fix(conversation): make realtime sessions cancellable and recoverable

apps/desktop/src/renderer/hooks/useRealtimeConversation.ts

@@ -3,27 +3,28 @@
 // Supabase realtime-token(IPC 경유)으로 ephemeral key를 받아
 // 렌더러가 OpenAI와 직접 WebRTC 연결한다 (마이크 캡처 + 오디오 재생 자동).
 //
-// 이벤트 매핑 (data channel 'oai-events'):
+// 이 훅은 얇은 React 어댑터다:
-//   response.created                              → thinking
+//   - 연결 수립/취소/재연결 정책 → services/realtimeSessionController.ts
-//   response.(output_)audio_transcript.delta      → speaking + 스트리밍 텍스트
+//   - 서버 이벤트 → 대화 상태 매핑 → services/realtimeConversationModel.ts
-//   conversation.item.input_audio_transcription.completed → 유저 메시지
+//   - WebRTC·IPC·fetch IO → services/realtimeSession.ts (RealtimeTransport 포트)
-//   response.done                                 → 어시스턴트 메시지 확정 + listening 복귀
 
-import { useState, useRef, useCallback, useEffect } from 'react'
+import { useEffect, useState, useSyncExternalStore } from 'react'
 import type { ConversationState, ConversationMessage } from '@d3ro/core/types'
+import {
+  createBrowserRealtimeTransport,
+  type RealtimeTransportFactory,
+} from '../services/realtimeSession'
+import {
+  RealtimeSessionController,
+  type RealtimeConnectionState,
+  type RealtimeStartResult,
+} from '../services/realtimeSessionController'
 
-const REALTIME_CALLS_URL = 'https://api.openai.com/v1/realtime/calls'
+export type { RealtimeConnectionState, RealtimeStartResult }
-const DATA_CHANNEL_NAME = 'oai-events'
 
-export type RealtimeConnectionState = 'idle' | 'connecting' | 'live' | 'error'
+export interface UseRealtimeConversationOptions {
-
+  /** 전송 계층 주입 (기본값: 브라우저 WebRTC) */
-interface RealtimeServerEvent {
+  createTransport?: RealtimeTransportFactory
-  type: string
-  delta?: string
-  transcript?: string
-  response?: { id?: string }
-  item_id?: string
-  error?: { message?: string }
 }
 
 export interface UseRealtimeConversationResult {
@@ -34,273 +35,41 @@ export interface UseRealtimeConversationResult {
   streamingText: string
   streamingMsgId: string | null
   error: string | null
-  /** 연결 시작. 실패 시 false (로컬 파이프라인 fallback 유도) */
+  /**
-  start: () => Promise<boolean>
+   * 연결 시작. 이벤트 채널이 열린 뒤 'live'로 resolve한다.
+   * 'failed'면 로컬 파이프라인 fallback을 유도하고, 'cancelled'(시작 중 stop/언마운트)면 아무것도 하지 않는다.
+   */
+  start: () => Promise<RealtimeStartResult>
   stop: () => void
+  /** 연결 중이면 큐에 쌓았다가 채널이 열리면 보낸다 */
   sendText: (text: string) => void
   cancelResponse: () => void
   clearMessages: () => void
 }
 
-let messageSeq = 0
+export function useRealtimeConversation(
-function nextMessageId(prefix: string): string {
+  options: UseRealtimeConversationOptions = {},
-  messageSeq += 1
+): UseRealtimeConversationResult {
-  return `rt-${prefix}-${Date.now()}-${messageSeq}`
+  const [controller] = useState(
-}
+    () => new RealtimeSessionController(options.createTransport ?? createBrowserRealtimeTransport),
-
-export function useRealtimeConversation(): UseRealtimeConversationResult {
-  const [connectionState, setConnectionState] = useState<RealtimeConnectionState>('idle')
-  const [conversationState, setConversationState] = useState<ConversationState>('idle')
-  const [messages, setMessages] = useState<ConversationMessage[]>([])
-  const [streamingText, setStreamingText] = useState('')
-  const [streamingMsgId, setStreamingMsgId] = useState<string | null>(null)
-  const [error, setError] = useState<string | null>(null)
-
-  const pcRef = useRef<RTCPeerConnection | null>(null)
-  const dcRef = useRef<RTCDataChannel | null>(null)
-  const micStreamRef = useRef<MediaStream | null>(null)
-  const audioElRef = useRef<HTMLAudioElement | null>(null)
-  const assistantBufferRef = useRef('')
-  const assistantMsgIdRef = useRef<string | null>(null)
-
-  const cleanup = useCallback(() => {
-    dcRef.current?.close()
-    dcRef.current = null
-    pcRef.current?.close()
-    pcRef.current = null
-    micStreamRef.current?.getTracks().forEach((track) => track.stop())
-    micStreamRef.current = null
-    if (audioElRef.current) {
-      audioElRef.current.srcObject = null
-      audioElRef.current.remove()
-      audioElRef.current = null
-    }
-    assistantBufferRef.current = ''
-    assistantMsgIdRef.current = null
-  }, [])
-
-  // 언마운트 시 연결 정리
-  useEffect(() => cleanup, [cleanup])
-
-  const flushAssistantMessage = useCallback(() => {
-    const content = assistantBufferRef.current.trim()
-    const msgId = assistantMsgIdRef.current
-    assistantBufferRef.current = ''
-    assistantMsgIdRef.current = null
-    setStreamingText('')
-    setStreamingMsgId(null)
-    if (content && msgId) {
-      setMessages((prev) => [
-        ...prev,
-        { id: msgId, role: 'assistant', content, timestamp: Date.now() },
-      ])
-    }
-  }, [])
-
-  const handleServerEvent = useCallback(
-    (event: RealtimeServerEvent) => {
-      switch (event.type) {
-        case 'response.created': {
-          assistantBufferRef.current = ''
-          assistantMsgIdRef.current = nextMessageId('assistant')
-          setConversationState('thinking')
-          break
-        }
-
-        // GA/신규 이벤트명 모두 수용
-        case 'response.output_audio_transcript.delta':
-        case 'response.audio_transcript.delta':
-        case 'response.output_text.delta':
-        case 'response.text.delta': {
-          if (event.delta) {
-            assistantBufferRef.current += event.delta
-            setStreamingMsgId(assistantMsgIdRef.current)
-            setStreamingText(assistantBufferRef.current)
-            setConversationState('speaking')
-          }
-          break
-        }
-
-        case 'conversation.item.input_audio_transcription.completed': {
-          const transcript = (event.transcript ?? '').trim()
-          if (transcript) {
-            setMessages((prev) => [
-              ...prev,
-              {
-                id: nextMessageId('user'),
-                role: 'user',
-                content: transcript,
-                timestamp: Date.now(),
-              },
-            ])
-          }
-          break
-        }
-
-        case 'response.done': {
-          flushAssistantMessage()
-          setConversationState('listening')
-          break
-        }
-
-        case 'error': {
-          setError(event.error?.message ?? 'Realtime error')
-          break
-        }
-
-        default:
-          break
-      }
-    },
-    [flushAssistantMessage],
   )
+  const snapshot = useSyncExternalStore(controller.subscribe, controller.getSnapshot)
 
