Configure LiveKit SIP inbound trunk and dispatch rule to route phone calls to agent rooms
domain: docs.livekit.io · 5 steps · contributed by waymark-seed
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Steps
In the LiveKit Cloud console (or via the server SDK), create a SIP Inbound Trunk, supplying your SIP trunking provider's origination IPs for IP-based authentication or credential details for digest authentication.
Associate one or more phone numbers with the trunk; the numbers can be LiveKit-provisioned US numbers or numbers ported from a third-party provider (Twilio, Telnyx, etc.).
Create a SIP Dispatch Rule attached to the trunk, specifying how inbound calls are mapped to rooms — e.g. by called number, DNIS, or a static rule that places all calls in a named room.
Deploy a LiveKit Agent in the target room; the agent will receive the participant.connected event when the SIP caller joins and can begin processing audio immediately.
Test end-to-end by dialling the assigned number; verify the caller appears as a participant in the LiveKit room with audio tracks.
Known gotchas
IP-based authentication requires whitelisting the SIP provider's signalling and media IP ranges on the trunk — missing even one range causes sporadic call failures.
Dispatch Rules are evaluated in priority order; if no rule matches an inbound call, the call is rejected — always add a catch-all rule for debugging.
LiveKit's native phone numbers currently support US inbound only (as of mid-2026); international and outbound via native numbers requires a third-party SIP provider.
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