Twilio SIP for voice AI: routing and acceptance checklist
Connect Twilio Elastic SIP Trunking to a voice runtime with clear inbound/outbound setup, caller identity and failure testing.
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Twilio Elastic SIP Trunking can connect telephone traffic to a SIP-capable voice service. It is one part of the system: you still need the receiving runtime, assistant dispatch, permitted business actions and a way to inspect failed calls. Twilio SIP documentation
In Burki, the current voice runtime is LiveKit. Twilio readiness is carrier-specific; an integration guide does not establish that every account, destination or feature combination has passed production acceptance.
Keep inbound and outbound setup separate
Inbound traffic needs a number and route that reach the intended service. Outbound traffic needs the correct termination arrangement, authentication, destination permissions and caller identity. Use Twilio's current terminology and the exact values supplied by your deployment when configuring those directions.
Do not assume an existing programmable-voice webhook is automatically the right SIP route. A browser assistant test also bypasses these carrier requirements.
Prepare a configuration record
| Item | Record without exposing secrets |
|---|---|
| Number | Owning account, purpose and active state |
| Route | Inbound destination and outbound arrangement |
| Authentication | Method and credential owner, not the secret value |
| Runtime | Intended assistant or dispatch rule |
| Limits | Allowed destinations, concurrency and spending controls |
| Recovery | Previous route and person responsible for rollback |
Keep credentials in supported secure settings. Avoid copying a tutorial's example hostname or network rule into a live account without verifying that it describes your service.
Validate the full telephone journey
Call an authorized test number and confirm that the expected assistant answers. Check both audio directions, a caller correction, interruption and hangup. Inspect runtime and carrier identifiers so an error can be traced across the boundary.
If the workflow includes handoff, test answered, rejected and unanswered destinations. Verify the original caller remains accounted for and that a resumed assistant can answer the next question. Do not equate a successful SIP response with a successful human handoff.
Troubleshoot by stage
If the assistant never receives the call, inspect number routing, authentication and dispatch before editing its prompt. If the call connects with one-way audio, investigate media connectivity and negotiation. If conversation works but booking fails, inspect the action connection and returned result.
These are investigation directions, not diagnoses; use the exact call's evidence before changing production settings.
Review cost and ownership
Include number rental and every telephone leg, particularly during transfers. Direct Twilio expenses may sit outside the platform bill. Keep the old route until acceptance is complete, then remove unused resources only after checking dependencies.
See the Twilio integration page, BYO SIP guide and LiveKit telephony documentation for the relevant boundaries. This setup should be demonstrated on your account before real callers depend on it.
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