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/resources/camera_rtsp_urls.sample.json

@@ -1,4 +0,0 @@
-{
-  "or-cam-01": "rtsp://admin:ChangeMe@192.168.1.101:554/Streaming/Channels/101",
-  "or-cam-02": "rtsp://admin:ChangeMe@192.168.1.102:554/Streaming/Channels/101"
-}

app/resources/or_site_config.sample.json

@@ -0,0 +1,15 @@
+{
+  "video_rtsp_urls": {
+    "or-cam-01": "rtsp://127.0.0.1:18554/demo1"
+  },
+  "voice_or_room_bindings": [
+    {
+      "camera_ids": [
+        "or-cam-01",
+        "or-cam-02"
+      ],
+      "or_room_id": "OR-DEMO",
+      "voice_terminal_id": "desktop-1"
+    }
+  ]
+}