chpt 5

Author: PinJinx

src/hooks/useWebRtcStream.ts

@@ -33,8 +33,31 @@ export function useWebRtcStream({ token }: UseWebRtcStreamOptions) {
 		}
 	}, [])
 
+	const checkServerActive = useCallback(async (): Promise<boolean> => {
+		try {
+			const headers: Record<string, string> = {}
+			if (token) {
+				headers.Authorization = `Bearer ${token}`
+			}
+			const controller = new AbortController()
+			const timeoutId = setTimeout(() => controller.abort(), 2000)
+			const response = await fetch("/api/host/status", {
+				headers,
+				signal: controller.signal,
+			})
+			clearTimeout(timeoutId)
+			return response.ok
+		} catch {
+			return false
+		}
+	}, [token])
+
 	const triggerRetry = useCallback(() => {
-		if (trackActiveRef.current || retryTimerRef.current) return
+		if (retryTimerRef.current) return
+
+		setTrackActive(false)
+		setVideoStream(null)
+
 		console.log(
 			"[WebRTC] Startup/transient network failure, retrying automatically in 2 seconds...",
 		)
@@ -45,6 +68,15 @@ export function useWebRtcStream({ token }: UseWebRtcStreamOptions) {
 		}, 2000)
 	}, [])
 
+	const handleNetworkFailure = useCallback(async () => {
+		const isServerOnline = await checkServerActive()
+		if (isServerOnline) {
+			triggerRetry()
+		} else {
+			setError("Server has quit or is unreachable")
+		}
+	}, [checkServerActive, triggerRetry])
+
 	const reconnect = () => {
 		if (retryTimerRef.current) {
 			clearTimeout(retryTimerRef.current)
@@ -84,9 +116,9 @@ export function useWebRtcStream({ token }: UseWebRtcStreamOptions) {
 			})
 			.catch((err) => {
 				console.error("[WebRTC] Session init failed:", err)
-				triggerRetry()
+				handleNetworkFailure()
 			})
-	}, [token, reconnectAttempt, triggerRetry])
+	}, [token, reconnectAttempt, handleNetworkFailure])
 
 	useEffect(() => {
 		if (!activeSessionId) return
@@ -123,7 +155,7 @@ export function useWebRtcStream({ token }: UseWebRtcStreamOptions) {
 				}),
 			}).catch((err) => {
 				console.error(err)
-				triggerRetry()
+				handleNetworkFailure()
 			})
 		}
 
@@ -132,7 +164,7 @@ export function useWebRtcStream({ token }: UseWebRtcStreamOptions) {
 				videoPc.connectionState === "failed" ||
 				videoPc.connectionState === "disconnected"
 			) {
-				triggerRetry()
+				handleNetworkFailure()
 			}
 		}
 
@@ -166,7 +198,7 @@ export function useWebRtcStream({ token }: UseWebRtcStreamOptions) {
 				}),
 			}).catch((err) => {
 				console.error(err)
-				triggerRetry()
+				handleNetworkFailure()
 			})
 		}
 
@@ -175,7 +207,7 @@ export function useWebRtcStream({ token }: UseWebRtcStreamOptions) {
 				inputPc.connectionState === "failed" ||
 				inputPc.connectionState === "disconnected"
 			) {
-				triggerRetry()
+				handleNetworkFailure()
 			}
 		}
 
@@ -191,7 +223,7 @@ export function useWebRtcStream({ token }: UseWebRtcStreamOptions) {
 
 		sse.onerror = (event) => {
 			console.error("[WebRTC] SSE error:", event)
-			triggerRetry()
+			handleNetworkFailure()
 		}
 
 		sse.addEventListener("stream-error", (event) => {
@@ -232,7 +264,7 @@ export function useWebRtcStream({ token }: UseWebRtcStreamOptions) {
 				}
 			} catch (err) {
 				console.error("[WebRTC] Video offer handling failed:", err)
-				triggerRetry()
+				handleNetworkFailure()
 			}
 		})
 
@@ -278,7 +310,7 @@ export function useWebRtcStream({ token }: UseWebRtcStreamOptions) {
 				}
 			} catch (err) {
 				console.error("[WebRTC] Input answer failed:", err)
-				triggerRetry()
+				handleNetworkFailure()
 			}
 		})
 
@@ -319,17 +351,51 @@ export function useWebRtcStream({ token }: UseWebRtcStreamOptions) {
 
 		sendInputOffer().catch((err) => {
 			console.error(err)
-			triggerRetry()
+			handleNetworkFailure()
 		})
 
+		// Video stream watchdog to automatically recover from silent stream freezes
+		let lastBytesReceived = 0
+		let lastBytesTime = Date.now()
+
+		const statsInterval = setInterval(async () => {
+			if (!trackActiveRef.current) return
+			try {
+				const stats = await videoPc.getStats()
+				let videoInbound = null
+				for (const report of stats.values()) {
+					if (report.type === "inbound-rtp" && report.kind === "video") {
+						videoInbound = report
+						break
+					}
+				}
+				if (videoInbound) {
+					const bytes = videoInbound.bytesReceived
+					const now = Date.now()
+					if (bytes > lastBytesReceived) {
+						lastBytesReceived = bytes
+						lastBytesTime = now
+					} else if (now - lastBytesTime > 4000) {
+						console.warn(
+							"[WebRTC] Video stream freeze detected (no bytes received for 4s), reconnecting...",
+						)
+						handleNetworkFailure()
+					}
+				}
+			} catch (err) {
+				console.error("[WebRTC] Failed to fetch stats:", err)
+			}
+		}, 2000)
+
 		return () => {
+			clearInterval(statsInterval)
 			sse.close()
 			videoPc.close()
 			inputPc.close()
 			setTrackActive(false)
 			setVideoStream(null)
 		}
-	}, [activeSessionId, token, registerDataChannel, triggerRetry])
+	}, [activeSessionId, token, registerDataChannel, handleNetworkFailure])
 
 	return { trackActive, videoStream, error, reconnect, sendInputEvent }
 }