,
+        message_type=request.signal_type,
+        extra_data=request.payload
+    )
+    
+    return {
+        "code": 200,
+        "message": "信令发送成功",
+        "data": {
+            "signal_type": request.signal_type,
+            "sequence": message.sequence
+        }
+    }
+
+
 @router.get("/{room_code}/messages", summary="获取消息列表")
 async def get_messages(
     room_code: str,

backend/app/services/practice_room_service.py

@@ -271,36 +271,48 @@ class PracticeRoomService:
         self,
         room_id: int,
         user_id: int,
-        content: str,
+        content: Optional[str],
-        role_name: Optional[str] = None
+        role_name: Optional[str] = None,
+        message_type: Optional[str] = None,
+        extra_data: Optional[dict] = None
     ) -> PracticeRoomMessage:
         """
-        发送聊天消息
+        发送聊天消息或信令消息
         
         Args:
             room_id: 房间ID
             user_id: 发送者ID
             content: 消息内容
             role_name: 角色名称
+            message_type: 消息类型（默认为 chat）
+            extra_data: 额外数据（用于 WebRTC 信令等）
             
         Returns:
             PracticeRoomMessage: 消息对象
         """
+        import json
+        
         # 获取当前消息序号
         sequence = await self._get_next_sequence(room_id)
         
+        # 如果是信令消息，将 extra_data 序列化到 content 中
+        actual_content = content
+        if extra_data and not content:
+            actual_content = json.dumps(extra_data)
+        
         message = PracticeRoomMessage(
             room_id=room_id,
             user_id=user_id,
-            message_type=self.MSG_TYPE_CHAT,
+            message_type=message_type or self.MSG_TYPE_CHAT,
-            content=content,
+            content=actual_content,
             role_name=role_name,
             sequence=sequence
         )
         
         self.db.add(message)
         
-        # 更新房间统计
+        # 只有聊天消息才更新房间统计
+        if (message_type or self.MSG_TYPE_CHAT) == self.MSG_TYPE_CHAT:
             room = await self.get_room_by_id(room_id)
             if room:
                 room.total_turns += 1

frontend/src/api/duoPractice.ts

@@ -76,17 +76,35 @@ export interface RoomDetailResponse {
   is_host: boolean
 }
 
+export type MessageType = 
+  | 'chat' 
+  | 'system' 
+  | 'join' 
+  | 'leave' 
+  | 'start' 
+  | 'end'
+  | 'voice_start'
+  | 'voice_offer'
+  | 'voice_answer'
+  | 'ice_candidate'
+  | 'voice_end'
+
 export interface RoomMessage {
   id: number
   room_id: number
   user_id?: number
-  message_type: 'chat' | 'system' | 'join' | 'leave' | 'start' | 'end'
+  message_type: MessageType
   content?: string
   role_name?: string
   sequence: number
   created_at: string
 }
 
+export interface WebRTCSignalRequest {
+  signal_type: 'voice_start' | 'voice_offer' | 'voice_answer' | 'ice_candidate' | 'voice_end'
+  payload: Record<string, any>
+}
+
 export interface MessagesResponse {
   messages: RoomMessage[]
   room_status: string
@@ -185,3 +203,20 @@ export function generateShareLink(roomCode: string): string {
   const baseUrl = window.location.origin
   return `${baseUrl}/trainee/duo-practice/join/${roomCode}`
 }
+
+/**
+ * 发送 WebRTC 信令
+ */
+export function sendSignal(roomCode: string, signalType: string, payload: Record<string, any>) {
+  return request.post(`/api/v1/practice/rooms/${roomCode}/signal`, {
+    signal_type: signalType,
+    payload
+  })
+}
+
+/**
+ * 获取对练报告
+ */
+export function getPracticeReport(roomCode: string) {
+  return request.get(`/api/v1/practice/rooms/${roomCode}/report`)
+}

