feat: 站点 JSON、语音终端 WebSocket 指派与客户端联调

- 用 OR_SITE_CONFIG_JSON_FILE 统一术间配置（video_rtsp_urls + voice_or_room_bindings）
- VoiceTerminalHub：assignment、WS 推送与 HTTP 查询；开录/停录后 notify
- 一键联调 orchestrate-and-start 与 /client/surgeries/start 共用指派逻辑，修复 demo 路径不发 WS
- 语音桌面端：SIGINT 退出、shutdown 清理、仅 WS 指派、固定 pending 轮询间隔、界面仅保留录音时长
- 新增/调整契约与绑定测试，文档与示例配置同步

Made-with: Cursor

app/services/video/session_manager.py

@@ -223,7 +223,17 @@ class CameraSessionManager:
             await self.stop_surgery(surgery_id, require_active=True)
             raise
 
-    async def stop_surgery(self, surgery_id: str, *, require_active: bool = True) -> None:
+    def set_voice_terminal_id(self, surgery_id: str, terminal_id: str | None) -> None:
+        """开录成功后写入，供停录时向对应桌面终端推送 end。"""
+        run = self._registry.get_running(surgery_id)
+        if run is None:
+            return
+        tid = (terminal_id or "").strip()
+        run.state.voice_terminal_id = tid or None
+
+    async def stop_surgery(
+        self, surgery_id: str, *, require_active: bool = True
+    ) -> str | None:
         run = await self._registry.unregister(surgery_id)
         if run is None:
             if require_active:
@@ -231,8 +241,9 @@ class CameraSessionManager:
                     "RECORDING_NOT_STOPPED",
                     "停录未能完成：当前没有该手术的活跃录制会话。",
                 )
-            return
+            return None
 
+        voice_tid = run.state.voice_terminal_id
         run.stop_event.set()
         results = await asyncio.gather(*run.tasks, return_exceptions=True)
         for res in results:
@@ -255,6 +266,7 @@ class CameraSessionManager:
         append_consumption_log_summary(surgery_id, totals)
         print_consumption_summary_markdown(totals)
         await self._archive.persist_or_archive(surgery_id, details)
+        return voice_tid
 
     # ------------------------------------------------------------------
     # PendingConfirmationStore 协议委托