-  const start = useCallback(async (): Promise<boolean> => {
+  // 언마운트 시 진행 중인 시작까지 취소하고 연결 정리 (컨트롤러는 재사용 가능 → StrictMode 재마운트 안전)
-    if (pcRef.current) return true
+  useEffect(() => controller.stop, [controller])
-
-    setError(null)
-    setConnectionState('connecting')
-
-    try {
-      // 1) ephemeral token (Supabase realtime-token → OpenAI client_secrets)
-      const tokenResult = await window.electronAPI.voiceConversation.getRealtimeToken({})
-      if (!tokenResult.success) {
-        throw new Error(tokenResult.error?.message ?? 'token request failed')
-      }
-      const { value: ephemeralKey, model } = tokenResult.data
-
-      // 2) 마이크 + peer connection
-      const micStream = await navigator.mediaDevices.getUserMedia({ audio: true })
-      micStreamRef.current = micStream
-
-      const pc = new RTCPeerConnection()
-      pcRef.current = pc
-      const micTrack = micStream.getAudioTracks()[0]
-      if (micTrack) pc.addTrack(micTrack, micStream)
-
-      // 3) 원격 오디오 재생
-      const audioEl = document.createElement('audio')
-      audioEl.autoplay = true
-      audioElRef.current = audioEl
-      pc.ontrack = (e) => {
-        audioEl.srcObject = e.streams[0] ?? null
-      }
-
-      pc.onconnectionstatechange = () => {
-        const cs = pc.connectionState
-        if (cs === 'failed' || cs === 'disconnected' || cs === 'closed') {
-          setConnectionState((prev) => (prev === 'live' ? 'error' : prev))
-        }
-      }
-
-      // 4) 이벤트 데이터 채널
-      const dc = pc.createDataChannel(DATA_CHANNEL_NAME)
-      dcRef.current = dc
-      dc.addEventListener('message', (e: MessageEvent) => {
-        try {
-          handleServerEvent(JSON.parse(e.data as string) as RealtimeServerEvent)
-        } catch {
-          // 파싱 불가 이벤트 무시
-        }
-      })
-      dc.addEventListener('open', () => {
-        // 유저 발화 전사 활성화 — 미지원 모델이면 error 이벤트만 오고 대화는 유지됨
-        dc.send(
-          JSON.stringify({
-            type: 'session.update',
-            session: {
-              type: 'realtime',
-              audio: {
-                input: { transcription: { model: 'whisper-1' } },
-              },
-            },
-          }),
-        )
-      })
-
-      // 5) SDP 교환
-      const offer = await pc.createOffer()
-      await pc.setLocalDescription(offer)
-
-      const sdpResponse = await fetch(`${REALTIME_CALLS_URL}?model=${encodeURIComponent(model)}`, {
-        method: 'POST',
-        body: offer.sdp,
-        headers: {
-          Authorization: `Bearer ${ephemeralKey}`,
-          'Content-Type': 'application/sdp',
-        },
-      })
-      if (!sdpResponse.ok) {
-        const text = await sdpResponse.text()
-        throw new Error(`SDP exchange failed (${sdpResponse.status}): ${text.slice(0, 200)}`)
-      }
-
-      const answerSdp = await sdpResponse.text()
-      await pc.setRemoteDescription({ type: 'answer', sdp: answerSdp })
-
-      setConnectionState('live')
-      setConversationState('listening')
-      return true
-    } catch (err) {
-      cleanup()
-      setConnectionState('error')
-      setConversationState('idle')
-      setError(err instanceof Error ? err.message : String(err))
-      return false
-    }
-  }, [cleanup, handleServerEvent])
-
-  const stop = useCallback(() => {
-    flushAssistantMessage()
-    cleanup()
-    setConnectionState('idle')
-    setConversationState('idle')
-  }, [cleanup, flushAssistantMessage])
-
-  const sendText = useCallback((text: string) => {
-    const dc = dcRef.current
-    if (!dc || dc.readyState !== 'open') return
-    setMessages((prev) => [
-      ...prev,
-      { id: nextMessageId('user'), role: 'user', content: text, timestamp: Date.now() },
-    ])
-    dc.send(
-      JSON.stringify({
-        type: 'conversation.item.create',
-        item: {
-          type: 'message',
-          role: 'user',
-          content: [{ type: 'input_text', text }],
-        },
-      }),
-    )
-    dc.send(JSON.stringify({ type: 'response.create' }))
-  }, [])
-
-  const cancelResponse = useCallback(() => {
-    const dc = dcRef.current
-    if (!dc || dc.readyState !== 'open') return
-    dc.send(JSON.stringify({ type: 'response.cancel' }))
-    flushAssistantMessage()
-    setConversationState('listening')
-  }, [flushAssistantMessage])
-
-  const clearMessages = useCallback(() => {
-    setMessages([])
-  }, [])
 
   return {
-    connectionState,
+    connectionState: snapshot.connectionState,
-    conversationState,
+    conversationState: snapshot.conversationState,
-    messages,
+    messages: snapshot.messages,
-    isActive: connectionState === 'connecting' || connectionState === 'live',
+    isActive: snapshot.connectionState === 'connecting' || snapshot.connectionState === 'live',
-    streamingText,
+    streamingText: snapshot.streamingText,
-    streamingMsgId,
+    streamingMsgId: snapshot.streamingMsgId,
-    error,
+    error: snapshot.error,
-    start,
+    start: controller.start,
-    stop,
+    stop: controller.stop,
-    sendText,
+    sendText: controller.sendText,
-    cancelResponse,
+    cancelResponse: controller.cancelResponse,
-    clearMessages,
+    clearMessages: controller.clearMessages,
   }
 }

apps/desktop/src/renderer/pages/VoiceConversationPage.tsx

@@ -124,8 +124,9 @@ export function VoiceConversationPage(): React.ReactElement {
     isStartingSessionRef.current = true
     try {
       if (isRealtime) {
-        const ok = await realtime.start()
+        // 'cancelled'(시작 중 정지)는 fallback 대상이 아니다
-        if (!ok) {
+        const result = await realtime.start()
+        if (result === 'failed') {
           setBackend('local')
           setErrorBanner({ phase: 'llm', message: t('conversation.realtime.fallback') })
           await window.electronAPI.voiceConversation.startSession()
@@ -159,15 +160,21 @@ export function VoiceConversationPage(): React.ReactElement {
     try {
       if (isRealtime) {
         if (!realtime.isActive) {
-          const ok = await realtime.start()
+          const result = await realtime.start()
-          if (!ok) {
+          if (result === 'failed') {
             setBackend('local')
             setErrorBanner({ phase: 'llm', message: t('conversation.realtime.fallback') })
             await window.electronAPI.voiceConversation.startSession()
             await window.electronAPI.voiceConversation.sendMessage({ text })
             return
           }
+          if (result === 'cancelled') {
+            // 시작 중 사용자가 멈췄다 — 보내지 못한 입력을 되돌려 준다
+            setTextInput((current) => current || text)
+            return
+          }
         }
+        // start()는 이벤트 채널 open 후 resolve하고, 연결 중이면 컨트롤러가 큐에 쌓는다
         realtime.sendText(text)
         return
       }

