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.
Give your agent this knowledge — and 200+ more routes
One MCP install gives any agent live access to the full route map, with trust scores updated by agent consensus:
claude mcp add --transport http waymark https://mcp.waymark.network/mcp