frontend/src/composables/useVoiceCall.ts

@@ -0,0 +1,462 @@
+/**
+ * 语音通话组合式函数
+ * 
+ * 功能：
+ * - 整合 WebRTC 管理和信令服务
+ * - 管理通话状态
+ * - 处理语音转文字
+ */
+import { ref, computed, onUnmounted } from 'vue'
+import { ElMessage } from 'element-plus'
+import { WebRTCManager, createWebRTCManager, type ConnectionState } from '@/utils/webrtc'
+import request from '@/api/request'
+
+export type VoiceCallState = 'idle' | 'requesting' | 'ringing' | 'connecting' | 'connected' | 'ended'
+
+export interface UseVoiceCallOptions {
+  roomCode: string
+  onTranscript?: (text: string, isFinal: boolean) => void
+  onRemoteTranscript?: (text: string) => void
+}
+
+export function useVoiceCall(options: UseVoiceCallOptions) {
+  const { roomCode, onTranscript, onRemoteTranscript } = options
+
+  // ==================== 状态 ====================
+  const callState = ref<VoiceCallState>('idle')
+  const connectionState = ref<ConnectionState>('idle')
+  const isMuted = ref(false)
+  const isRemoteMuted = ref(false)
+  const localAudioLevel = ref(0)
+  const remoteAudioLevel = ref(0)
+  const callDuration = ref(0)
+  const errorMessage = ref<string | null>(null)
+
+  // 语音识别相关
+  const isTranscribing = ref(false)
+  const currentTranscript = ref('')
+
+  // 内部状态
+  let webrtcManager: WebRTCManager | null = null
+  let recognition: any = null // SpeechRecognition
+  let callTimer: number | null = null
+  let audioLevelTimer: number | null = null
+
+  // ==================== 计算属性 ====================
+  const isCallActive = computed(() => 
+    ['requesting', 'ringing', 'connecting', 'connected'].includes(callState.value)
+  )
+
+  const canStartCall = computed(() => callState.value === 'idle')
+  const canEndCall = computed(() => isCallActive.value)
+
+  // ==================== 信令 API ====================
+  
+  async function sendSignal(signalType: string, payload: any) {
+    try {
+      await request.post(`/api/v1/practice/rooms/${roomCode}/signal`, {
+        signal_type: signalType,
+        payload
+      })
+    } catch (error) {
+      console.error('[VoiceCall] 发送信令失败:', error)
+      throw error
+    }
+  }
+
+  // ==================== 通话控制 ====================
+
+  /**
+   * 发起语音通话
+   */
+  async function startCall() {
+    if (!canStartCall.value) {
+      console.warn('[VoiceCall] 无法发起通话，当前状态:', callState.value)
+      return
+    }
+
+    try {
+      callState.value = 'requesting'
+      errorMessage.value = null
+
+      // 创建 WebRTC 管理器
+      webrtcManager = createWebRTCManager({
+        onConnectionStateChange: handleConnectionStateChange,
+        onIceCandidate: handleIceCandidate,
+        onRemoteStream: handleRemoteStream,
+        onError: handleError
+      })
+
+      // 创建 Offer
+      const offer = await webrtcManager.createOffer()
+
+      // 发送开始信令
+      await sendSignal('voice_start', {})
+
+      // 发送 Offer
+      await sendSignal('voice_offer', {
+        type: offer.type,
+        sdp: offer.sdp
+      })
+
+      callState.value = 'ringing'
+      ElMessage.info('正在呼叫对方...')
+
+    } catch (error: any) {
+      console.error('[VoiceCall] 发起通话失败:', error)
+      errorMessage.value = error.message || '发起通话失败'
+      callState.value = 'idle'
+      webrtcManager?.close()
+      webrtcManager = null
+      ElMessage.error(errorMessage.value)
+    }
+  }
+
+  /**
+   * 接听语音通话
+   */
+  async function answerCall(offer: RTCSessionDescriptionInit) {
+    try {
+      callState.value = 'connecting'
+      errorMessage.value = null
+
+      // 创建 WebRTC 管理器
+      webrtcManager = createWebRTCManager({
+        onConnectionStateChange: handleConnectionStateChange,
+        onIceCandidate: handleIceCandidate,
+        onRemoteStream: handleRemoteStream,
+        onError: handleError
+      })
+
+      // 处理 Offer 并创建 Answer
+      const answer = await webrtcManager.handleOffer(offer)
+
+      // 发送 Answer
+      await sendSignal('voice_answer', {
+        type: answer.type,