apps/desktop/src/renderer/services/realtimeConversationModel.ts

@@ -0,0 +1,139 @@
+// src/renderer/services/realtimeConversationModel.ts
+// OpenAI Realtime 서버 이벤트 → 대화 UI 상태 매핑 (순수 함수).
+// IO·React에 의존하지 않으므로 node 환경에서 그대로 테스트할 수 있다.
+//
+// 이벤트 매핑 (data channel 'oai-events'):
+//   response.created                              → thinking
+//   response.(output_)audio_transcript.delta      → speaking + 스트리밍 텍스트
+//   conversation.item.input_audio_transcription.completed → 유저 메시지
+//   response.done                                 → 어시스턴트 메시지 확정 + listening 복귀
+//   error                                         → error 메시지
+
+import type { ConversationState, ConversationMessage } from '@d3ro/core/types'
+
+export interface RealtimeServerEvent {
+  type: string
+  delta?: string
+  transcript?: string
+  response?: { id?: string }
+  item_id?: string
+  error?: { message?: string }
+}
+
+export interface RealtimeConversationModel {
+  conversationState: ConversationState
+  messages: ConversationMessage[]
+  streamingText: string
+  streamingMsgId: string | null
+  error: string | null
+  /** 진행 중인 어시스턴트 응답 누적 텍스트 */
+  assistantBuffer: string
+  /** 진행 중인 어시스턴트 응답의 메시지 id (response.created에서 발급) */
+  assistantMsgId: string | null
+}
+
+export interface RealtimeModelContext {
+  nextId: (prefix: 'user' | 'assistant') => string
+  now: () => number
+}
+
+export const INITIAL_REALTIME_MODEL: RealtimeConversationModel = {
+  conversationState: 'idle',
+  messages: [],
+  streamingText: '',
+  streamingMsgId: null,
+  error: null,
+  assistantBuffer: '',
+  assistantMsgId: null,
+}
+
+let messageSeq = 0
+export const defaultRealtimeModelContext: RealtimeModelContext = {
+  nextId: (prefix) => {
+    messageSeq += 1
+    return `rt-${prefix}-${Date.now()}-${messageSeq}`
+  },
+  now: () => Date.now(),
+}
+
+/** 진행 중인 어시스턴트 응답을 메시지로 확정하고 스트리밍 상태를 비운다. */
+export function flushAssistantMessage(
+  model: RealtimeConversationModel,
+  ctx: RealtimeModelContext,
+): RealtimeConversationModel {
+  const content = model.assistantBuffer.trim()
+  const msgId = model.assistantMsgId
+  const messages =
+    content && msgId
+      ? [...model.messages, { id: msgId, role: 'assistant' as const, content, timestamp: ctx.now() }]
+      : model.messages
+  return {
+    ...model,
+    messages,
+    assistantBuffer: '',
+    assistantMsgId: null,
+    streamingText: '',
+    streamingMsgId: null,
+  }
+}
+
+export function appendUserMessage(
+  model: RealtimeConversationModel,
+  content: string,
+  ctx: RealtimeModelContext,
+): RealtimeConversationModel {
+  return {
+    ...model,
+    messages: [
+      ...model.messages,
+      { id: ctx.nextId('user'), role: 'user', content, timestamp: ctx.now() },
+    ],
+  }
+}
+
+/** 서버 이벤트 하나를 적용한다. 관심 없는 이벤트면 같은 참조를 돌려준다. */
+export function reduceRealtimeServerEvent(
+  model: RealtimeConversationModel,
+  event: RealtimeServerEvent,
+  ctx: RealtimeModelContext,
+): RealtimeConversationModel {
+  switch (event.type) {
+    case 'response.created':
+      return {
+        ...model,
+        assistantBuffer: '',
+        assistantMsgId: ctx.nextId('assistant'),
+        conversationState: 'thinking',
+      }
+
+    // GA/신규 이벤트명 모두 수용
+    case 'response.output_audio_transcript.delta':
+    case 'response.audio_transcript.delta':
+    case 'response.output_text.delta':
+    case 'response.text.delta': {
+      if (!event.delta) return model
+      const assistantBuffer = model.assistantBuffer + event.delta
+      return {
+        ...model,
+        assistantBuffer,
+        streamingMsgId: model.assistantMsgId,
+        streamingText: assistantBuffer,
+        conversationState: 'speaking',
+      }
+    }
+
+    case 'conversation.item.input_audio_transcription.completed': {
+      const transcript = (event.transcript ?? '').trim()
+      return transcript ? appendUserMessage(model, transcript, ctx) : model
+    }
+
+    case 'response.done':
+      return { ...flushAssistantMessage(model, ctx), conversationState: 'listening' }
+
+    case 'error':
+      return { ...model, error: event.error?.message ?? 'Realtime error' }
+
+    default:
+      return model
+  }
+}

