SIP Trunk
A SIP trunk is the carrier-side endpoint that connects your PBX (or AI voice agent, or softphone) to the PSTN over SIP. It is the modern replacement for the analog or T1/E1 'trunk' that PBXs used to have, and it is how every cloud-native telecom service reaches the phone network.
What a SIP trunk provides
- Outbound calling to any E.164 destination, billed per minute.
- Inbound calling on one or more DIDs assigned to the trunk.
- SIP signaling endpoint (SIP server FQDN + port) and authentication mode (digest or IP-ACL).
- Concurrent-call capacity (channels), usually metered or capped per plan.
Authentication modes
- SIP digest auth: username + password. Works behind any NAT. Required for dynamic IPs.
- IP-ACL: the customer's static public IP is pinned. No registration overhead. Recommended for cloud / static deployments.
Routing model
- Outbound: PBX places call → SIP trunk → carrier's PSTN gateway → destination.
- Inbound: caller dials your DID → PSTN → carrier → SIP trunk → PBX → extension/IVR/agent.
Related terms
PBX (Private Branch Exchange)
SBC (Session Border Controller)
BYOC (Bring Your Own Carrier)
DID (Direct Inward Dialing)
SIP Authentication (Digest vs IP-ACL vs mTLS)
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