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LiveKit SIP Trunk Missing After a Container Recreate or Host Reboot?

A missing LiveKit SIP trunk can mean a missing API object or a stopped self-hosted SIP service. Identify the layer before restoring configuration or checking call connectivity.

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If a LiveKit SIP trunk appears to be missing after a container recreate or host reboot, first determine what is actually absent: the SIP server process, or a trunk configuration object identified by a trunk ID. They are separate things, and a missing container does not by itself establish that a stored trunk was deleted. The fix depends on which layer is missing.

First identify what “does not exist” refers to

LiveKit uses “SIP” for both a separately deployed service and API-managed configuration. A self-hosted SIP server runs as its own service alongside LiveKit and Redis; a stored inbound or outbound trunk is a configuration object managed through the SIP APIs. Restoring one does not automatically restore the other. LiveKit describes the separate service and its deployment in the self-hosted SIP server guide, while its SIP documentation covers trunk configuration.

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  • The error names a trunk ID or says “object cannot be found”: investigate the stored object, the ID, and the project or deployment receiving the API request.
  • The SIP container or service is absent, stopped, or not ready: investigate your container or host service configuration and startup logs.
  • The trunk exists and the SIP service is running, but calls fail: check call-flow configuration and network reachability separately.

Capture the exact error and identify whether the affected flow is inbound or outbound before changing configuration.

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For “object cannot be found,” check the trunk object and ID

LiveKit’s SIP troubleshooting guide explains that “Trunk ID references a trunk that doesn’t exist or is inaccessible.” That points to an object lookup or access problem; it does not establish that a container recreate or reboot deleted the trunk.

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  1. Confirm the target environment. Check that the application is sending its request to the intended LiveKit Cloud project or self-hosted deployment, and that the trunk was created there.
  2. Verify the exact ID. Compare the ID in the failing request with the trunk record, including whether the application is still using an old or incorrect value.
  3. Check API access. SIP API requests require SIP admin permission unless an endpoint specifies otherwise. Confirm the credential used by the request has that permission. See LiveKit’s SIP API documentation.
  4. For a stored outbound trunk, list the records. Use the SIP API’s ListSIPOutboundTrunk operation and compare the returned IDs with the ID configured in the application. The outbound calling guide describes creating and querying stored outbound trunks.

If the expected trunk is not returned, verify where and how it was created before recreating it. If it is returned but the request still fails, check the request’s destination environment, ID, and credential rather than assuming the record is gone.

For inbound calls, check both the trunk and dispatch rule

Inbound call routing commonly depends on an inbound trunk and a dispatch rule. Check that the expected trunk exists and that the associated dispatch rule is present and matches the intended routing. LiveKit’s SIP documentation describes the inbound configuration; its Phone Numbers documentation notes an exception to creating an inbound trunk when using LiveKit Phone Numbers.

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Do not diagnose a missing inbound trunk from a failed call alone: the dispatch rule, API target, or SIP service may be the failing layer instead.

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If the self-hosted SIP server container or service is missing

LiveKit documents the self-hosted SIP server as a separately deployed service. Its sample Docker Compose setup connects the SIP service to LiveKit and Redis, but it does not prescribe every operator’s restart policy or persistence setup. Inspect the configuration used by your deployment rather than assuming a container recreation preserves or restores every service automatically.

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  • Check the Compose or orchestrator desired state and whether the SIP service is included and scheduled to start.
  • Review the container or service logs around startup for configuration, connection, or readiness errors.
  • Verify the mounted SIP configuration is present and current, including the LiveKit WebSocket URL, API credentials, and Redis address.
  • Check the host’s service manager or container startup policy if the service fails to return after a reboot.

For LiveKit’s documented VM deployment, configuration is under /opt/livekit and a livekit-docker systemd service is used. This applies only if your host was set up using that documented method; other deployments may use different paths and service names. See the VM deployment guide.

If the service and trunk exist but calls still fail

Only after confirming the SIP service is running and the relevant trunk configuration exists should you treat connectivity as the next suspect. For the documented self-hosted setup, LiveKit identifies SIP signaling on port 5060 and RTP media on ports 10000–20000 by default. Confirm your actual configured ports and that the host firewall and network policy permit the required traffic; defaults may differ when configuration has been changed. See the self-hosted SIP server guide and firewall and ports guide.

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A blocked signaling or media port can prevent calls from connecting, but it is not evidence that an API-managed trunk record has disappeared.

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What to collect if the cause is still unclear

To narrow down the failure, gather the deployment mode (LiveKit Cloud or self-hosted), how it is run (such as Compose, Helm, systemd, or another orchestrator), what was recreated, the sanitized relevant configuration, how the trunk was created (stored or inline), the exact error, and logs around service startup or API lookup. Redact credentials and secrets before sharing configuration or logs.

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