apps/desktop/src/renderer/services/realtimeSession.ts

@@ -0,0 +1,308 @@
+// src/renderer/services/realtimeSession.ts
+// OpenAI Realtime 세션 전송 계층.
+//
+// - RealtimeTransport: 컨트롤러/훅이 의존하는 포트 (연결·송신·종료·이벤트 구독).
+// - WebRtcRealtimeTransport: WebRTC 구현. 토큰 → 마이크 → PeerConnection → SDP 교환 →
+//   data channel 'open'까지를 취소 가능한 connect(signal) 하나로 묶는다.
+//   각 await 뒤에 취소 여부를 확인해, 취소되면 그때까지 얻은 마이크·PC를 즉시 해제한다.
+// - IO 세부(IPC 토큰, getUserMedia, RTCPeerConnection, fetch, <audio>)는 전부 주입받는다.
+//   createBrowserRealtimeTransport()가 실제 브라우저/Electron 의존성을 묶는 기본 조립점이다.
+
+import type { RealtimeServerEvent } from './realtimeConversationModel'
+
+export const REALTIME_CALLS_URL = 'https://api.openai.com/v1/realtime/calls'
+export const REALTIME_DATA_CHANNEL_NAME = 'oai-events'
+
+/** 클라이언트 → 서버 이벤트 (JSON 직렬화 가능한 객체) */
+export interface RealtimeClientEvent {
+  type: string
+  [key: string]: unknown
+}
+
+/** 컨트롤러가 의존하는 전송 포트. 인스턴스는 1회용이다 (connect는 한 번만). */
+export interface RealtimeTransport {
+  /**
+   * 연결을 수립하고 이벤트 채널이 열리면 resolve한다.
+   * signal이 abort되면 잡고 있던 자원을 모두 해제하고 RealtimeAbortError로 reject한다.
+   */
+  connect(signal: AbortSignal): Promise<void>
+  /** 채널이 열려 있으면 전송하고 true, 아니면 false */
+  send(event: RealtimeClientEvent): boolean
+  /** 자원 해제 (멱등). 호출자가 직접 닫은 경우 onClosed는 불리지 않는다. */
+  close(): void
+  onEvent(listener: (event: RealtimeServerEvent) => void): () => void
+  /** 연결이 열린 뒤 원격/네트워크 사유로 끊겼을 때 한 번 호출된다. */
+  onClosed(listener: () => void): () => void
+}
+
+export type RealtimeTransportFactory = () => RealtimeTransport
+
+export class RealtimeAbortError extends Error {
+  constructor() {
+    super('realtime connect aborted')
+    this.name = 'RealtimeAbortError'
+  }
+}
+
+export function isRealtimeAbortError(err: unknown): err is RealtimeAbortError {
+  return err instanceof RealtimeAbortError
+}
+
+// ── WebRTC 구현 ────────────────────────────────────────────────
+
+export interface RealtimeToken {
+  value: string
+  model: string
+}
+
+export interface SdpHttpResponse {
+  ok: boolean
+  status: number
+  text(): Promise<string>
+}
+
+/** 원격 오디오 재생 대상 */
+export interface RealtimeAudioSink {
+  setStream(stream: MediaStream | null): void
+  dispose(): void
+}
+
+export interface WebRtcTransportDeps {
+  getToken(): Promise<RealtimeToken>
+  getUserMedia(): Promise<MediaStream>
+  createPeerConnection(): RTCPeerConnection
+  postSdp(url: string, init: { body: string; headers: Record<string, string> }): Promise<SdpHttpResponse>
+  createAudioSink(): RealtimeAudioSink
+  callsUrl?: string
+}
+
+type Phase = 'new' | 'connecting' | 'open' | 'closed'
+
+const TERMINAL_PC_STATES: ReadonlySet<RTCPeerConnectionState> = new Set(['failed', 'closed'])
+
+export class WebRtcRealtimeTransport implements RealtimeTransport {
+  private phase: Phase = 'new'
+  private stream: MediaStream | null = null
+  private pc: RTCPeerConnection | null = null
+  private dc: RTCDataChannel | null = null
+  private sink: RealtimeAudioSink | null = null
+  private pendingOpen: { resolve: () => void; reject: (err: Error) => void } | null = null
+  private readonly eventListeners = new Set<(event: RealtimeServerEvent) => void>()
+  private readonly closedListeners = new Set<() => void>()
+
+  constructor(private readonly deps: WebRtcTransportDeps) {}
+
+  async connect(signal: AbortSignal): Promise<void> {
+    if (this.phase !== 'new') throw new Error('realtime transport is single-use')
+    this.phase = 'connecting'
+
+    const onAbort = (): void => this.close()
+    signal.addEventListener('abort', onAbort, { once: true })
+    try {
+      this.ensureActive(signal)
+
+      // 1) ephemeral token
+      const { value: ephemeralKey, model } = await this.deps.getToken()
+      this.ensureActive(signal)
+
+      // 2) 마이크 — 먼저 필드에 담아야 취소 시 close()가 트랙을 멈춘다
+      this.stream = await this.deps.getUserMedia()
+      this.ensureActive(signal)
+
+      // 3) peer connection + 원격 오디오
+      const pc = this.deps.createPeerConnection()
+      this.pc = pc
+      const micTrack = this.stream.getAudioTracks()[0]
+      if (micTrack) pc.addTrack(micTrack, this.stream)
+
+      const sink = this.deps.createAudioSink()
+      this.sink = sink
+      pc.ontrack = (e: RTCTrackEvent): void => {
+        sink.setStream(e.streams[0] ?? null)
+      }
+      pc.onconnectionstatechange = (): void => {
+        if (TERMINAL_PC_STATES.has(pc.connectionState)) {
+          this.handleRemoteClose(`peer connection ${pc.connectionState}`)
+        }
+      }
+
+      // 4) 이벤트 데이터 채널
+      const dc = pc.createDataChannel(REALTIME_DATA_CHANNEL_NAME)
+      this.dc = dc
+      dc.onmessage = (e: MessageEvent): void => this.handleMessage(e.data)
+      dc.onopen = (): void => this.handleOpen()
+      dc.onclose = (): void => this.handleRemoteClose('data channel closed')
+
+      // 5) SDP 교환
+      const offer = await pc.createOffer()
+      this.ensureActive(signal)
+      await pc.setLocalDescription(offer)
+      this.ensureActive(signal)
+
+      const url = `${this.deps.callsUrl ?? REALTIME_CALLS_URL}?model=${encodeURIComponent(model)}`
+      const sdpResponse = await this.deps.postSdp(url, {
+        body: offer.sdp ?? '',
+        headers: {
+          Authorization: `Bearer ${ephemeralKey}`,
+          'Content-Type': 'application/sdp',
+        },
+      })
+      this.ensureActive(signal)
+      const answerSdp = await sdpResponse.text()
+      this.ensureActive(signal)
+      if (!sdpResponse.ok) {
+        throw new Error(`SDP exchange failed (${sdpResponse.status}): ${answerSdp.slice(0, 200)}`)
+      }
+
+      await pc.setRemoteDescription({ type: 'answer', sdp: answerSdp })
+      this.ensureActive(signal)
+
+      // 6) 이벤트 채널 open까지 대기 — 이후 send가 버려지지 않도록
+      await this.waitForOpen()
+    } catch (err) {
+      this.close()
+      throw err
+    } finally {
+      signal.removeEventListener('abort', onAbort)
+    }
+  }
+
+  send(event: RealtimeClientEvent): boolean {
+    const dc = this.dc
+    if (this.phase !== 'open' || !dc || dc.readyState !== 'open') return false
+    dc.send(JSON.stringify(event))
+    return true
+  }
+
+  close(): void {
+    this.phase = 'closed'
+    this.release(new RealtimeAbortError())
+  }
+
+  onEvent(listener: (event: RealtimeServerEvent) => void): () => void {
+    this.eventListeners.add(listener)
+    return () => this.eventListeners.delete(listener)
+  }
+
+  onClosed(listener: () => void): () => void {
+    this.closedListeners.add(listener)
+    return () => this.closedListeners.delete(listener)
+  }
+
+  // ── 내부 ──
+
+  private ensureActive(signal: AbortSignal): void {
+    if (signal.aborted || this.phase === 'closed') {
+      this.close()
+      throw new RealtimeAbortError()
+    }
+  }
+
+  private waitForOpen(): Promise<void> {
+    if (this.phase === 'open') return Promise.resolve()
+    if (this.dc?.readyState === 'open') {
+      this.phase = 'open'
+      return Promise.resolve()
+    }
+    return new Promise<void>((resolve, reject) => {
+      this.pendingOpen = { resolve, reject }
+    })
+  }
+
+  private handleOpen(): void {
+    if (this.phase !== 'connecting') return
+    this.phase = 'open'
+    const pending = this.pendingOpen
+    this.pendingOpen = null
+    pending?.resolve()
+  }
+
+  private handleMessage(data: unknown): void {
+    if (this.phase === 'closed' || typeof data !== 'string') return
+    let parsed: unknown
+    try {
+      parsed = JSON.parse(data)
+    } catch {
+      return // 파싱 불가 이벤트 무시
+    }
+    if (!parsed || typeof parsed !== 'object') return
+    const event = parsed as RealtimeServerEvent
+    if (typeof event.type !== 'string') return
+    for (const listener of this.eventListeners) listener(event)
+  }
+
+  /** 원격/네트워크 사유 종료: 자원을 스스로 해제하고, 열린 뒤였다면 onClosed를 한 번 알린다. */
+  private handleRemoteClose(reason: string): void {
+    if (this.phase === 'closed' || this.phase === 'new') return
+    const wasOpen = this.phase === 'open'
+    this.phase = 'closed'
+    this.release(new Error(`Realtime connection lost: ${reason}`))
+    if (wasOpen) {
+      for (const listener of this.closedListeners) listener()
+    }
+  }
+
+  private release(pendingReason: Error): void {
+    const pending = this.pendingOpen
+    this.pendingOpen = null
+
+    const dc = this.dc
+    this.dc = null
+    if (dc) {
+      dc.onopen = null
+      dc.onclose = null
+      dc.onmessage = null
+      dc.close()
+    }
+    const pc = this.pc
+    this.pc = null
+    if (pc) {
+      pc.ontrack = null
+      pc.onconnectionstatechange = null
+      pc.close()
+    }
+    this.stream?.getTracks().forEach((track) => track.stop())
+    this.stream = null
+    if (this.sink) {
+      this.sink.setStream(null)
+      this.sink.dispose()
+      this.sink = null
+    }
+
+    pending?.reject(pendingReason)
+  }
+}
+
+// ── 기본 조립 (브라우저/Electron 렌더러) ─────────────────────────
+
+function createBrowserAudioSink(): RealtimeAudioSink {
+  const audioEl = document.createElement('audio')
+  audioEl.autoplay = true
+  return {
+    setStream: (stream) => {
+      audioEl.srcObject = stream
+    },
+    dispose: () => {
+      audioEl.srcObject = null
+      audioEl.remove()
+    },
+  }
+}
+
+export const browserWebRtcDeps: WebRtcTransportDeps = {
+  getToken: async () => {
+    // Supabase realtime-token → OpenAI client_secrets (IPC 경유)
+    const result = await window.electronAPI.voiceConversation.getRealtimeToken({})
+    if (!result.success) {
+      throw new Error(result.error?.message ?? 'token request failed')
+    }
+    return { value: result.data.value, model: result.data.model }
+  },
+  getUserMedia: () => navigator.mediaDevices.getUserMedia({ audio: true }),
+  createPeerConnection: () => new RTCPeerConnection(),
+  postSdp: (url, init) => fetch(url, { method: 'POST', body: init.body, headers: init.headers }),
+  createAudioSink: createBrowserAudioSink,
+}
+
+export const createBrowserRealtimeTransport: RealtimeTransportFactory = () =>
+  new WebRtcRealtimeTransport(browserWebRtcDeps)

