SIP trunking explained: how SIP trunks replace traditional phone lines

Who this guide is for

Audience: IT managers replacing legacy PRI or analogue phone lines.

Understand how VoIP calling works — SIP, PBX, codecs, trunking, DID numbers, STIR/SHAKEN, and the protocols behind business phone systems.

Use this guide when you want the setup to be correct the first time and easy for another admin, manager, or supervisor to verify later.

What this workflow helps you accomplish

SIP trunking replaces traditional analogue phone lines or PRI circuits with virtual connections over the internet. A SIP trunk carries multiple concurrent voice channels through a single IP connection to your PBX or phone system, eliminating per-line hardware costs.

This workflow matters because numbers, routing, access, and reporting in CallOrbit are connected. Skipping one setup detail usually creates avoidable support work later.

• Step 1: Understand what a SIP trunk actually replaces: a single SIP trunk replaces multiple physical copper lines or PRI circuits. Each trunk can carry multiple simultaneous calls depending on the bandwidth and channel allocation your provider provisions.
• Step 2: Compare SIP trunking to PRI: PRI (Primary Rate Interface) provides 23 voice channels plus one signalling channel over a dedicated T1 line. SIP trunking provides unlimited channels over your existing internet connection with no dedicated circuit required.
• Step 3: Know the cost advantage: SIP trunking eliminates monthly rental costs for physical copper lines, PRI circuits, and ISDN maintenance contracts. SIP trunks are billed per channel or per concurrent call path, and you pay only for what you actually use.
• Step 4: Configure SIP trunk security: restrict SIP trunk registration to static IP addresses or IP ranges, enable authentication, apply least-privilege firewall rules, and monitor for unexpected call patterns. Unsecured SIP trunks are a common vector for toll fraud attacks.
• Step 5: Plan for failover and redundancy: configure multiple SIP trunks across separate internet connections or providers so inbound and outbound calling continues if one link fails. A well-designed SIP trunking setup includes automatic failover routing.

Setup checklist

• Understand what a SIP trunk actually replaces: a single SIP trunk replaces multiple physical copper lines or PRI circuits. Each trunk can carry multiple simultaneous calls depending on the bandwidth and channel allocation your provider provisions.
• Compare SIP trunking to PRI: PRI (Primary Rate Interface) provides 23 voice channels plus one signalling channel over a dedicated T1 line. SIP trunking provides unlimited channels over your existing internet connection with no dedicated circuit required.
• Know the cost advantage: SIP trunking eliminates monthly rental costs for physical copper lines, PRI circuits, and ISDN maintenance contracts. SIP trunks are billed per channel or per concurrent call path, and you pay only for what you actually use.
• Configure SIP trunk security: restrict SIP trunk registration to static IP addresses or IP ranges, enable authentication, apply least-privilege firewall rules, and monitor for unexpected call patterns. Unsecured SIP trunks are a common vector for toll fraud attacks.
• Plan for failover and redundancy: configure multiple SIP trunks across separate internet connections or providers so inbound and outbound calling continues if one link fails. A well-designed SIP trunking setup includes automatic failover routing.

Operational follow-up

After you complete this flow, confirm the live experience from both the agent and customer side so ownership, routing, permissions, and reporting all match what the workspace expects.

If your team is rolling this out across multiple users, queues, or phone numbers, pair this article with the broader knowledge base and the relevant routing or numbers guides to keep deployment consistent.

• What is the CallOrbit Knowledge Base for? — It is the public help hub for how CallOrbit works, covering numbers, webphone setup, SIP, extensions, routing, users, roles, and billing basics.
• Can customers read this without signing in? — Yes. The Knowledge Base now lives on a public route so customers can read setup guidance before or after they enter the portal.
• Does the portal still have its own Knowledge Base page? — No. The signed-in portal navigation no longer carries a separate Knowledge Base page, and the old portal path now redirects to this public version.
• What is VoIP and how does it work? — VoIP (Voice over Internet Protocol) converts analogue voice signals into digital packets and transmits them over IP networks. Unlike traditional PSTN phone lines that require dedicated copper wiring per line, VoIP calls use your existing internet connection, which makes them cheaper, more flexible, and easier to scale.
• What is SIP trunking? — SIP trunking is a virtual connection that replaces traditional analogue phone lines or PRI circuits. A SIP trunk carries multiple concurrent voice channels over a single IP connection to your PBX or phone system, eliminating per-line hardware costs and monthly line rental fees.
• What is the difference between hosted PBX and cloud PBX? — Hosted PBX runs on dedicated virtual infrastructure managed by a provider, while cloud PBX uses shared multi-tenant cloud infrastructure. Hosted PBX suits organisations needing custom configuration and predictable pricing. Cloud PBX is better for instant scalability and per-user monthly billing.
• What is a DID number? — A DID (Direct Inward Dialling) number is a virtual phone number that routes directly to a specific extension, IVR menu, queue, or user within a phone system without an operator. DIDs decouple the phone number from the physical phone line, so you can have hundreds of numbers routed through a single SIP trunk.
• What are G.711, Opus, and G.729 codecs used for? — These are VoIP codecs that convert voice into digital data. G.711 uses 64 Kbps for toll-grade quality and is the PSTN standard. Opus uses 6-510 Kbps and adjusts to network conditions. G.729 uses 8 Kbps for bandwidth-constrained links. The right codec depends on your available bandwidth and call quality requirements.

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