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SIP trunking fundamentals: how business calls reach the carrier

Updated August 17, 2026

A SIP trunk is the logical connection that lets a private phone system exchange calls with a carrier over IP. It usually includes signalling, media, number presentation, routing rules, capacity limits, and security controls.

The call path


phone -> PBX -> SBC or edge firewall -> SIP carrier -> PSTN/mobile network

The PBX decides how to route the call. The carrier validates the account, number, destination, and permissions. RTP carries the audio separately from SIP signalling.

What to specify when ordering a trunk

Area	Questions to answer
Authentication	IP-based, digest credentials, certificates, or a combination?
Transport	UDP, TCP, or TLS for SIP? Is SRTP available for media?
Numbers	Which DIDs, inbound routes, caller IDs, and emergency records are included?
Capacity	How many concurrent sessions are permitted in each direction?
Codecs	Which codecs are accepted, and where does transcoding occur?
Media	What RTP ranges and public addresses are used?
Resilience	Are there secondary hosts, DNS SRV records, or geographic failover paths?
Security	Which source IPs, ports, fraud controls, and rate limits apply?

Common failure boundaries

403 often points to identity, permissions, caller ID, or destination policy.
404 or 484 may indicate a number formatting or routing problem.
488 often indicates an incompatible media offer.
A connected call with no audio usually requires an RTP/NAT investigation.