Use these guides when a carrier sends an existing number to the platform by SIP. Provider settings and signaling networks can change; the source pages below were checked on 22 August 2026. Re-check the linked provider source before changing a production trunk.
The rule that prevents most inbound failures
The platform SIP FQDN is the destination for the provider’s INVITE. It is not an authentication method. Configure both sides:
- Copy the platform SIP URI from Settings → Phone numbers → Add SIP integration.
- Route the provider’s number to that FQDN.
- Create the platform trunk with the exact DID the provider sends. Use
+E.164 unless the provider guide explicitly says otherwise. A SIP extension is not a wildcard.
- Choose SIP username/password only when the provider documents digest authentication toward the destination. Otherwise choose Provider source IPs and enter the provider’s current SIP-signaling IPs or CIDRs.
- Keep outbound credentials and the provider termination host separate from inbound authentication.
Never use RTP/media ranges as Provider source IPs, never copy an IP list from another carrier, and never remove the exact DID to make matching broader.
Supported guides
Prove the setup
Call the exact DID from an external phone. The setup is only confirmed when a new inbound item appears in Calls, the assigned assistant joins, and two-way audio works. Ringing, a delay, or a provider-side call duration alone does not prove that the call reached the platform.
Sources: LiveKit inbound trunks, LiveKit provider quickstarts. Accessed 22 August 2026. 
