What Is SIP Trunking? Meaning, How It Works, and When to Use It (2026)

Quick answer: what is SIP trunking?

SIP trunking is a way of connecting a business phone system (a PBX or contact center platform) to the public telephone network over the internet instead of over physical phone lines. SIP stands for Session Initiation Protocol, the standard that sets up, manages and ends voice and video calls over IP. A SIP trunk replaces the copper PRI or analog lines a carrier used to install, carries as many simultaneous calls as your bandwidth allows, and is billed per minute or per channel by a SIP trunking provider such as Signalmash.

This article is part of the Signalmash SIP trunking resource series: what SIP trunking is, the best providers in 2026, what it costs, wholesale SIP for resellers and Twilio Elastic SIP alternatives.

Ask three people what SIP trunking means and you will get a protocol lecture, a sales pitch and a shrug. This guide is the plain-English version: what a SIP trunk actually is, how a call travels across one, how it differs from PRI and hosted VoIP, what you need to run one, and how to judge a provider.

SIP, the protocol, in one paragraph

The Session Initiation Protocol (defined in IETF RFC 3261) is the signalling language of modern telephony. It does not carry the audio itself; it handles the conversation about the call: INVITE (someone wants to talk), 180 Ringing, 200 OK (answered), BYE (hang up). The voice travels separately as RTP packets, encoded with a codec such as G.711 (uncompressed, about 87 kbps per call including packet overhead) or G.729 (compressed, about 32 kbps). SIP is also what your softphone, your video meeting client and most contact center platforms speak under the hood.

What a SIP trunk is

A trunk, in telecom, is a shared path that carries many calls at once between two switches. A SIP trunk is that path implemented as an IP connection between your phone system and your carrier. Instead of a T1/PRI circuit with a fixed 23 voice channels, a SIP trunk is a logical connection that can carry 5 calls or 500, limited only by the bandwidth you have and the concurrent-call limit in your contract. Nothing is physically installed at your office: the carrier authenticates your PBX by IP address or SIP credentials, your PBX sends outbound calls to the carrier's SIP gateway, and the carrier routes inbound calls for your phone numbers back to it.

How a SIP trunk call works, step by step

1. A user dials a number on a desk phone or softphone registered to your PBX (3CX, FreePBX, Asterisk, Cisco, Avaya, Genesys, Five9, Microsoft Teams Direct Routing, and so on).
2. The PBX sends a SIP INVITE to the provider's trunk address, with the called number, your caller ID and the codecs it can use.
3. The provider authenticates the request, checks it against your allowed IPs or credentials, applies STIR/SHAKEN attestation and forwards the call to the terminating carrier that serves the dialled number.
4. The far end answers; a 200 OK comes back and RTP audio starts flowing in both directions.
5. Either party hangs up, a BYE ends the session, and the provider records the duration for billing.

Inbound is the mirror image: a caller dials one of your numbers, the originating carrier hands the call to your SIP provider, and the provider sends an INVITE to your PBX.

SIP trunking vs PRI

A Primary Rate Interface (PRI) is a T1 circuit that carries 23 voice channels plus one signalling channel at 1.544 Mbps. It was the standard for business phone systems for thirty years and it is being retired: incumbent carriers are withdrawing copper and TDM services market by market. The practical differences:

• Capacity: PRI is fixed at 23 calls per circuit; a SIP trunk scales in units of one call, and elastic SIP trunks scale automatically with demand.
• Cost: a PRI costs several hundred dollars a month per circuit plus per-minute long distance; SIP is billed per minute (often under a cent for US calls) or per channel, with no circuit rental.
• Provisioning: a new PRI takes weeks; a SIP trunk is live in hours.
• Portability: a PRI is tied to a building; a SIP trunk follows the PBX wherever it has internet, which is what makes remote and multi-site operation simple.
• Resilience: a PRI has one physical failure point; SIP can fail over to a second internet connection, a second data center, or a mobile number.

