feat: 添加 WebSocket 心跳保活和资源清理逻辑

- 在 `RealtimeMeetingWebSocketConfig` 中添加配置，关闭 Tomcat 内置的 WebSocket keepalive 检测
- 在 `RealtimeAsrSession` 组件中添加心跳定时器和资源清理逻辑，防止资源泄漏
- 在 `RealtimeMeetingProxyWebSocketHandler` 中过滤前端心跳消息，避免转发给上游 ASR 服务

backend/src/main/java/com/imeeting/config/RealtimeMeetingWebSocketConfig.java

@@ -2,6 +2,9 @@ package com.imeeting.config;
 
 import com.imeeting.websocket.RealtimeMeetingProxyWebSocketHandler;
 import lombok.RequiredArgsConstructor;
+import org.springframework.boot.web.embedded.tomcat.TomcatServletWebServerFactory;
+import org.springframework.boot.web.server.WebServerFactoryCustomizer;
+import org.springframework.context.annotation.Bean;
 import org.springframework.context.annotation.Configuration;
 import org.springframework.web.socket.config.annotation.EnableWebSocket;
 import org.springframework.web.socket.config.annotation.WebSocketConfigurer;
@@ -19,4 +22,25 @@ public class RealtimeMeetingWebSocketConfig implements WebSocketConfigurer {
         registry.addHandler(realtimeMeetingProxyWebSocketHandler, "/ws/meeting/realtime")
                 .setAllowedOriginPatterns("*");
     }
+
+    /**
+     * 关闭 Tomcat 内置的 WebSocket keepalive ping 检测（sessionIdleTimeout）。
+     * <p>
+     * Tomcat 默认会对 WebSocket session 设置 sessionIdleTimeout（-1 表示无限，但某些版本默认非 -1），
+     * 并通过后台线程定期发送 Ping 帧，若在超时内未收到 Pong 响应，触发
+     * code=1011 "keepalive ping timeout" 强制断开。
+     * 实时 ASR 场景中，客户端持续发送音频帧，由前端心跳保活，
+     * 因此显式将 sessionIdleTimeout 设为 -1（无限）。
+     * </p>
+     */
+    @Bean
+    public WebServerFactoryCustomizer<TomcatServletWebServerFactory> wsSessionIdleTimeoutCustomizer() {
+        return factory -> factory.addConnectorCustomizers(connector -> {
+            // 通过系统属性通知 Tomcat WS 容器关闭 sessionIdleTimeout 检查
+            // org.apache.tomcat.websocket.DEFAULT_SESSION_IDLE_TIMEOUT=-1
+            if (System.getProperty("org.apache.tomcat.websocket.DEFAULT_SESSION_IDLE_TIMEOUT") == null) {
+                System.setProperty("org.apache.tomcat.websocket.DEFAULT_SESSION_IDLE_TIMEOUT", "-1");
+            }
+        });
+    }
 }

backend/src/main/java/com/imeeting/websocket/RealtimeMeetingProxyWebSocketHandler.java

@@ -81,6 +81,12 @@ public class RealtimeMeetingProxyWebSocketHandler extends AbstractWebSocketHandl
 
     @Override
     protected void handleTextMessage(WebSocketSession session, TextMessage message) {
+        // 过滤前端发来的心跳保活消息，不转发给上游 ASR 服务
+        if (looksLikeKeepaliveMessage(message.getPayload())) {
+            log.debug("Frontend keepalive received, ignored: meetingId={}, sessionId={}",
+                    session.getAttributes().get(ATTR_MEETING_ID), session.getId());
+            return;
+        }
         MeetingChannelSession meetingSession = getMeetingSession(session);
         if (meetingSession == null || !meetingSession.isChannelOpen()) {
             log.warn("前端文本消息已忽略：上游 ASR 连接不可用，meetingId={}, sessionId={}",
@@ -115,11 +121,10 @@ public class RealtimeMeetingProxyWebSocketHandler extends AbstractWebSocketHandl
 
     @Override
     protected void handlePongMessage(WebSocketSession session, PongMessage message) {
-      if (getMeetingSession(session) == null) {
+        // Pong 是浏览器对服务端（Tomcat）发出 Ping 帧的回应，代理层在此消化即可，无需转发给上游 ASR。
-            return;
+        // 转发 Pong 给上游没有语义意义，且可能引起上游协议混乱。
-        }
+        log.debug("Frontend pong received (keepalive): meetingId={}, sessionId={}",
-      log.debug("前端 pong 已在本地忽略：meetingId={}, sessionId={}, bytes={}",
+                session.getAttributes().get(ATTR_MEETING_ID), session.getId());
-        session.getAttributes().get(ATTR_MEETING_ID), session.getId(), message.getPayloadLength());
     }
 
     @Override
@@ -346,6 +351,14 @@ public class RealtimeMeetingProxyWebSocketHandler extends AbstractWebSocketHandl
         return normalized.substring(0, 240) + "...";
     }
 
