self hosted sip service keeps closing calls instan...
# ask-aiq
quick-stone-30848
02/18/2026, 12:25 PMself hosted sip service keeps closing calls instantly, reason: "flood"
how can i configure flood settings
t
tall-airline-16756
02/18/2026, 12:25 PMThanks for reaching out, sanidhya. I'm an AI assistant and I'm here to help. Give me a moment while I research this topic.
tall-airline-16756
02/18/2026, 12:25 PMThanks — I hear you: your self‑hosted SIP service is dropping calls immediately with the disconnect reason "flood," and you want to know how to configure flood settings.
I don’t have any documentation here that describes a "flood" disconnect reason or shows flood-related configuration options for LiveKit’s SIP integration. To help you quickly and accurately, I need a few details so I can either point to the right setting or walk you through troubleshooting:
1. Which SIP component are you running self‑hosted?
◦ For example: LiveKit SIP gateway, Asterisk, FreeSWITCH, Kamailio, OpenSIPS, or a vendor trunk?
1. Version and deployment details
◦ The exact version of the SIP component and whether it’s running in a container, VM, or on bare metal.
1. Where the disconnect message appears
◦ Is "flood" shown in LiveKit logs, in your SIP server logs, or as a SIP response/reason to the remote peer?
1. Example logs or SIP trace
◦ A short redacted log snippet showing the call flow and the "flood" reason (timestamps and SIP messages like INVITE/200/BYE). That will help identify whether the server itself is terminating the call or an upstream peer is sending the reason.
1. Any recent changes
◦ Network, config, scaling, or rate-limiting changes that coincided with this behavior.
If you aren’t sure about some of the above, start by pasting the line(s) from your SIP server or LiveKit logs that include the word "flood" and the surrounding messages. With that I can:
• Tell you whether "flood" is coming from the SIP server or from an upstream provider.
• Suggest which configuration file or parameter to adjust (rate limits, anti‑flood thresholds, SIP messages/minute, connection limits) for the specific SIP software.
• Provide step‑by‑step troubleshooting to reproduce and safely raise flood thresholds or disable the check temporarily so calls aren’t closed while we diagnose.
Send the component name/version and a short log/SIP trace snippet and I’ll guide you through the next steps.
Sources: sip | livekit/livekit_composite | DeepWiki | Making calls overview | LiveKit Documentation | Agents telephony integration | LiveKit Documentation | examples/frontdesk | livekit/livekit_composite | DeepWiki | How to do call transfers with Five9 | LiveKit
q
quick-stone-30848
02/18/2026, 12:29 PMi am using livekit sip, running in a docker container on a vm
here are logs (redacted sensitive info)
2026-02-18T122604.287Z DEBUG New connection {"nodeID": "NE_oYJ3EjFWfGVh", "caller": "transport<TCP>", "raddr": ""}
2026-02-18T122604.288Z DEBUG Getting connection {"nodeID": "NE_oYJ3EjFWfGVh", "caller": "transport<TCP>", "addr": ""}
2026-02-18T122604.289Z INFO sip sip/inbound.go:335 processing invite {"nodeID": "NE_oYJ3EjFWfGVh", "callID": "SCL_9pADEPXmGBMP", "traceID": "397041444550586d", "fromIP": "", "toIP": "", "transport": "tcp", "fromHost": "", "fromUser": "+", "toHost": "", "toUser": "+", "sipTag": "as2522347e", "sipCallID": ""}
2026-02-18T122604.307Z DEBUG sip sip/inbound.go:707 Rejecting inbound flood {"nodeID": "NE_oYJ3EjFWfGVh", "callID": "SCL_9pADEPXmGBMP", "traceID": "397041444550586d", "fromIP": "", "toIP": "", "transport": "tcp", "fromHost": "", "fromUser": "+", "toHost": "", "toUser": "+", "sipTag": "as2522347e", "sipCallID": "", "jitterBuf": false}
2026-02-18T122604.307Z DEBUG Server transaction terminating {"nodeID": "NE_oYJ3EjFWfGVh", "caller": "transaction.Layer"}
2026-02-18T122604.307Z DEBUG Server transaction destroyed {"nodeID": "NE_oYJ3EjFWfGVh", "caller": "transaction.Layer", "tx": "z9hG4bKccd.a9d716034c33bec2c97373022cac7d98.0____5070__INVITE"}
2026-02-18T122604.307Z INFO sip sip/inbound.go:1101 Closing inbound call {"nodeID": "NE_oYJ3EjFWfGVh", "callID": "SCL_9pADEPXmGBMP", "traceID": "397041444550586d", "fromIP": "", "toIP": "", "transport": "tcp", "fromHost": "", "fromUser": "+", "toHost": "", "toUser": "+", "sipTag": "as2522347e", "sipCallID": "", "jitterBuf": false, "status": 486, "reason": "flood"}
2026-02-18T122604.307Z INFO sip sip/inbound.go:1084 call statistics {"nodeID": "NE_oYJ3EjFWfGVh", "callID": "SCL_9pADEPXmGBMP", "traceID": "397041444550586d", "fromIP": "", "toIP": "", "transport": "tcp", "fromHost": "", "fromUser": "+", "toHost": "", "toUser": "+", "sipTag": "as2522347e", "sipCallID": "", "jitterBuf": false, "status": 486, "reason": "flood", "stats": {"port":{"streams":0,"packets":0,"packets_ignored":0,"packets_input":0,"mux_packets":0,"mux_bytes":0,"audio_packets":0,"audio_bytes":0,"dtmf_packets":0,"dtmf_bytes":0,"closed":false},"room":{"input_packets":0,"input_bytes":0,"dtmf_packets":0,"mixer_samples":0,"mixer_frames":0,"output_samples":0,"output_frames":0,"closed":true},"mixer":{"tracks":0,"tracks_total":0,"restarts":0,"mixes":0,"mixes_timed":0,"mixes_jump":0,"mixes_zero":0,"input_samples":0,"input_frames":0,"mixed_samples":0,"mixed_frames":0,"output_samples":0,"output_frames":0},"closed":true}, "durMin": 0}
2026-02-18T122604.307Z INFO sip service/service.go:258 SIP call ended {"nodeID": "NE_oYJ3EjFWfGVh", "callID": "SCL_9pADEPXmGBMP", "reason": "flood"}
2026-02-18T122604.310Z DEBUG Server transaction terminating {"nodeID": "NE_oYJ3EjFWfGVh", "caller": "transaction.Layer"}
2026-02-18T122604.310Z DEBUG Server transaction destroyed {"nodeID": "NE_oYJ3EjFWfGVh", "caller": "transaction.Layer", "tx": "z9hG4bKccd.a9d716034c33bec2c97373022cac7d98.0____5070__INVITE"}
t
tall-airline-16756
02/18/2026, 12:29 PMI don't have the answer you're looking for.
You could also try asking your question:
• in one of the other Slack channels
or
• to https://deepwiki.com/livekit/livekit_composite which is trained on all LiveKit source code
If you find the answer, please post it here to help others!
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