Initial commit: AI Video Assistant fullstack platform.

Add pipecat-based backend with WebRTC/WS voice routes, Next.js frontend, and Docker Compose orchestration.

Co-authored-by: Cursor <cursoragent@cursor.com>

backend/services/pipecat/transports.py

@@ -0,0 +1,54 @@
+"""Transport 工厂——管线与"输出方式"解耦的关键。
+
+同一条 STT→LLM→TTS 管线,可以挂在不同 transport 上:
+  - WebRTC:浏览器,低延迟,带 NAT 穿透           -> build_webrtc_transport
+  - WS:    裸音频流,服务端/话务/自定义客户端,简单 -> build_ws_transport
+
+未来加电话(Twilio/Vonage)只是再加一个 build_xxx_transport + 对应 serializer。
+对应 dograh 的 transport_setup.py(WebRTC)+ 各 telephony provider 的 transport.py(WS)。
+"""
+
+from fastapi import WebSocket
+
+from pipecat.transports.base_transport import TransportParams
+from pipecat.audio.vad.silero import SileroVADAnalyzer
+
+# WebRTC
+from pipecat.transports.smallwebrtc.connection import SmallWebRTCConnection
+from pipecat.transports.smallwebrtc.transport import SmallWebRTCTransport
+
+# 裸 WS 音频流
+from pipecat.transports.network.fastapi_websocket import (
+    FastAPIWebsocketTransport,
+    FastAPIWebsocketParams,
+)
+from pipecat.serializers.protobuf import ProtobufFrameSerializer
+
+
+def _base_params() -> dict:
+    """两种 transport 共享的音频参数。"""
+    return dict(
+        audio_in_enabled=True,
+        audio_out_enabled=True,
+        vad_analyzer=SileroVADAnalyzer(),  # 本地 VAD,打断功能依赖它
+    )
+
+
+def build_webrtc_transport(connection: SmallWebRTCConnection) -> SmallWebRTCTransport:
+    return SmallWebRTCTransport(
+        webrtc_connection=connection,
+        params=TransportParams(**_base_params()),
+    )
+
+
+def build_ws_transport(websocket: WebSocket) -> FastAPIWebsocketTransport:
+    """裸 WS 输出。序列化用 protobuf(自定义客户端用同款解码);
+    若对接电话商,把 serializer 换成对应的 TwilioFrameSerializer 等即可。
+    """
+    return FastAPIWebsocketTransport(
+        websocket=websocket,
+        params=FastAPIWebsocketParams(
+            serializer=ProtobufFrameSerializer(),
+            **_base_params(),
+        ),
+    )