+        sdp: answer.sdp
+      })
+
+      ElMessage.success('已接听通话')
+
+    } catch (error: any) {
+      console.error('[VoiceCall] 接听通话失败:', error)
+      errorMessage.value = error.message || '接听通话失败'
+      callState.value = 'idle'
+      webrtcManager?.close()
+      webrtcManager = null
+      ElMessage.error(errorMessage.value)
+    }
+  }
+
+  /**
+   * 拒绝来电
+   */
+  async function rejectCall() {
+    try {
+      await sendSignal('voice_end', { reason: 'rejected' })
+      callState.value = 'idle'
+    } catch (error) {
+      console.error('[VoiceCall] 拒绝通话失败:', error)
+    }
+  }
+
+  /**
+   * 结束通话
+   */
+  async function endCall() {
+    try {
+      await sendSignal('voice_end', { reason: 'ended' })
+    } catch (error) {
+      console.error('[VoiceCall] 发送结束信令失败:', error)
+    }
+
+    cleanup()
+    callState.value = 'ended'
+
+    // 延迟恢复到 idle 状态
+    setTimeout(() => {
+      callState.value = 'idle'
+    }, 1000)
+  }
+
+  /**
+   * 切换静音
+   */
+  function toggleMute() {
+    if (webrtcManager) {
+      isMuted.value = !isMuted.value
+      webrtcManager.setMuted(isMuted.value)
+    }
+  }
+
+  // ==================== 信令处理 ====================
+
+  /**
+   * 处理接收到的信令消息
+   */
+  async function handleSignal(signalType: string, payload: any, fromUserId: number) {
+    console.log('[VoiceCall] 收到信令:', signalType)
+
+    switch (signalType) {
+      case 'voice_start':
+        // 收到通话请求
+        if (callState.value === 'idle') {
+          callState.value = 'ringing'
+          ElMessage.info('收到语音通话请求')
+        }
+        break
+
+      case 'voice_offer':
+        // 收到 Offer，自动接听
+        if (callState.value === 'ringing' || callState.value === 'idle') {
+          await answerCall({
+            type: payload.type,
+            sdp: payload.sdp
+          })
+        }
+        break
+
+      case 'voice_answer':
+        // 收到 Answer
+        if (webrtcManager && callState.value === 'ringing') {
+          await webrtcManager.handleAnswer({
+            type: payload.type,
+            sdp: payload.sdp
+          })
+        }
+        break
+
+      case 'ice_candidate':
+        // 收到 ICE 候选
+        if (webrtcManager && payload.candidate) {
+          await webrtcManager.addIceCandidate(payload)
+        }
+        break
+
+      case 'voice_end':
+        // 对方结束通话
+        cleanup()
+        callState.value = 'ended'
+        ElMessage.info('通话已结束')
+        setTimeout(() => {
+          callState.value = 'idle'
+        }, 1000)
+        break
+    }
+  }
+
+  // ==================== WebRTC 回调 ====================
+
+  function handleConnectionStateChange(state: ConnectionState) {
+    connectionState.value = state
+    
+    if (state === 'connected') {
+      callState.value = 'connected'
+      startCallTimer()
+      startAudioLevelMonitor()
+      startSpeechRecognition()
+      ElMessage.success('语音通话已连接')
+    } else if (state === 'failed' || state === 'disconnected') {
+      if (callState.value === 'connected') {
+        ElMessage.warning('通话连接断开')
+      }
+    }
+  }
+
+  async function handleIceCandidate(candidate: RTCIceCandidate) {
+    try {
+      await sendSignal('ice_candidate', candidate.toJSON())
+    } catch (error) {
+      console.error('[VoiceCall] 发送 ICE 候选失败:', error)
+    }
+  }
+
+  function handleRemoteStream(stream: MediaStream) {
+    console.log('[VoiceCall] 收到远程音频流')
+    
+    // 播放远程音频
+    const audio = new Audio()
+    audio.srcObject = stream
+    audio.play().catch(e => console.error('[VoiceCall] 播放远程音频失败:', e))
+  }
+
+  function handleError(error: Error) {
+    console.error('[VoiceCall] WebRTC 错误:', error)
+    errorMessage.value = error.message
+  }
+
+  // ==================== 语音识别 ====================
+
+  function startSpeechRecognition() {
+    // 检查浏览器支持
+    const SpeechRecognition = (window as any).SpeechRecognition || (window as any).webkitSpeechRecognition
+    
+    if (!SpeechRecognition) {
