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Why does this error happend ```{"message": "livek...
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g
Why does this error happend
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{
  "message": "livekit::rtc_engine:453:livekit::rtc_engine - received session close: \"signal client closed: \\\"stream closed\\\"\" UnknownReason Resume",
  "level": "WARNING",
  "name": "livekit",
  "pid": 7474,
  "job_id": "AJ_yuPkRxcLhWFf",
  "timestamp": "2025-09-12T20:27:29.732182+00:00"
}
t
Thanks for reaching out, Snehaal. I'm an AI assistant and I'm here to help. Give me a moment while I research this topic.
It looks like the error happens when the signaling WebSocket connection between your client and the LiveKit server closes unexpectedly, which is often due to brief network hiccups. The good news is that the SDK has an automatic recovery process to help get things back on track. Here are some friendly steps to help you address this: • Keep an Eye on Network Conditions: It’s a great idea to check that the client’s internet connection is stable and that the server infrastructure isn’t facing any network or load challenges. • Use Client Application Listeners: You can leverage the LiveKit SDK's reconnecting and reconnected events to keep your users informed about the reconnection progress, which really helps with transparency. • Review Fallback Configuration: Make sure your connection code supports region fallback so the WebSocket reconnections are more resilient. • Check Server Health: Double-check that the LiveKit server is running smoothly and not over its resource limits. While the SDK does its best to recover automatically, taking these steps can really help reduce any disruptions you might experience. If you need any more guidance, I’m here to help! Sources: Docs::Home | livekit/src/rtc_engine/rtc_session.rs | livekit-rtc/livekit/rtc/_proto/participant_pb2.pyi | packages/livekit-rtc/src/proto/participant_pb.ts | Sources/LiveKit/Protos/livekit_models.pb.swift
g
This is a voice call on not a web client experience. This is a issue during a phone call experience over sip
t
I'm sorry to hear you're experiencing an error during your SIP phone call. This issue usually happens when the signaling WebSocket connection, which is key for managing the WebRTC session, closes unexpectedly. This can sometimes be due to network interruptions or server-side challenges. Here are some friendly steps to help you troubleshoot this: • Ensure Network Stability: It’s a good idea to check that your network supports stable WebSocket connections, as any disruptions here can cause the signaling to disconnect. • Monitor Reconnection Events: Try implementing handlers for reconnection events like 
Reconnecting
and 
Reconnected
in your client app. This can really help you spot and manage any hiccups. • Review Server Logs: Taking a look at your server logs might reveal unexpected WebSocket closures or heavy server load that could be affecting the connection. • Validate SIP-Specific Behavior: Also, double-check that SIP-specific errors such as 
UserUnavailable
or 
UserRejected
aren’t playing a part in this issue. Just so you know, LiveKit tries to help by automatically reconnecting to the signaling WebSocket and performing an ICE restart. By handling these events smoothly in your application, you can really reduce any disruptions. If you need any more help with this, feel free to reach out—I'm here to support you! Sources: lib/src/proto/livekit_models.pbenum.dart | packages/livekit-rtc/src/proto/participant_pb.ts | Sources/LiveKit/Protos/livekit_models.pb.swift | Docs::Home | Connecting to LiveKit | LiveKit Docs
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