SIP trunking vs hosted VoIP (UCaaS)

Both use SIP, but they sit at different layers. With hosted VoIP or UCaaS, the provider runs the phone system for you and sells seats. With SIP trunking, you keep your own PBX or contact center platform and buy only the connection to the telephone network. SIP trunking is the right choice when you already own a PBX you like, need call-center features the hosted product lacks, want to bring your own carrier (BYOC) to a CCaaS platform, or have call volumes where per-seat pricing stops making sense. Our BYOC SIP trunking guide works through that decision for contact centers.

What you need to run a SIP trunk

• A SIP-capable phone system. Nearly every IP-PBX and contact center platform qualifies; older TDM systems need a SIP gateway.
• Bandwidth. Plan roughly 100 kbps per concurrent G.711 call in each direction, with QoS on the router so voice packets are prioritised over file downloads.
• A firewall or session border controller (SBC) that handles SIP NAT traversal and only accepts traffic from the provider's addresses.
• Phone numbers ported to the provider or newly assigned, plus E911 addresses for every location.
• A provider with direct carrier interconnects, STIR/SHAKEN signing, and support that answers when a call fails at 4 p.m. on a Friday.

Security and compliance basics

SIP over TLS encrypts signalling; SRTP encrypts the audio. Restrict the trunk to your PBX's public IP, use strong SIP credentials, set an outbound spend limit, and disable international destinations you never call, because toll fraud on an open trunk can cost thousands overnight. In the US, your provider must sign outbound calls with STIR/SHAKEN so they are not labelled Scam Likely, and Kari's Law and RAY BAUM's Act require direct 911 dialling with dispatchable location from every device.

What SIP trunking costs

US providers publish per-minute rates: outbound calls to the lower 48 states run roughly $0.005 to $0.014 per minute depending on the provider, inbound from $0.003 to $0.009, and phone numbers about $1 a month each. Signalmash lists $0.0068 per minute outbound and $0.0053 inbound on its pay-as-you-go Launch plan with no channel fees up to 10 calls per second. The full comparison is in our SIP trunking pricing guide.

How Signalmash delivers SIP trunking

Signalmash connects directly to AT&T, Verizon, T-Mobile and Bandwidth rather than reselling another aggregator's route, so calls take one hop to the destination carrier. Trunks are elastic (no channel packs to buy), STIR/SHAKEN attestation and branded caller ID are included, failover to a backup destination is configured on the trunk, and support is a Slack channel with an engineer in it, not a ticket queue. See the SIP trunking product page for the feature list.

Talk to a Signalmash voice engineer about your call volumes, PBX and number inventory, and we will quote a SIP trunk with the exact per-minute rates you will pay.

People also ask

What does SIP trunking mean in simple terms?

SIP trunking means running your business phone lines over the internet. A SIP trunk connects your phone system to a carrier over IP, so you no longer need physical phone lines, and you pay per minute or per simultaneous call instead of renting circuits.

What is the difference between SIP trunking and VoIP?

VoIP is the general term for voice over the internet. SIP trunking is a specific use of VoIP: a connection between your own phone system and the public telephone network. Hosted VoIP replaces your phone system entirely and charges per user; SIP trunking keeps your PBX and charges for the connection and minutes.

Is SIP trunking better than PRI?

For almost every business, yes. PRI is fixed at 23 channels per circuit, takes weeks to install, is tied to one building and is being withdrawn by carriers. SIP trunks scale per call, provision in hours, work from any location with internet and cost less per minute.

How many calls can a SIP trunk handle?

As many as your bandwidth and contract allow. Each G.711 call needs about 100 kbps in each direction, so a 100 Mbps connection could in theory carry hundreds of calls. Providers such as Signalmash set the limit in calls per second (CPS) rather than a fixed channel count, and elastic trunks grow automatically with demand.

Do I need a session border controller for SIP trunking?

Not always. Small deployments often work with a SIP-aware firewall and a PBX that handles NAT correctly. An SBC is recommended for contact centers, multi-site PBXs, Microsoft Teams Direct Routing and any environment that needs topology hiding, codec transcoding or strict security policies.