SIP trunking is a carrier service that connects a business phone system to the public switched telephone network (PSTN) over an IP connection. It replaces analog lines and PRI circuits: instead of physical channels to the phone company, calls travel as Session Initiation Protocol (SIP) signaling and voice packets over the internet or a private network. You keep your own phone system, on site or in your own cloud environment, and the carrier provides phone numbers, outside calling and emergency call routing.
At a glance
- SIP trunking connects a PBX, contact center or Microsoft Teams (through Direct Routing) to the public phone network over IP.
- Capacity is sold per concurrent call path or as a pooled or burstable allowance, rather than in fixed 23- or 30-channel circuits.
- Call quality depends on the network connection: bandwidth, latency, jitter and packet loss.
- A session border controller (SBC) often sits between your phone system and the carrier for security and interoperability.
- Emergency calling, number porting and failover need to be designed, not assumed.
What problem it solves
Traditional trunks were analog lines or PRI circuits, sold in fixed blocks and delivered to one building. In North America, a site that needed 30 call paths bought two PRIs and paid for 46; if the circuit failed, calls to that site stopped. Adding capacity meant waiting for a new circuit to be installed.
SIP trunks deliver the same connection to the phone network over IP. Depending on the carrier, you can size capacity to actual concurrent calls, add paths faster than ordering a new circuit, use one carrier across many sites, and set up rerouting of inbound calls to another site or carrier if one location goes down. Because it runs over data connections the business already has or can add, it is also the usual path when carriers retire PRI and copper services.
How it works
The carrier provides a SIP trunk: an account on its voice network, a set of phone numbers and the right to place a certain number of concurrent calls. Your phone system (or an SBC in front of it) connects to the carrier’s SIP servers, either by registering with credentials or by a static connection between known IP addresses.
- Signaling. When someone dials out, the PBX sends a SIP INVITE to the carrier, which routes the call onward through the PSTN. Inbound calls to your DID numbers arrive the same way in reverse.
- Media. The voice travels as RTP packets, or SRTP if encryption is enabled, between your system and the carrier.
- The edge. A session border controller or a firewall with voice support protects the phone system, handles address translation and fixes differences between SIP implementations. Some carriers include a managed SBC.
Connection type. SIP trunks can run over the public internet, an internet connection from the same carrier, or a private connection such as MPLS. A private connection gives the carrier more control over quality; the public internet is cheaper and more flexible. Either way, size bandwidth for peak calls and set voice prioritization on your network.
Emergency calling. Each number needs a registered address, and multi-site or remote users need correct location handling. See E911 for how location rules apply to business phone systems; requirements differ by country and, in the US, by state.
Caller ID. In the US and Canada, carriers authenticate caller ID using STIR/SHAKEN. The attestation your calls receive depends on whether the carrier can verify you are entitled to use the number, which matters if you send calls with numbers from another provider.
When it matters for buyers
- When PRI or analog lines are being retired or repriced at one or more sites.
- When moving to Microsoft Teams Phone with your own carrier, through Direct Routing.
- When consolidating carriers across many locations onto one contract.
- When call capacity is mismatched to use, either paying for idle channels or hitting busy signals at peak.
- When resilience is a concern, since SIP trunks can fail over between sites, SBCs and carriers.
Questions to ask vendors
- Do you price per concurrent call path, by usage or as an unlimited bundle, and what happens when we exceed capacity?
- Which PBXs and SBCs have you certified or tested with your service, including ours?
- Do you support TLS and SRTP encryption, and over which connection types?
- How do you handle 911 for our sites and any remote users, and how do we update addresses?
- What failover options exist for inbound calls if our site or SBC is unreachable?
- What attestation do our outbound calls receive, and how do you help prevent them being labeled as spam?
- What are the rates for domestic, international and toll-free calling, and which fees and taxes apply?
- How long will porting our numbers take, and who manages the cutover?
How it differs from PRI
A PRI is a dedicated digital circuit with a fixed 23 or 30 channels. A SIP trunk does the same job over IP, with capacity sized to need and the ability to spread calls across multiple paths and locations. PRI quality is predictable because the channels are reserved; SIP trunk quality depends on how well the IP connection is sized and managed.
How it differs from SIP and hosted PBX
SIP is the protocol that SIP trunks use; you can run SIP phones without buying SIP trunks. A hosted PBX replaces the phone system itself and includes the connection to the phone network, so you don’t buy SIP trunks separately.
For buying guidance, see our SIP trunking solution page and Planning For a Successful SIP Trunking Implementation.
Problems this helps with
- Our PRI or analog lines are being retired or getting expensive
- Paying for more call capacity than we use
- Inbound calls stop when one circuit or site goes down
- Connecting Teams or a PBX to a carrier of our choice