+      console.warn('[VoiceCall] 浏览器不支持语音识别')
+      return
+    }
+
+    recognition = new SpeechRecognition()
+    recognition.continuous = true
+    recognition.interimResults = true
+    recognition.lang = 'zh-CN'
+
+    recognition.onstart = () => {
+      isTranscribing.value = true
+      console.log('[VoiceCall] 语音识别已启动')
+    }
+
+    recognition.onresult = (event: any) => {
+      let interimTranscript = ''
+      let finalTranscript = ''
+
+      for (let i = event.resultIndex; i < event.results.length; i++) {
+        const transcript = event.results[i][0].transcript
+        if (event.results[i].isFinal) {
+          finalTranscript += transcript
+        } else {
+          interimTranscript += transcript
+        }
+      }
+
+      currentTranscript.value = interimTranscript || finalTranscript
+      
+      if (finalTranscript) {
+        onTranscript?.(finalTranscript, true)
+      } else if (interimTranscript) {
+        onTranscript?.(interimTranscript, false)
+      }
+    }
+
+    recognition.onerror = (event: any) => {
+      console.error('[VoiceCall] 语音识别错误:', event.error)
+      if (event.error !== 'no-speech') {
+        isTranscribing.value = false
+      }
+    }
+
+    recognition.onend = () => {
+      // 如果通话还在进行，重新启动识别
+      if (callState.value === 'connected' && !isMuted.value) {
+        recognition.start()
+      } else {
+        isTranscribing.value = false
+      }
+    }
+
+    try {
+      recognition.start()
+    } catch (error) {
+      console.error('[VoiceCall] 启动语音识别失败:', error)
+    }
+  }
+
+  function stopSpeechRecognition() {
+    if (recognition) {
+      recognition.stop()
+      recognition = null
+    }
+    isTranscribing.value = false
+  }
+
+  // ==================== 辅助功能 ====================
+
+  function startCallTimer() {
+    callDuration.value = 0
+    callTimer = window.setInterval(() => {
+      callDuration.value++
+    }, 1000)
+  }
+
+  function stopCallTimer() {
+    if (callTimer) {
+      clearInterval(callTimer)
+      callTimer = null
+    }
+  }
+
+  function startAudioLevelMonitor() {
+    audioLevelTimer = window.setInterval(async () => {
+      if (webrtcManager) {
+        const localStream = webrtcManager.getLocalStream()
+        const remoteStream = webrtcManager.getRemoteStream()
+
+        if (localStream) {
+          localAudioLevel.value = await webrtcManager.getAudioLevel(localStream)
+        }
+        if (remoteStream) {
+          remoteAudioLevel.value = await webrtcManager.getAudioLevel(remoteStream)
+        }
+      }
+    }, 100)
+  }
+
+  function stopAudioLevelMonitor() {
+    if (audioLevelTimer) {
+      clearInterval(audioLevelTimer)
+      audioLevelTimer = null
+    }
+  }
+
+  function formatDuration(seconds: number): string {
+    const mins = Math.floor(seconds / 60)
+    const secs = seconds % 60
+    return `${mins.toString().padStart(2, '0')}:${secs.toString().padStart(2, '0')}`
+  }
+
+  // ==================== 清理 ====================
+
+  function cleanup() {
+    stopCallTimer()
+    stopAudioLevelMonitor()
+    stopSpeechRecognition()
+    
+    webrtcManager?.close()
+    webrtcManager = null
+
+    isMuted.value = false
+    isRemoteMuted.value = false
+    localAudioLevel.value = 0
+    remoteAudioLevel.value = 0
+    currentTranscript.value = ''
+  }
+
+  // 组件卸载时清理
+  onUnmounted(() => {
+    if (isCallActive.value) {
+      endCall()
+    }
+    cleanup()
+  })
+
+  // ==================== 返回 ====================
+
+  return {
+    // 状态
+    callState,
+    connectionState,
+    isMuted,
+    isRemoteMuted,
+    localAudioLevel,
+    remoteAudioLevel,
+    callDuration,
+    errorMessage,
+    isTranscribing,
+    currentTranscript,
+
+    // 计算属性
+    isCallActive,
+    canStartCall,
+    canEndCall,
+
+    // 方法
+    startCall,
+    answerCall,
+    rejectCall,
+    endCall,
+    toggleMute,
+    handleSignal,
+    formatDuration
+  }
+}