apps/desktop/src/renderer/services/realtimeSessionController.ts

@@ -0,0 +1,217 @@
+// src/renderer/services/realtimeSessionController.ts
+// Realtime 대화 세션 수명주기 컨트롤러 (React 비의존).
+//
+// - start/stop/언마운트 경쟁: 시도마다 AbortController를 두고, stop이 오면 abort + close.
+//   취소된 시도는 'cancelled'로 끝나며 연결·마이크가 되살아나지 않는다.
+// - 동시 start는 진행 중인 시도 하나로 합친다.
+// - 연결이 끊기면(transport.onClosed) 전송 계층을 버리고 'error'로 전환해 재시작을 허용한다.
+// - 연결 중 sendText는 큐에 쌓았다가 채널이 열리면 순서대로 보낸다.
+// - 상태는 불변 스냅샷으로 노출해 useSyncExternalStore로 구독할 수 있다.
+
+import {
+  INITIAL_REALTIME_MODEL,
+  appendUserMessage,
+  defaultRealtimeModelContext,
+  flushAssistantMessage,
+  reduceRealtimeServerEvent,
+  type RealtimeConversationModel,
+  type RealtimeModelContext,
+  type RealtimeServerEvent,
+} from './realtimeConversationModel'
+import type { RealtimeClientEvent, RealtimeTransport, RealtimeTransportFactory } from './realtimeSession'
+
+export type RealtimeConnectionState = 'idle' | 'connecting' | 'live' | 'error'
+
+/** 'failed'일 때만 로컬 파이프라인 fallback을 유도한다. 'cancelled'는 사용자가 멈춘 경우. */
+export type RealtimeStartResult = 'live' | 'failed' | 'cancelled'
+
+export interface RealtimeSessionSnapshot extends RealtimeConversationModel {
+  connectionState: RealtimeConnectionState
+}
+
+/** 채널이 열리면 보내는 세션 설정 — 유저 발화 전사 활성화 (미지원 모델이면 error 이벤트만 오고 대화는 유지됨) */
+export const REALTIME_SESSION_UPDATE: RealtimeClientEvent = {
+  type: 'session.update',
+  session: {
+    type: 'realtime',
+    audio: {
+      input: { transcription: { model: 'whisper-1' } },
+    },
+  },
+}
+
+function userTextEvents(text: string): RealtimeClientEvent[] {
+  return [
+    {
+      type: 'conversation.item.create',
+      item: {
+        type: 'message',
+        role: 'user',
+        content: [{ type: 'input_text', text }],
+      },
+    },
+    { type: 'response.create' },
+  ]
+}
+
+export class RealtimeSessionController {
+  private snapshot: RealtimeSessionSnapshot = { ...INITIAL_REALTIME_MODEL, connectionState: 'idle' }
+  private readonly listeners = new Set<() => void>()
+  private transport: RealtimeTransport | null = null
+  private attempt: AbortController | null = null
+  private pendingStart: Promise<RealtimeStartResult> | null = null
+  private outbox: RealtimeClientEvent[] = []
+
+  constructor(
+    private readonly createTransport: RealtimeTransportFactory,
+    private readonly ctx: RealtimeModelContext = defaultRealtimeModelContext,
+  ) {}
+
+  // ── 구독 (useSyncExternalStore 계약) ──
+
+  subscribe = (listener: () => void): (() => void) => {
+    this.listeners.add(listener)
+    return () => {
+      this.listeners.delete(listener)
+    }
+  }
+
+  getSnapshot = (): RealtimeSessionSnapshot => this.snapshot
+
+  // ── 명령 ──
+
+  start = (): Promise<RealtimeStartResult> => {
+    if (this.snapshot.connectionState === 'live' && this.transport) return Promise.resolve('live')
+    if (this.pendingStart) return this.pendingStart
+
+    const attempt = new AbortController()
+    const transport = this.createTransport()
+    this.attempt = attempt
+    this.transport = transport
+    this.outbox = []
+    transport.onEvent((event) => {
+      if (this.transport === transport) this.applyServerEvent(event)
+    })
+    // 채널이 열리자마자 끊겨 connect가 아직 반환 전이면, 그 시도를 실패로 마무리한다
+    let lostWhileConnecting = false
+    transport.onClosed(() => {
+      if (this.transport !== transport) return
+      if (this.snapshot.connectionState === 'connecting') lostWhileConnecting = true
+      else this.handleConnectionLost()
+    })
+    this.update({ connectionState: 'connecting', error: null })
+
+    const run = async (): Promise<RealtimeStartResult> => {
+      const isCurrent = (): boolean => !attempt.signal.aborted && this.transport === transport
+      const fail = (message: string): RealtimeStartResult => {
+        this.transport = null
+        this.attempt = null
+        this.outbox = []
+        this.update({
+          ...flushAssistantMessage(this.snapshot, this.ctx),
+          connectionState: 'error',
+          conversationState: 'idle',
+          error: message,
+        })
+        return 'failed'
+      }
+      try {
+        await transport.connect(attempt.signal)
+      } catch (err) {
+        transport.close()
+        if (!isCurrent()) return 'cancelled'
+        return fail(err instanceof Error ? err.message : String(err))
+      }
+      if (!isCurrent()) {
+        transport.close()
+        return 'cancelled'
+      }
+      if (lostWhileConnecting) {
+        transport.close()
+        return fail('Realtime connection lost')
+      }
+
+      this.attempt = null
+      transport.send(REALTIME_SESSION_UPDATE)
+      const queued = this.outbox
+      this.outbox = []
+      for (const event of queued) transport.send(event)
+      this.update({ connectionState: 'live', conversationState: 'listening' })
+      return 'live'
+    }
+
+    const pending = run().finally(() => {
+      if (this.pendingStart === pending) this.pendingStart = null
+    })
+    this.pendingStart = pending
+    return pending
+  }
+
+  stop = (): void => {
+    this.attempt?.abort()
+    this.attempt = null
+    const transport = this.transport
+    this.transport = null
+    this.pendingStart = null
+    this.outbox = []
+    transport?.close()
+    this.update({
+      ...flushAssistantMessage(this.snapshot, this.ctx),
+      connectionState: 'idle',
+      conversationState: 'idle',
+    })
+  }
+
+  sendText = (text: string): void => {
+    const { connectionState } = this.snapshot
+    const transport = this.transport
+    if (!transport) return
+
+    if (connectionState === 'connecting') {
+      // 채널이 열리면 start()가 순서대로 보낸다
+      this.outbox.push(...userTextEvents(text))
+      this.update(appendUserMessage(this.snapshot, text, this.ctx))
+      return
+    }
+    if (connectionState !== 'live') return
+
+    const [itemCreate, responseCreate] = userTextEvents(text)
+    if (!transport.send(itemCreate)) return
+    transport.send(responseCreate)
+    this.update(appendUserMessage(this.snapshot, text, this.ctx))
+  }
+
+  cancelResponse = (): void => {
+    const transport = this.transport
+    if (!transport || this.snapshot.connectionState !== 'live') return
+    if (!transport.send({ type: 'response.cancel' })) return
+    this.update({ ...flushAssistantMessage(this.snapshot, this.ctx), conversationState: 'listening' })
+  }
+
+  clearMessages = (): void => {
+    this.update({ messages: [] })
+  }
+
+  // ── 내부 ──
+
+  private applyServerEvent(event: RealtimeServerEvent): void {
+    const next = reduceRealtimeServerEvent(this.snapshot, event, this.ctx)
+    if (next !== this.snapshot) this.update(next)
+  }
+
+  private handleConnectionLost(): void {
+    this.transport = null
+    this.attempt = null
+    this.outbox = []
+    this.update({
+      ...flushAssistantMessage(this.snapshot, this.ctx),
+      connectionState: 'error',
+      conversationState: 'idle',
+    })
+  }
+
+  private update(patch: Partial<RealtimeSessionSnapshot>): void {
+    this.snapshot = { ...this.snapshot, ...patch }
+    for (const listener of this.listeners) listener()
+  }
+}