+    private boolean looksLikeKeepaliveMessage(String payload) {
+        if (payload == null || payload.isBlank()) {
+            return false;
+        }
+        String normalized = payload.replaceAll("\\s+", "");
+        return normalized.contains("\"type\":\"keepalive\"");
+    }
+
     static final class MeetingChannelSession {
         private final Long meetingId;
         private final RealtimeAsrChannel channel;

frontend/src/pages/business/RealtimeAsrSession.tsx

@@ -244,6 +244,7 @@ export function RealtimeAsrSession() {
   const [sessionStatus, setSessionStatus] = useState<RealtimeMeetingSessionStatus | null>(null);
   const transcriptRef = useRef<HTMLDivElement | null>(null);
   const wsRef = useRef<WebSocket | null>(null);
+  const wsHeartbeatRef = useRef<ReturnType<typeof setInterval> | null>(null);
   const audioContextRef = useRef<AudioContext | null>(null);
   const processorRef = useRef<ScriptProcessorNode | null>(null);
   const audioSourceRef = useRef<MediaStreamAudioSourceNode | null>(null);
@@ -373,6 +374,33 @@ export function RealtimeAsrSession() {
     return () => window.removeEventListener("pagehide", handlePageHide);
   }, [meetingId]);
 
+  // 组件卸载（切换路由）时统一清理所有资源，防止定时器、WebSocket、音频管道泄漏
+  useEffect(() => {
+    return () => {
+      // 1. 清理心跳定时器
+      if (wsHeartbeatRef.current !== null) {
+        clearInterval(wsHeartbeatRef.current);
+        wsHeartbeatRef.current = null;
+      }
+      // 2. 关闭 WebSocket（移除回调避免触发不必要的状态更新）
+      if (wsRef.current) {
+        wsRef.current.onclose = null;
+        wsRef.current.onerror = null;
+        wsRef.current.onmessage = null;
+        wsRef.current.close();
+        wsRef.current = null;
+      }
+      // 3. 关闭音频管道（同步处理，无需 await）
+      processorRef.current?.disconnect();
+      audioSourceRef.current?.disconnect();
+      streamRef.current?.getTracks().forEach((t) => t.stop());
+      if (audioContextRef.current && audioContextRef.current.state !== "closed") {
+        void audioContextRef.current.close();
+      }
+    };
+  // eslint-disable-next-line react-hooks/exhaustive-deps
+  }, []);
+
   const shutdownAudioPipeline = async () => {
     processorRef.current?.disconnect();
     audioSourceRef.current?.disconnect();
@@ -433,6 +461,10 @@ export function RealtimeAsrSession() {
     setRecording(false);
     setStatusText("连接失败");
     sessionStartedRef.current = false;
+    if (wsHeartbeatRef.current !== null) {
+      clearInterval(wsHeartbeatRef.current);
+      wsHeartbeatRef.current = null;
+    }
     wsRef.current?.close();
     wsRef.current = null;
     await shutdownAudioPipeline();
@@ -530,6 +562,13 @@ export function RealtimeAsrSession() {
       }
       const pauseRes = await pauseRealtimeMeeting(meetingId);
       await closeFrontendSocket(false);
+      if (wsHeartbeatRef.current !== null) {
+        clearInterval(wsHeartbeatRef.current);
+        wsHeartbeatRef.current = null;
+      }
+      wsRef.current?.close();
+      wsRef.current = null;
+      sessionStartedRef.current = false;
       await shutdownAudioPipeline();
       setSessionStatus(pauseRes.data.data);
       setRecording(false);
@@ -575,12 +614,41 @@ export function RealtimeAsrSession() {
         hotwords: sessionDraft.hotwords || [],
       });
       const socketSession = socketSessionRes.data.data;
+
+      // 如果已有旧的 WebSocket（比如重连），先强制关闭并清理心跳
+      if (wsHeartbeatRef.current !== null) {
+        clearInterval(wsHeartbeatRef.current);
+        wsHeartbeatRef.current = null;
+      }
+      if (wsRef.current) {
+        wsRef.current.onclose = null;
+        wsRef.current.onerror = null;
+        wsRef.current.onmessage = null;
+        wsRef.current.close();
+        wsRef.current = null;
+      }
+
       const socket = new WebSocket(buildRealtimeProxyWsUrl(socketSession));
       socket.binaryType = "arraybuffer";
       wsRef.current = socket;
 
       socket.onopen = () => {
         setStatusText("识别服务连接中，等待第三方服务就绪...");
+        // 先清除旧定时器（防止 onopen 被意外重复触发时叠加）
+        if (wsHeartbeatRef.current !== null) {
+          clearInterval(wsHeartbeatRef.current);
+          wsHeartbeatRef.current = null;
+        }
+        // 启动心跳保活：每 20 秒发送一次 keepalive JSON，防止服务端 ping timeout 断开
+        wsHeartbeatRef.current = setInterval(() => {
+          if (socket.readyState === WebSocket.OPEN) {
+            try {
+              socket.send(JSON.stringify({ type: "keepalive" }));
+            } catch {
+              // ignore heartbeat send failure
+            }
+          }
+        }, 20000);
       };
 
       socket.onmessage = (event) => {
@@ -641,6 +709,10 @@ export function RealtimeAsrSession() {
       };
 
       socket.onclose = () => {
+        if (wsHeartbeatRef.current !== null) {
+          clearInterval(wsHeartbeatRef.current);
+          wsHeartbeatRef.current = null;
+        }
         setConnecting(false);
         setRecording(false);
         sessionStartedRef.current = false;
@@ -668,6 +740,16 @@ export function RealtimeAsrSession() {
     setStatusText("结束会议中...");
 
     await closeFrontendSocket(true);
+    if (wsRef.current?.readyState === WebSocket.OPEN) {
+      wsRef.current.send(JSON.stringify({ is_speaking: false }));
+    }
+    if (wsHeartbeatRef.current !== null) {
+      clearInterval(wsHeartbeatRef.current);
+      wsHeartbeatRef.current = null;
+    }
+    wsRef.current?.close();
+    wsRef.current = null;
+    sessionStartedRef.current = false;
 
     await shutdownAudioPipeline();