frontend/src/utils/webrtc.ts

@@ -0,0 +1,324 @@
+/**
+ * WebRTC 连接管理模块
+ * 
+ * 功能：
+ * - 管理 RTCPeerConnection 生命周期
+ * - 处理 SDP 交换
+ * - 处理 ICE 候选收集
+ * - 音频流管理
+ */
+
+export type ConnectionState = 'idle' | 'connecting' | 'connected' | 'disconnected' | 'failed'
+
+export interface WebRTCConfig {
+  iceServers?: RTCIceServer[]
+  onLocalStream?: (stream: MediaStream) => void
+  onRemoteStream?: (stream: MediaStream) => void
+  onConnectionStateChange?: (state: ConnectionState) => void
+  onIceCandidate?: (candidate: RTCIceCandidate) => void
+  onError?: (error: Error) => void
+}
+
+// 默认 ICE 服务器配置
+const DEFAULT_ICE_SERVERS: RTCIceServer[] = [
+  { urls: 'stun:stun.l.google.com:19302' },
+  { urls: 'stun:stun1.l.google.com:19302' },
+  { urls: 'stun:stun2.l.google.com:19302' }
+]
+
+export class WebRTCManager {
+  private peerConnection: RTCPeerConnection | null = null
+  private localStream: MediaStream | null = null
+  private remoteStream: MediaStream | null = null
+  private config: WebRTCConfig
+  private connectionState: ConnectionState = 'idle'
+  private pendingIceCandidates: RTCIceCandidate[] = []
+
+  constructor(config: WebRTCConfig = {}) {
+    this.config = {
+      iceServers: DEFAULT_ICE_SERVERS,
+      ...config
+    }
+  }
+
+  /**
+   * 获取当前连接状态
+   */
+  getConnectionState(): ConnectionState {
+    return this.connectionState
+  }
+
+  /**
+   * 获取本地音频流
+   */
+  getLocalStream(): MediaStream | null {
+    return this.localStream
+  }
+
+  /**
+   * 获取远程音频流
+   */
+  getRemoteStream(): MediaStream | null {
+    return this.remoteStream
+  }
+
+  /**
+   * 初始化本地音频流
+   */
+  async initLocalStream(): Promise<MediaStream> {
+    try {
+      this.localStream = await navigator.mediaDevices.getUserMedia({
+        audio: {
+          echoCancellation: true,
+          noiseSuppression: true,
+          autoGainControl: true
+        },
+        video: false
+      })
+      
+      this.config.onLocalStream?.(this.localStream)
+      return this.localStream
+    } catch (error) {
+      const err = error instanceof Error ? error : new Error('获取麦克风权限失败')
+      this.config.onError?.(err)
+      throw err
+    }
+  }
+
+  /**
+   * 创建 PeerConnection
+   */
+  private createPeerConnection(): RTCPeerConnection {
+    const pc = new RTCPeerConnection({
+      iceServers: this.config.iceServers
+    })
+
+    // 监听 ICE 候选
+    pc.onicecandidate = (event) => {
+      if (event.candidate) {
+        console.log('[WebRTC] ICE candidate:', event.candidate.candidate?.substring(0, 50))
+        this.config.onIceCandidate?.(event.candidate)
+      }
+    }
+
+    // 监听连接状态变化
+    pc.onconnectionstatechange = () => {
+      console.log('[WebRTC] Connection state:', pc.connectionState)
+      this.updateConnectionState(pc.connectionState)
+    }
+
+    // 监听 ICE 连接状态
+    pc.oniceconnectionstatechange = () => {
+      console.log('[WebRTC] ICE connection state:', pc.iceConnectionState)
+      if (pc.iceConnectionState === 'failed') {
+        this.updateConnectionState('failed')
+      }
+    }
+
+    // 监听远程流
+    pc.ontrack = (event) => {
+      console.log('[WebRTC] Remote track received')
+      if (event.streams && event.streams[0]) {
+        this.remoteStream = event.streams[0]
+        this.config.onRemoteStream?.(this.remoteStream)
+      }
+    }
+
+    return pc
+  }
+
+  /**
+   * 更新连接状态
+   */
+  private updateConnectionState(state: RTCPeerConnectionState | string) {
+    const stateMap: Record<string, ConnectionState> = {
+      'new': 'connecting',
+      'connecting': 'connecting',
+      'connected': 'connected',
+      'disconnected': 'disconnected',
+      'failed': 'failed',
+      'closed': 'disconnected'
+    }
+    
+    this.connectionState = stateMap[state] || 'idle'
+    this.config.onConnectionStateChange?.(this.connectionState)
+  }
+
+  /**
+   * 创建 Offer（发起方调用）
+   */
+  async createOffer(): Promise<RTCSessionDescriptionInit> {
+    if (!this.localStream) {
+      await this.initLocalStream()
+    }
+
+    this.peerConnection = this.createPeerConnection()
+    this.updateConnectionState('connecting')
+
+    // 添加本地音频轨道
+    this.localStream!.getTracks().forEach(track => {
+      this.peerConnection!.addTrack(track, this.localStream!)
