Add conversation history management and API endpoints

- Introduce new database models for conversation sessions, messages, and artifacts to support conversation history tracking.
- Implement API routes for listing conversations and retrieving detailed conversation data, enhancing user interaction with historical records.
- Add a conversation recorder service to persist conversation messages in real-time without disrupting ongoing calls.
- Update the frontend to display conversation history, including filtering and sorting options, improving user experience.
- Enhance the pipeline to integrate conversation history recording seamlessly during interactions.

frontend/src/hooks/use-voice-preview.ts

@@ -230,6 +230,7 @@ export function useVoicePreview(
   const dataChannelRef = useRef<RTCDataChannel | null>(null);
   const localStreamRef = useRef<MediaStream | null>(null);
   const startingRef = useRef(false);
+  const negotiationInProgressRef = useRef(false);
   const messageSeqRef = useRef(0);
   const networkStatsRef = useRef<NetworkStatsSample | null>(null);
   const videoBitrateRef = useRef(INITIAL_VIDEO_BITRATE);
@@ -341,6 +342,7 @@ export function useVoicePreview(
     localStreamRef.current = null;
     if (audioRef.current) audioRef.current.srcObject = null;
     startingRef.current = false;
+    negotiationInProgressRef.current = false;
     onNodeActiveRef.current?.(null);
   }, []);
 
@@ -436,10 +438,18 @@ export function useVoicePreview(
       try {
         const msg = JSON.parse(event.data);
         if (msg.type === "answer") {
-          await pcRef.current?.setRemoteDescription({
-            type: "answer",
-            sdp: msg.payload.sdp,
-          });
+          const currentPc = pcRef.current;
+          if (!currentPc) return;
+          try {
+            await currentPc.setRemoteDescription({
+              type: "answer",
+              sdp: msg.payload.sdp,
+            });
+          } finally {
+            if (pcRef.current === currentPc) {
+              negotiationInProgressRef.current = false;
+            }
+          }
         } else if (msg.type === "ice-candidate" && msg.payload?.candidate) {
           // 后端当前不主动 trickle,留兼容
           try {
@@ -478,6 +488,42 @@ export function useVoicePreview(
       const pc = new RTCPeerConnection({ iceServers });
       pcRef.current = pc;
 
+      const sendOffer = async () => {
+        if (
+          pcRef.current !== pc ||
+          ws.readyState !== WebSocket.OPEN ||
+          pc.signalingState !== "stable" ||
+          negotiationInProgressRef.current
+        ) {
+          return;
+        }
+
+        negotiationInProgressRef.current = true;
+        try {
+          const offer = await pc.createOffer();
+          await pc.setLocalDescription(offer);
+          const localDescription = pc.localDescription;
+          if (!localDescription?.sdp) {
+            throw new Error("浏览器无法创建 WebRTC offer。");
+          }
+          ws.send(
+            JSON.stringify({
+              type: "offer",
+              payload: {
+                pc_id: pcId,
+                sdp: localDescription.sdp,
+                type: localDescription.type,
+                assistant_id: assistantId,
+                vision_enabled: Boolean(options.visionEnabled),
+              },
+            }),
+          );
+        } catch (offerError) {
+          negotiationInProgressRef.current = false;
+          throw offerError;
+        }
+      };
+
       pc.onicecandidate = (e) => {
         if (ws.readyState !== WebSocket.OPEN) return;
         ws.send(
@@ -503,10 +549,15 @@ export function useVoicePreview(
       channel.onopen = () => {
         channel.send(JSON.stringify({ type: "client-ready" }));
       };
-      channel.onmessage = (event) => {
+      channel.onmessage = async (event) => {
         try {
           const msg = JSON.parse(event.data);
           if (
+            msg?.type === "signalling" &&
+            msg?.message?.type === "renegotiate"
+          ) {
+            await sendOffer();
+          } else if (
             msg?.type === "assistant-text-start" &&
             typeof msg.turn_id === "string"
           ) {
@@ -631,25 +682,8 @@ export function useVoicePreview(
         }
       }
 
-      // 4) 生成 offer 并发给后端(assistant_id 在 payload 顶层)
-      const offer = await pc.createOffer();
-      await pc.setLocalDescription(offer);
-      const localDescription = pc.localDescription;
-      if (!localDescription?.sdp) {
-        throw new Error("浏览器无法创建 WebRTC offer。");
-      }
-      ws.send(
-        JSON.stringify({
-          type: "offer",
-          payload: {
-            pc_id: pcId,
-            sdp: localDescription.sdp,
-            type: localDescription.type,
-            assistant_id: assistantId,
-            vision_enabled: Boolean(options.visionEnabled),
-          },
-        }),
-      );
+      // 4) 生成 offer 并发给后端；同一方法也响应 SmallWebRTC 的重新协商请求。
+      await sendOffer();
     } catch (connectionError) {
       fail(errorMessage(connectionError, "无法连接语音服务。"));
     } finally {