apps/desktop/tests/unit/realtime-session-redteam-r1-30.test.ts

@@ -0,0 +1,529 @@
+// tests/unit/realtime-session-redteam-r1-30.test.ts
+// Realtime 대화 세션: 시작 중 취소 누수, 끊김 후 재연결, open 전 첫 메시지 유실 회귀 테스트.
+// node 환경에서 fake 전송 계층/WebRTC 의존성으로 검증한다.
+
+import { describe, it, expect, vi } from 'vitest'
+import {
+  INITIAL_REALTIME_MODEL,
+  reduceRealtimeServerEvent,
+  type RealtimeModelContext,
+  type RealtimeServerEvent,
+} from '../../src/renderer/services/realtimeConversationModel'
+import {
+  RealtimeAbortError,
+  WebRtcRealtimeTransport,
+  type RealtimeClientEvent,
+  type RealtimeTransport,
+  type WebRtcTransportDeps,
+} from '../../src/renderer/services/realtimeSession'
+import {
+  REALTIME_SESSION_UPDATE,
+  RealtimeSessionController,
+} from '../../src/renderer/services/realtimeSessionController'
+
+// ── helpers ──
+
+interface Deferred<T> {
+  promise: Promise<T>
+  resolve: (value: T) => void
+  reject: (err: Error) => void
+}
+
+function deferred<T>(): Deferred<T> {
+  let resolve!: (value: T) => void
+  let reject!: (err: Error) => void
+  const promise = new Promise<T>((res, rej) => {
+    resolve = res
+    reject = rej
+  })
+  return { promise, resolve, reject }
+}
+
+const flush = async (): Promise<void> => {
+  for (let i = 0; i < 10; i++) await Promise.resolve()
+}
+
+function makeCtx(): RealtimeModelContext {
+  let seq = 0
+  return {
+    nextId: (prefix) => `${prefix}-${++seq}`,
+    now: () => 1000,
+  }
+}
+
+// ── 순수 reducer ──
+
+describe('reduceRealtimeServerEvent', () => {
+  it('maps a full assistant turn to thinking → speaking → message + listening', () => {
+    const ctx = makeCtx()
+    let m = reduceRealtimeServerEvent(INITIAL_REALTIME_MODEL, { type: 'response.created' }, ctx)
+    expect(m.conversationState).toBe('thinking')
+    expect(m.assistantMsgId).toBe('assistant-1')
+
+    m = reduceRealtimeServerEvent(m, { type: 'response.output_audio_transcript.delta', delta: '안녕' }, ctx)
+    m = reduceRealtimeServerEvent(m, { type: 'response.audio_transcript.delta', delta: '하세요' }, ctx)
+    expect(m.conversationState).toBe('speaking')
+    expect(m.streamingText).toBe('안녕하세요')
+    expect(m.streamingMsgId).toBe('assistant-1')
+
+    m = reduceRealtimeServerEvent(m, { type: 'response.done' }, ctx)
+    expect(m.conversationState).toBe('listening')
+    expect(m.streamingText).toBe('')
+    expect(m.streamingMsgId).toBeNull()
+    expect(m.messages).toEqual([
+      { id: 'assistant-1', role: 'assistant', content: '안녕하세요', timestamp: 1000 },
+    ])
+  })
+
+  it('adds trimmed user transcripts and ignores empty ones', () => {
+    const ctx = makeCtx()
+    const ev = (transcript: string): RealtimeServerEvent => ({
+      type: 'conversation.item.input_audio_transcription.completed',
+      transcript,
+    })
+    let m = reduceRealtimeServerEvent(INITIAL_REALTIME_MODEL, ev('  hi  '), ctx)
+    expect(m.messages).toEqual([{ id: 'user-1', role: 'user', content: 'hi', timestamp: 1000 }])
+    const same = reduceRealtimeServerEvent(m, ev('   '), ctx)
+    expect(same).toBe(m)
+    m = reduceRealtimeServerEvent(m, { type: 'error', error: { message: 'boom' } }, ctx)
+    expect(m.error).toBe('boom')
+    expect(reduceRealtimeServerEvent(m, { type: 'error' }, ctx).error).toBe('Realtime error')
+  })
+
+  it('returns the same reference for unknown events and empty deltas', () => {
+    const ctx = makeCtx()
+    expect(reduceRealtimeServerEvent(INITIAL_REALTIME_MODEL, { type: 'session.created' }, ctx)).toBe(
+      INITIAL_REALTIME_MODEL,
+    )
+    expect(
+      reduceRealtimeServerEvent(INITIAL_REALTIME_MODEL, { type: 'response.text.delta', delta: '' }, ctx),
+    ).toBe(INITIAL_REALTIME_MODEL)
+  })
+})
+
+// ── WebRTC transport (fake 의존성) ──
+
+class FakeTrack {
+  stop = vi.fn()
+}
+
+class FakeStream {
+  readonly track = new FakeTrack()
+  getTracks = (): FakeTrack[] => [this.track]
+  getAudioTracks = (): FakeTrack[] => [this.track]
+}
+
+class FakeDataChannel {
+  readyState = 'connecting'
+  onopen: (() => void) | null = null
+  onclose: (() => void) | null = null
+  onmessage: ((e: { data: unknown }) => void) | null = null
+  send = vi.fn()
+  close = vi.fn(() => {
+    this.readyState = 'closed'
+  })
+  open(): void {
+    this.readyState = 'open'
+    this.onopen?.()
+  }
+  receive(data: unknown): void {
+    this.onmessage?.({ data })
+  }
+}
+
+class FakePeerConnection {
+  connectionState = 'new'
+  ontrack: ((e: { streams: unknown[] }) => void) | null = null
+  onconnectionstatechange: (() => void) | null = null
+  readonly dc = new FakeDataChannel()
+  readonly remote = deferred<void>()
+  addTrack = vi.fn()
+  createDataChannel = vi.fn(() => this.dc)
+  createOffer = vi.fn(async () => ({ type: 'offer', sdp: 'offer-sdp' }))
+  setLocalDescription = vi.fn(async () => undefined)
+  setRemoteDescription = vi.fn(() => this.remote.promise)
+  close = vi.fn()
+  setState(state: string): void {
+    this.connectionState = state
+    this.onconnectionstatechange?.()
+  }
+}
+
+interface TransportHarness {
+  transport: WebRtcRealtimeTransport
+  token: Deferred<{ value: string; model: string }>
+  media: Deferred<FakeStream>
+  stream: FakeStream
+  pcs: FakePeerConnection[]
+  sink: { setStream: ReturnType<typeof vi.fn>; dispose: ReturnType<typeof vi.fn> }
+  deps: {
+    getUserMedia: ReturnType<typeof vi.fn>
+    createPeerConnection: ReturnType<typeof vi.fn>
+    postSdp: ReturnType<typeof vi.fn>
+  }
+}
+
+function makeTransport(sdpOk = true): TransportHarness {
+  const token = deferred<{ value: string; model: string }>()
+  const media = deferred<FakeStream>()
+  const stream = new FakeStream()
+  const pcs: FakePeerConnection[] = []
+  const sink = { setStream: vi.fn(), dispose: vi.fn() }
+  const getUserMedia = vi.fn(() => media.promise)
+  const createPeerConnection = vi.fn(() => {
+    const pc = new FakePeerConnection()
+    pcs.push(pc)
+    return pc
+  })
+  const postSdp = vi.fn(async () => ({
+    ok: sdpOk,
+    status: sdpOk ? 201 : 401,
+    text: async () => (sdpOk ? 'answer-sdp' : 'unauthorized'),
+  }))
+  const deps = {
+    getToken: () => token.promise,
+    getUserMedia,
+    createPeerConnection,
+    postSdp,
+    createAudioSink: () => sink,