+    })
+
+    // 创建 Offer
+    const offer = await this.peerConnection.createOffer()
+    await this.peerConnection.setLocalDescription(offer)
+
+    console.log('[WebRTC] Offer created')
+    return offer
+  }
+
+  /**
+   * 处理 Offer（接收方调用）
+   */
+  async handleOffer(offer: RTCSessionDescriptionInit): Promise<RTCSessionDescriptionInit> {
+    if (!this.localStream) {
+      await this.initLocalStream()
+    }
+
+    this.peerConnection = this.createPeerConnection()
+    this.updateConnectionState('connecting')
+
+    // 添加本地音频轨道
+    this.localStream!.getTracks().forEach(track => {
+      this.peerConnection!.addTrack(track, this.localStream!)
+    })
+
+    // 设置远程描述
+    await this.peerConnection.setRemoteDescription(new RTCSessionDescription(offer))
+
+    // 处理等待中的 ICE 候选
+    for (const candidate of this.pendingIceCandidates) {
+      await this.peerConnection.addIceCandidate(candidate)
+    }
+    this.pendingIceCandidates = []
+
+    // 创建 Answer
+    const answer = await this.peerConnection.createAnswer()
+    await this.peerConnection.setLocalDescription(answer)
+
+    console.log('[WebRTC] Answer created')
+    return answer
+  }
+
+  /**
+   * 处理 Answer（发起方调用）
+   */
+  async handleAnswer(answer: RTCSessionDescriptionInit): Promise<void> {
+    if (!this.peerConnection) {
+      throw new Error('PeerConnection not initialized')
+    }
+
+    await this.peerConnection.setRemoteDescription(new RTCSessionDescription(answer))
+
+    // 处理等待中的 ICE 候选
+    for (const candidate of this.pendingIceCandidates) {
+      await this.peerConnection.addIceCandidate(candidate)
+    }
+    this.pendingIceCandidates = []
+
+    console.log('[WebRTC] Answer handled')
+  }
+
+  /**
+   * 添加 ICE 候选
+   */
+  async addIceCandidate(candidate: RTCIceCandidateInit): Promise<void> {
+    const iceCandidate = new RTCIceCandidate(candidate)
+
+    if (this.peerConnection && this.peerConnection.remoteDescription) {
+      await this.peerConnection.addIceCandidate(iceCandidate)
+      console.log('[WebRTC] ICE candidate added')
+    } else {
+      // 如果远程描述还没设置，先缓存候选
+      this.pendingIceCandidates.push(iceCandidate)
+      console.log('[WebRTC] ICE candidate queued')
+    }
+  }
+
+  /**
+   * 静音/取消静音本地音频
+   */
+  setMuted(muted: boolean): void {
+    if (this.localStream) {
+      this.localStream.getAudioTracks().forEach(track => {
+        track.enabled = !muted
+      })
+    }
+  }
+
+  /**
+   * 检查是否静音
+   */
+  isMuted(): boolean {
+    if (this.localStream) {
+      const audioTrack = this.localStream.getAudioTracks()[0]
+      return audioTrack ? !audioTrack.enabled : true
+    }
+    return true
+  }
+
+  /**
+   * 获取音频音量级别（用于音量指示器）
+   */
+  async getAudioLevel(stream: MediaStream): Promise<number> {
+    return new Promise((resolve) => {
+      const audioContext = new AudioContext()
+      const analyser = audioContext.createAnalyser()
+      const source = audioContext.createMediaStreamSource(stream)
+      
+      source.connect(analyser)
+      analyser.fftSize = 256
+      
+      const dataArray = new Uint8Array(analyser.frequencyBinCount)
+      analyser.getByteFrequencyData(dataArray)
+      
+      // 计算平均音量
+      const average = dataArray.reduce((a, b) => a + b, 0) / dataArray.length
+      
+      audioContext.close()
+      resolve(average / 255) // 归一化到 0-1
+    })
+  }
+
+  /**
+   * 关闭连接
+   */
+  close(): void {
+    console.log('[WebRTC] Closing connection')
+
+    // 停止本地流
+    if (this.localStream) {
+      this.localStream.getTracks().forEach(track => track.stop())
+      this.localStream = null
+    }
+
+    // 停止远程流
+    if (this.remoteStream) {
+      this.remoteStream.getTracks().forEach(track => track.stop())
+      this.remoteStream = null
+    }
+
+    // 关闭 PeerConnection
+    if (this.peerConnection) {
+      this.peerConnection.close()
+      this.peerConnection = null
+    }
+
+    this.pendingIceCandidates = []
+    this.updateConnectionState('disconnected')
+  }
+
+  /**
+   * 重置管理器
+   */
+  reset(): void {
+    this.close()
+    this.connectionState = 'idle'
+  }
+}
+
+// 导出单例工厂函数
+export function createWebRTCManager(config?: WebRTCConfig): WebRTCManager {
+  return new WebRTCManager(config)
+}

