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

tests/test_app_integration.py

@@ -72,9 +72,12 @@ class _StubCameraSessionManager:
         )
         await self._registry.register(surgery_id, run)
 
+    def set_voice_terminal_id(self, surgery_id: str, terminal_id: str | None) -> None:
+        self._real.set_voice_terminal_id(surgery_id, terminal_id)
+
     async def stop_surgery(
         self, surgery_id: str, *, require_active: bool = True
-    ) -> None:
+    ) -> str | None:
         run = await self._registry.unregister(surgery_id)
         if run is None:
             if require_active:
@@ -84,9 +87,11 @@ class _StubCameraSessionManager:
                     "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
         details = list(run.state.details)
         await self._archive.persist_or_archive(surgery_id, details)
+        return voice_tid
 
     def __getattr__(self, name: str) -> Any:
         return getattr(self._real, name)