+    callsUrl: 'https://example.test/calls',
+  } as unknown as WebRtcTransportDeps
+  return {
+    transport: new WebRtcRealtimeTransport(deps),
+    token,
+    media,
+    stream,
+    pcs,
+    sink,
+    deps: { getUserMedia, createPeerConnection, postSdp },
+  }
+}
+
+async function driveToRemoteDescription(h: TransportHarness): Promise<FakePeerConnection> {
+  h.token.resolve({ value: 'ek_1', model: 'gpt-realtime' })
+  await flush()
+  h.media.resolve(h.stream)
+  await flush()
+  const pc = h.pcs[0]
+  expect(pc).toBeDefined()
+  return pc
+}
+
+describe('WebRtcRealtimeTransport', () => {
+  it('resolves connect only after the data channel opens', async () => {
+    const h = makeTransport()
+    const ac = new AbortController()
+    let settled = false
+    const p = h.transport.connect(ac.signal).then(() => {
+      settled = true
+    })
+    const pc = await driveToRemoteDescription(h)
+    expect(h.deps.postSdp).toHaveBeenCalledWith(
+      'https://example.test/calls?model=gpt-realtime',
+      expect.objectContaining({ body: 'offer-sdp' }),
+    )
+    pc.remote.resolve()
+    await flush()
+    expect(settled).toBe(false)
+    expect(h.transport.send({ type: 'x' })).toBe(false)
+
+    pc.dc.open()
+    await p
+    expect(settled).toBe(true)
+    expect(h.transport.send({ type: 'x' })).toBe(true)
+    expect(pc.dc.send).toHaveBeenCalledWith(JSON.stringify({ type: 'x' }))
+  })
+
+  it('abort during the token request never opens the microphone', async () => {
+    const h = makeTransport()
+    const ac = new AbortController()
+    const p = h.transport.connect(ac.signal)
+    ac.abort()
+    h.token.resolve({ value: 'ek', model: 'm' })
+    await expect(p).rejects.toBeInstanceOf(RealtimeAbortError)
+    expect(h.deps.getUserMedia).not.toHaveBeenCalled()
+    expect(h.deps.createPeerConnection).not.toHaveBeenCalled()
+  })
+
+  it('abort during getUserMedia stops the late microphone stream and creates no peer connection', async () => {
+    const h = makeTransport()
+    const ac = new AbortController()
+    const p = h.transport.connect(ac.signal)
+    h.token.resolve({ value: 'ek', model: 'm' })
+    await flush()
+    expect(h.deps.getUserMedia).toHaveBeenCalledTimes(1)
+    ac.abort()
+    h.media.resolve(h.stream)
+    await expect(p).rejects.toBeInstanceOf(RealtimeAbortError)
+    expect(h.stream.track.stop).toHaveBeenCalled()
+    expect(h.deps.createPeerConnection).not.toHaveBeenCalled()
+  })
+
+  it('abort while waiting for the channel releases pc, channel, mic and audio sink', async () => {
+    const h = makeTransport()
+    const ac = new AbortController()
+    const p = h.transport.connect(ac.signal)
+    const pc = await driveToRemoteDescription(h)
+    pc.remote.resolve()
+    await flush()
+    ac.abort()
+    await expect(p).rejects.toBeInstanceOf(RealtimeAbortError)
+    expect(pc.close).toHaveBeenCalled()
+    expect(pc.dc.close).toHaveBeenCalled()
+    expect(h.stream.track.stop).toHaveBeenCalled()
+    expect(h.sink.dispose).toHaveBeenCalled()
+  })
+
+  it('a failed SDP exchange rejects with the status and releases resources', async () => {
+    const h = makeTransport(false)
+    const p = h.transport.connect(new AbortController().signal)
+    const pc = await driveToRemoteDescription(h)
+    await expect(p).rejects.toThrow('SDP exchange failed (401): unauthorized')
+    expect(pc.close).toHaveBeenCalled()
+    expect(h.stream.track.stop).toHaveBeenCalled()
+  })
+
+  it('a peer connection failure before open rejects connect', async () => {
+    const h = makeTransport()
+    const p = h.transport.connect(new AbortController().signal)
+    const pc = await driveToRemoteDescription(h)
+    pc.remote.resolve()
+    await flush()
+    pc.setState('failed')
+    await expect(p).rejects.toThrow('Realtime connection lost')
+    expect(h.stream.track.stop).toHaveBeenCalled()
+  })
+
+  it('after open, a failed connection closes itself and notifies onClosed exactly once', async () => {
+    const h = makeTransport()
+    const onClosed = vi.fn()
+    h.transport.onClosed(onClosed)
+    const p = h.transport.connect(new AbortController().signal)
+    const pc = await driveToRemoteDescription(h)
+    pc.remote.resolve()
+    await flush()
+    pc.dc.open()
+    await p
+
+    pc.setState('disconnected') // 일시적 끊김은 종료로 보지 않는다
+    expect(onClosed).not.toHaveBeenCalled()
+
+    pc.setState('failed')
+    pc.setState('closed')
+    expect(onClosed).toHaveBeenCalledTimes(1)
+    expect(pc.close).toHaveBeenCalled()
+    expect(h.stream.track.stop).toHaveBeenCalled()
+    expect(h.transport.send({ type: 'x' })).toBe(false)
+  })
+
+  it('explicit close does not report onClosed, and forwards only valid JSON events', async () => {
+    const h = makeTransport()
+    const onClosed = vi.fn()
+    const onEvent = vi.fn()
+    h.transport.onClosed(onClosed)
+    h.transport.onEvent(onEvent)
+    const p = h.transport.connect(new AbortController().signal)
+    const pc = await driveToRemoteDescription(h)
+    pc.remote.resolve()
+    await flush()
+    pc.dc.open()
+    await p
+
+    pc.dc.receive('not json')
+    pc.dc.receive(JSON.stringify({ nope: true }))
+    pc.dc.receive(JSON.stringify({ type: 'response.created' }))
+    expect(onEvent).toHaveBeenCalledTimes(1)
+    expect(onEvent).toHaveBeenCalledWith({ type: 'response.created' })
+
+    h.transport.close()
+    expect(onClosed).not.toHaveBeenCalled()
+    expect(pc.close).toHaveBeenCalled()
+  })
+
+  it('is single-use', async () => {
+    const h = makeTransport()
+    h.transport.close()
+    await expect(h.transport.connect(new AbortController().signal)).rejects.toThrow('single-use')
+  })
+})
+
+// ── 컨트롤러 (fake transport) ──
+
+class FakeTransport implements RealtimeTransport {
+  readonly connectCall = deferred<void>()
+  open = false
+  ignoreAbort = false
+  sent: RealtimeClientEvent[] = []
+  close = vi.fn(() => {
+    this.open = false
+  })
+  private eventCb: ((e: RealtimeServerEvent) => void) | null = null
+  private closedCb: (() => void) | null = null
+
+  connect(signal: AbortSignal): Promise<void> {
+    signal.addEventListener('abort', () => {
+      if (!this.ignoreAbort) this.connectCall.reject(new RealtimeAbortError())
