Twilio
Global preset (E.164) for Twilio Elastic SIP Trunking. All fields are verified in the documentation.
Values are taken one-to-one from the product’s telephony preset (sip_presets.py). Check any field marked Not verified in the docs in your provider’s panel; it stays editable in the console with the same note.
- Provider key
twilio_sip- Region
- Global (E.164, non-TR numbers allowed)
- Connection test
- SIP OPTIONS (the provider does not accept REGISTER)
- Username
- Credential List username
- Caller ID (DID)
- Your Twilio number, E.164 (e.g. +14155550100)
Preset values
These values are pre-filled in the console; you can change every one of them.
| Field | Default | Status |
|---|---|---|
| SIP server | enter your own — <your-trunk>.pstn.twilio.com (regional: .pstn.dublin.twilio.com …) | ✓ Verified |
| Transport | UDP | ✓ Verified |
| Authentication | username / password (Digest) | ✓ Verified |
| Dialled number format | e164 | ✓ Verified |
| Inbound signalling IPs | 54.172.60.0/30, 54.244.51.0/30, 54.171.127.192/30, 35.156.191.128/30, 54.65.63.192/30, 54.169.127.128/30, 54.252.254.64/30, 177.71.206.192/30 | ✓ Verified |
| Suggested codecs | PCMU, PCMA | ✓ Verified |
| SIP REFER (transfer) | supported | ✓ Verified |
Provider notes
Twilio Console → Elastic SIP Trunking → Trunk: create a Termination URI (<name>.pstn.twilio.com) and a Credential List; the username/password come from that list. Twilio doesn't accept REGISTER (credentials are sent with every INVITE). Enter your LiveKit SIP address as the Origination URI (sip:<livekit-sip>;transport=tcp). For TLS add :5061 to the address. For REFER transfers, enable 'Call Transfers' (and 'PSTN Transfer') on the trunk.
Sources
- www.twilio.com/docs/sip-trunking
- www.twilio.com/docs/sip-trunking/ip-addresses
- www.twilio.com/docs/sip-trunking/codecs
- www.twilio.com/docs/sip-trunking/call-transfer
- docs.livekit.io/telephony/start/providers/twilio/
Connecting
Section titled “Connecting”- In Settings → Integrations → Phone line, pick Twilio as the provider. The preset fills in the fields.
- Enter the SIP username, password (write-only) and caller number (DID); change the server, transport or number format if needed.
- Press Test connection. Then map your inbound numbers to an assistant. Number mapping →
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