frontend/src/views/trainee/duo-practice-room.vue

@@ -18,6 +18,16 @@
         </div>
       </div>
       <div class="header-right">
+        <!-- 语音通话状态 -->
+        <div class="voice-status" v-if="voiceCall.isCallActive.value">
+          <el-icon class="voice-icon" :class="{ active: voiceCall.callState.value === 'connected' }">
+            <Microphone />
+          </el-icon>
+          <span class="call-duration" v-if="voiceCall.callState.value === 'connected'">
+            {{ voiceCall.formatDuration(voiceCall.callDuration.value) }}
+          </span>
+          <span class="call-status" v-else>{{ voiceCallStatusText }}</span>
+        </div>
         <el-tag :type="statusType" size="large">{{ statusText }}</el-tag>
       </div>
     </div>
@@ -81,6 +91,100 @@
           </el-button>
         </div>
 
+        <!-- 语音通话控制 -->
+        <div class="voice-control" v-if="store.isPracticing">
+          <h4>语音通话</h4>
+          
+          <!-- 未通话状态 -->
+          <div class="voice-idle" v-if="voiceCall.callState.value === 'idle'">
+            <el-button 
+              type="success" 
+              size="large" 
+              round
+              @click="voiceCall.startCall"
+            >
+              <el-icon><Microphone /></el-icon>
+              发起语音
+            </el-button>
+            <p class="voice-hint">点击发起语音通话，实时对话</p>
+          </div>
+
+          <!-- 呼叫中状态 -->
+          <div class="voice-ringing" v-else-if="voiceCall.callState.value === 'ringing'">
+            <div class="ringing-animation">
+              <el-icon :size="48" class="pulse"><Microphone /></el-icon>
+            </div>
+            <p>正在呼叫对方...</p>
+            <el-button type="danger" round @click="voiceCall.endCall">取消</el-button>
+          </div>
+
+          <!-- 来电状态 -->
+          <div class="voice-incoming" v-else-if="voiceCall.callState.value === 'ringing' && !isCallInitiator">
+            <div class="incoming-animation">
+              <el-icon :size="48" class="shake"><PhoneFilled /></el-icon>
+            </div>
+            <p>对方发起语音通话</p>
+            <div class="incoming-actions">
+              <el-button type="success" circle size="large" @click="voiceCall.answerCall">
+                <el-icon><Microphone /></el-icon>
+              </el-button>
+              <el-button type="danger" circle size="large" @click="voiceCall.rejectCall">
+                <el-icon><Close /></el-icon>
+              </el-button>
+            </div>
+          </div>
+
+          <!-- 连接中状态 -->
+          <div class="voice-connecting" v-else-if="voiceCall.callState.value === 'connecting'">
+            <el-icon :size="32" class="spin"><Loading /></el-icon>
+            <p>正在连接...</p>
+          </div>
+
+          <!-- 通话中状态 -->
+          <div class="voice-connected" v-else-if="voiceCall.callState.value === 'connected'">
+            <div class="call-info">
+              <div class="call-timer">{{ voiceCall.formatDuration(voiceCall.callDuration.value) }}</div>
+              <div class="audio-levels">
+                <div class="level-item">
+                  <span>我</span>
+                  <div class="level-bar">
+                    <div class="level-fill" :style="{ width: (voiceCall.localAudioLevel.value * 100) + '%' }"></div>
+                  </div>
+                </div>
+                <div class="level-item">
+                  <span>对方</span>
+                  <div class="level-bar">
+                    <div class="level-fill remote" :style="{ width: (voiceCall.remoteAudioLevel.value * 100) + '%' }"></div>
+                  </div>
+                </div>
+              </div>
+            </div>
+            
+            <!-- 实时转写 -->
+            <div class="transcription" v-if="voiceCall.currentTranscript.value">
+              <el-icon><Edit /></el-icon>
+              <span>{{ voiceCall.currentTranscript.value }}</span>
+            </div>
+
+            <div class="call-controls">
+              <el-button 
+                :type="voiceCall.isMuted.value ? 'danger' : 'default'" 
+                circle 
+                size="large"
+                @click="voiceCall.toggleMute"
+              >
+                <el-icon>
+                  <MuteNotification v-if="voiceCall.isMuted.value" />
+                  <Microphone v-else />
+                </el-icon>
+              </el-button>
+              <el-button type="danger" circle size="large" @click="voiceCall.endCall">
+                <el-icon><Close /></el-icon>
+              </el-button>
+            </div>
+          </div>
+        </div>
+
         <!-- 操作按钮 -->
         <div class="action-buttons">
           <template v-if="store.isHost && store.isReady">
@@ -170,14 +274,46 @@
 import { ref, computed, onMounted, onUnmounted, watch, nextTick } from 'vue'
 import { useRoute, useRouter } from 'vue-router'
 import { ElMessageBox } from 'element-plus'
-import { ArrowLeft, CopyDocument, Plus, Share } from '@element-plus/icons-vue'
+import { 
+  ArrowLeft, CopyDocument, Plus, Share, Microphone, 
+  Close, Loading, MuteNotification, Edit, PhoneFilled 
+} from '@element-plus/icons-vue'
 import { useDuoPracticeStore } from '@/stores/duoPracticeStore'
+import { useVoiceCall } from '@/composables/useVoiceCall'
 import type { RoomMessage } from '@/api/duoPractice'
 
 const route = useRoute()
 const router = useRouter()
 const store = useDuoPracticeStore()
 const messagesContainer = ref<HTMLElement | null>(null)
+const isCallInitiator = ref(false)
+
+// 语音通话
+const voiceCall = useVoiceCall({
+  roomCode: computed(() => store.roomCode).value,
+  onTranscript: handleTranscript
+})
+
+// 语音通话状态文本
+const voiceCallStatusText = computed(() => {
+  const statusMap: Record<string, string> = {
+    'idle': '未连接',
+    'requesting': '请求中...',
+    'ringing': '呼叫中...',
+    'connecting': '连接中...',
+    'connected': '通话中',
+    'ended': '已结束'
+  }
+  return statusMap[voiceCall.callState.value] || ''
+})
+
+// 处理语音转写
+function handleTranscript(text: string, isFinal: boolean) {
+  if (isFinal && text.trim()) {
+    // 将转写的文字作为消息发送
+    store.sendMessage(text)
+  }
+}
 
 // 计算属性
 const statusType = computed(() => {
@@ -277,10 +413,28 @@ const scrollToBottom = () => {
   })
 }
 
-// 监听消息变化，自动滚动到底部
+// 监听消息变化，处理信令消息并自动滚动
-watch(() => store.messages.length, () => {
+watch(() => store.messages, (messages) => {
+  // 处理新消息
+  const latestMsg = messages[messages.length - 1]
+  if (latestMsg) {
+    // 检查是否是信令消息
+    const signalTypes = ['voice_start', 'voice_offer', 'voice_answer', 'ice_candidate', 'voice_end']
+    if (signalTypes.includes(latestMsg.message_type)) {
+      // 不处理自己发的信令
+      const myUserId = store.isHost ? store.hostUser?.id : store.guestUser?.id
+      if (latestMsg.user_id !== myUserId) {
+        try {
+          const payload = latestMsg.content ? JSON.parse(latestMsg.content) : {}
+          voiceCall.handleSignal(latestMsg.message_type, payload, latestMsg.user_id || 0)
+        } catch (e) {
+          console.error('[DuoPracticeRoom] 解析信令失败:', e)
+        }
+      }
+    }
+  }
   scrollToBottom()
-})
+}, { deep: true })
 