+    })
+    return this.connectCall.promise.then(() => {
+      this.open = true
+    })
+  }
+  send(event: RealtimeClientEvent): boolean {
+    if (!this.open) return false
+    this.sent.push(event)
+    return true
+  }
+  onEvent(cb: (e: RealtimeServerEvent) => void): () => void {
+    this.eventCb = cb
+    return () => undefined
+  }
+  onClosed(cb: () => void): () => void {
+    this.closedCb = cb
+    return () => undefined
+  }
+  emit(e: RealtimeServerEvent): void {
+    this.eventCb?.(e)
+  }
+  drop(): void {
+    this.open = false
+    this.closedCb?.()
+  }
+}
+
+function makeController(): { controller: RealtimeSessionController; transports: FakeTransport[] } {
+  const transports: FakeTransport[] = []
+  const controller = new RealtimeSessionController(() => {
+    const t = new FakeTransport()
+    transports.push(t)
+    return t
+  }, makeCtx())
+  return { controller, transports }
+}
+
+describe('RealtimeSessionController', () => {
+  it('stop during connecting cancels the attempt, closes the transport and never goes live', async () => {
+    const { controller, transports } = makeController()
+    const p = controller.start()
+    expect(controller.getSnapshot().connectionState).toBe('connecting')
+    controller.stop()
+    expect(transports[0].close).toHaveBeenCalled()
+    await expect(p).resolves.toBe('cancelled')
+    expect(controller.getSnapshot().connectionState).toBe('idle')
+  })
+
+  it('a connect that completes after stop (abort ignored) is closed and never goes live', async () => {
+    const { controller, transports } = makeController()
+    const p = controller.start()
+    const t = transports[0]
+    t.ignoreAbort = true
+    controller.stop()
+    t.close.mockClear()
+    t.connectCall.resolve()
+    await expect(p).resolves.toBe('cancelled')
+    expect(t.close).toHaveBeenCalled()
+    expect(t.sent).toEqual([])
+    expect(controller.getSnapshot().connectionState).toBe('idle')
+  })
+
+  it('concurrent start calls share one connection attempt', async () => {
+    const { controller, transports } = makeController()
+    const a = controller.start()
+    const b = controller.start()
+    expect(transports).toHaveLength(1)
+    transports[0].connectCall.resolve()
+    await expect(a).resolves.toBe('live')
+    await expect(b).resolves.toBe('live')
+    await expect(controller.start()).resolves.toBe('live')
+    expect(transports).toHaveLength(1)
+  })
+
+  it('sends session.update when live and sendText right after start is delivered', async () => {
+    const { controller, transports } = makeController()
+    const p = controller.start()
+    transports[0].connectCall.resolve()
+    await expect(p).resolves.toBe('live')
+    controller.sendText('hello')
+    const t = transports[0]
+    expect(t.sent[0]).toEqual(REALTIME_SESSION_UPDATE)
+    expect(t.sent.slice(1).map((e) => e.type)).toEqual(['conversation.item.create', 'response.create'])
+    expect(controller.getSnapshot().messages.map((m) => m.content)).toEqual(['hello'])
+    expect(controller.getSnapshot().conversationState).toBe('listening')
+  })
+
+  it('queues text sent while connecting and flushes it after session.update', async () => {
+    const { controller, transports } = makeController()
+    const p = controller.start()
+    controller.sendText('first')
+    expect(controller.getSnapshot().messages.map((m) => m.content)).toEqual(['first'])
+    expect(transports[0].sent).toEqual([])
+    transports[0].connectCall.resolve()
+    await p
+    expect(transports[0].sent.map((e) => e.type)).toEqual([
+      'session.update',
+      'conversation.item.create',
+      'response.create',
+    ])
+  })
+
+  it('after the connection drops, the session reports error and start reconnects with a new transport', async () => {
+    const { controller, transports } = makeController()
+    const p = controller.start()
+    transports[0].connectCall.resolve()
+    await p
+    transports[0].emit({ type: 'response.created' })
+    transports[0].emit({ type: 'response.text.delta', delta: 'partial' })
+
+    transports[0].drop()
+    const snap = controller.getSnapshot()
+    expect(snap.connectionState).toBe('error')
+    expect(snap.conversationState).toBe('idle')
+    expect(snap.messages.map((m) => m.content)).toEqual(['partial'])
+
+    const again = controller.start()
+    expect(transports).toHaveLength(2)
+    transports[1].connectCall.resolve()
+    await expect(again).resolves.toBe('live')
+    // 이전 전송 계층의 늦은 이벤트는 무시된다
+    transports[0].emit({ type: 'response.created' })
+    expect(controller.getSnapshot().conversationState).toBe('listening')
+  })
+
+  it('a failed connect reports failed with the error message and allows retry', async () => {
+    const { controller, transports } = makeController()
+    const p = controller.start()
+    transports[0].connectCall.reject(new Error('token request failed'))
+    await expect(p).resolves.toBe('failed')
+    expect(controller.getSnapshot()).toMatchObject({
+      connectionState: 'error',
+      conversationState: 'idle',
+      error: 'token request failed',
+    })
+    void controller.start()
+    expect(transports).toHaveLength(2)
+    expect(controller.getSnapshot().error).toBeNull()
+  })
+
+  it('notifies subscribers and applies server events only from the current transport', async () => {
+    const { controller, transports } = makeController()
+    const listener = vi.fn()
+    const unsubscribe = controller.subscribe(listener)
+    const p = controller.start()
+    transports[0].connectCall.resolve()
+    await p
+    listener.mockClear()
+    transports[0].emit({ type: 'response.created' })
+    expect(listener).toHaveBeenCalledTimes(1)
+    expect(controller.getSnapshot().conversationState).toBe('thinking')
+    controller.cancelResponse()
+    expect(transports[0].sent.at(-1)).toEqual({ type: 'response.cancel' })
+    expect(controller.getSnapshot().conversationState).toBe('listening')
+    unsubscribe()
+  })
+
+  it('sendText is ignored when no session exists', () => {
+    const { controller, transports } = makeController()
+    controller.sendText('x')
+    expect(transports).toHaveLength(0)
+    expect(controller.getSnapshot().messages).toEqual([])
+  })
+})