What Is SIP Trunking?
SIP (Session Initiation Protocol) trunking delivers voice calls over IP networks instead of traditional TDM circuits (PRI, POTS). SIP trunks connect an on-premise PBX or session border controller to a carrier's voice network via the internet or a dedicated connection. Each SIP trunk supports one concurrent call, and capacity scales instantly without physical circuit installation.
SIP Trunking vs. PRI
A PRI circuit delivers exactly 23 concurrent call channels over a dedicated T1 line at a fixed monthly cost regardless of usage. SIP trunking offers flexible channel counts (add or remove in minutes), per-minute or per-channel pricing options, and geographic number portability. Organizations typically save 30–50% by replacing PRIs with SIP trunks while gaining flexibility and disaster recovery capabilities.
Architecture Considerations
SIP trunks require a Session Border Controller (SBC) at the network edge for security, interoperability, and media handling. Quality of Service (QoS) configuration on your network must prioritize voice packets over data traffic. For reliability, deploy SIP trunks over dedicated internet with a broadband or cellular failover path.
When to Choose SIP Trunking
SIP trunking is ideal when you have a functioning on-premise PBX that you want to keep but need to reduce voice circuit costs, when you need to consolidate voice services across multiple locations to a central PBX, or as a bridge technology during a phased migration to UCaaS.
Common Pitfalls
Not provisioning sufficient bandwidth for concurrent calls causes quality degradation under load. Choosing a SIP trunk provider without verifying codec compatibility with your PBX leads to one-way audio or call drops. Failing to implement an SBC exposes your PBX to SIP-based attacks and toll fraud.