 // 初始化
 onMounted(async () => {
@@ -296,6 +450,10 @@ onMounted(async () => {
 
 onUnmounted(() => {
   store.stopPolling()
+  // 结束语音通话
+  if (voiceCall.isCallActive.value) {
+    voiceCall.endCall()
+  }
 })
 </script>
 
@@ -340,6 +498,41 @@ onUnmounted(() => {
       }
     }
   }
+  
+  .header-right {
+    display: flex;
+    align-items: center;
+    gap: 16px;
+    
+    .voice-status {
+      display: flex;
+      align-items: center;
+      gap: 8px;
+      padding: 6px 12px;
+      background: #f0f9eb;
+      border-radius: 20px;
+      
+      .voice-icon {
+        color: #67c23a;
+        
+        &.active {
+          animation: pulse 1.5s infinite;
+        }
+      }
+      
+      .call-duration {
+        font-family: monospace;
+        font-size: 14px;
+        font-weight: 500;
+        color: #67c23a;
+      }
+      
+      .call-status {
+        font-size: 13px;
+        color: #67c23a;
+      }
+    }
+  }
 }
 
 .room-body {
@@ -442,6 +635,137 @@ onUnmounted(() => {
     }
   }
   
+  .voice-control {
+    margin-top: 24px;
+    padding: 20px;
+    background: #f8f9fc;
+    border-radius: 12px;
+    
+    h4 {
+      margin: 0 0 16px 0;
+      font-size: 14px;
+      font-weight: 600;
+      color: #333;
+    }
+    
+    .voice-idle {
+      text-align: center;
+      
+      .voice-hint {
+        margin-top: 12px;
+        font-size: 12px;
+        color: #999;
+      }
+    }
+    
+    .voice-ringing, .voice-incoming, .voice-connecting {
+      text-align: center;
+      
+      p {
+        margin: 12px 0;
+        color: #666;
+      }
+      
+      .ringing-animation, .incoming-animation {
+        .el-icon {
+          color: #67c23a;
+        }
+        
+        .pulse {
+          animation: pulse 1.5s infinite;
+        }
+        
+        .shake {
+          animation: shake 0.5s infinite;
+        }
+      }
+      
+      .incoming-actions {
+        display: flex;
+        justify-content: center;
+        gap: 24px;
+        margin-top: 16px;
+      }
+    }
+    
+    .voice-connecting {
+      .spin {
+        animation: spin 1s linear infinite;
+        color: #409eff;
+      }
+    }
+    
+    .voice-connected {
+      .call-info {
+        .call-timer {
+          text-align: center;
+          font-size: 24px;
+          font-weight: 600;
+          font-family: monospace;
+          color: #67c23a;
+          margin-bottom: 16px;
+        }
+        
+        .audio-levels {
+          .level-item {
+            display: flex;
+            align-items: center;
+            gap: 8px;
+            margin-bottom: 8px;
+            
+            span {
+              width: 40px;
+              font-size: 12px;
+              color: #666;
+            }
+            
+            .level-bar {
+              flex: 1;
+              height: 6px;
+              background: #e0e0e0;
+              border-radius: 3px;
+              overflow: hidden;
+              
+              .level-fill {
+                height: 100%;
+                background: linear-gradient(90deg, #67c23a, #95d475);
+                transition: width 0.1s ease;
+                
+                &.remote {
+                  background: linear-gradient(90deg, #409eff, #79bbff);
+                }
+              }
+            }
+          }
+        }
+      }
+      
+      .transcription {
+        display: flex;
+        align-items: flex-start;
+        gap: 8px;
+        padding: 12px;
+        background: #fff;
+        border-radius: 8px;
+        margin: 16px 0;
+        font-size: 13px;
+        color: #666;
+        
+        .el-icon {
+          color: #409eff;
+          margin-top: 2px;
+        }
+      }
+      
+      .call-controls {
+        display: flex;
+        justify-content: center;
+        gap: 24px;
+        margin-top: 16px;
+      }
+    }
+  }
+  
   .action-buttons {
     margin-top: auto;
     display: flex;
@@ -450,6 +774,23 @@ onUnmounted(() => {
   }
 }
 
+// 动画
+@keyframes pulse {
+  0%, 100% { transform: scale(1); opacity: 1; }
+  50% { transform: scale(1.1); opacity: 0.8; }
+}
+
+@keyframes shake {
+  0%, 100% { transform: translateX(0); }
+  25% { transform: translateX(-5px); }
+  75% { transform: translateX(5px); }
+}
+
+@keyframes spin {
+  from { transform: rotate(0deg); }
+  to { transform: rotate(360deg); }
+}
+
 .chat-panel {
   flex: 1;
   display: flex;