What Is a SIP Trunk? A Plain-English Guide for 2026

A SIP trunk is a virtual phone line that connects your phone system to the public telephone network over the internet. Instead of a copper pair or an E1/T1 circuit running into your building, your PBX sends calls as SIP signalling and RTP audio to a carrier, and the carrier hands them off to mobile and landline networks around the world.

That one sentence covers most of it. The rest of this guide explains what actually happens on a call, what "channels" mean, how SIP compares to PRI, and what you end up paying for.

How a SIP trunk works

Every call on a SIP trunk has two parts that travel separately:

• Signalling — SIP messages that set the call up, ring the far end and tear it down. These are small text messages, a bit like HTTP requests, usually sent on UDP or TCP port 5060 (or 5061 with TLS).
• Media — the actual voice, carried as RTP packets on a range of UDP ports. Each packet holds 20 milliseconds of audio.

A typical outbound call looks like this:

1. Your PBX sends an INVITE to the carrier's SIP proxy with the dialled number, your caller ID and a list of codecs it can use.
2. The carrier checks who you are — either by your source IP address or by a username and password — and checks you have balance.
3. The carrier's routing engine picks an onward route for that destination and forwards the call.
4. You get back 100 Trying, then 180 Ringing or 183 Session Progress (often with early audio for ringback), and finally 200 OK when the person answers.
5. Audio flows in both directions as RTP until one side sends BYE.

Billing starts at the 200 OK and stops at the BYE. Everything before the answer — ringing, busy tones, failed attempts — is normally free.

IP authentication vs registration

There are two common ways a carrier knows a call is yours:

IP authentication is simpler and safer for production because there are no credentials to steal. Registration is handy when you can't guarantee a static address. Our SIP trunking service supports both, and you can run them side by side.

Channels: the unit you size by

A channel is one simultaneous call. If your trunk has 30 channels, you can have 30 calls up at the same moment; the 31st gets rejected, usually with a 503 or 486.

Unlike a PRI, channels on a SIP trunk aren't physical. They're a limit set by the carrier (and by your bandwidth). That has two consequences:

• You can change the number of channels in minutes rather than waiting weeks for a new circuit.
• Some carriers charge per channel each month; others, including us, don't charge for channels at all and simply bill the minutes you use.

As a rough rule, an office needs about one channel for every three to five staff. A call centre running outbound campaigns may need one channel per agent or more. We cover the maths in how many SIP channels you need.

Bandwidth per channel

Each channel needs bandwidth in both directions. With 20 ms packets over Ethernet, real figures including packet overhead are about:

So 30 concurrent G.711 calls need roughly 2.6 Mbps up and 2.6 Mbps down. That's trivial for a modern business line, but upload is usually the bottleneck on consumer connections. See G.711 vs G.729 for when compression is worth it.

SIP trunk vs PRI

PRI (an ISDN E1 in most of the world, T1 in North America) was the standard way to connect a PBX for decades. Most incumbent telcos are switching ISDN off — BT in the UK, Deutsche Telekom in Germany and many others have already done so or have dates set.

The trade-off is that SIP depends on your internet connection. A good setup uses a business-grade line, prioritises voice traffic (QoS), and has a fallback route — either a second connection or call forwarding to mobiles. Our SIP trunk vs PRI article has a migration checklist.

What you actually pay for

SIP trunk pricing varies a lot between providers, but almost every bill is made of the same parts:

• Per-minute rates by destination. A US landline might cost a fraction of a cent; a mobile in a high-termination-fee country can cost many times that. Rates are usually per minute with a billing increment — see below.
• Billing increments. "60/60" means the first minute is charged in full, then every following minute in full. "1/1" means per-second billing. "60/6" means a one-minute minimum, then six-second steps. On short calls the difference is large.
• Numbers (DIDs). Local or toll-free numbers so people can call you. Usually a one-off setup fee plus a monthly fee. See DID numbers.
• Channels or a monthly plan, if your provider charges for them.
• Taxes and regulatory fees in some countries.

At SIPTrunkHub we keep this simple: prepaid balance from $200, no setup fee, no monthly fee and no charge per channel. You pay the per-minute rate for each destination, and the full current list is on our pricing page.

CLI and non-CLI routes

In wholesale VoIP, routes are often sold in tiers depending on whether your caller ID (CLI) reaches the person you're calling intact:

• CLI routes deliver your number to the callee. Calls look legitimate, answer rates are higher, and they're usually required for business traffic.
• Non-CLI routes may replace or drop the caller ID. They're cheaper but answer rates are lower and they're not suitable for anything where the callee needs to call you back.

For a business phone system you almost always want CLI routes. Our explainer on CLI vs non-CLI routes goes through the differences in detail.

Security basics

An internet-facing trunk is a target. Fraudsters scan for open SIP ports, guess extension passwords and then pump expensive international calls through your account — often over a weekend. To protect yourself:

1. Use IP authentication where you can, and restrict SIP to your carrier's IPs at the firewall.
2. Use long, random passwords on every extension and on the trunk itself.
3. Block destinations you never call, especially premium-rate ranges.
4. Set a daily spend limit and channel cap with your carrier.
5. Prefer a prepaid account. If the worst happens, your loss is capped at your balance.

Our toll fraud guide covers each of these in more depth.

Will it work with my PBX?

Almost certainly. SIP is a standard, and every mainstream IP PBX supports it: Asterisk, FreePBX, 3CX, FreeSWITCH, FusionPBX, Yeastar, Grandstream, Cisco CUBE and many more. You typically need five pieces of information from your carrier:

• The SIP server hostname or IP address
• The transport and port (UDP 5060 is the usual default)
• Your authentication details (whitelisted IP, or username and password)
• The number format the carrier expects — usually E.164 without the plus, e.g. 14155550123
• The codecs supported, in order of preference

We publish step-by-step setup guides for the most common platforms, including Asterisk, FreePBX and 3CX.

Choosing a provider

A cheap rate card means little if calls don't connect. When you compare providers, look at:

• Route quality — ask for ASR and ACD figures on the destinations you care about. Our guide to ASR and ACD explains what good looks like.
• Coverage — which countries they terminate directly versus through resellers.
• CLI handling — whether your caller ID is delivered on the routes you'll actually use.
• Support — whether you can reach a real network engineer when something breaks, and how fast.
• Billing terms — increments, minimums, contract length and whether the account is prepaid.

The best test is a real one: put a small amount of balance on, send your actual traffic for a week, and read the CDRs.

Summary

A SIP trunk replaces physical phone lines with an internet connection between your PBX and a carrier. You size it in channels, pay mostly for the minutes you use, and can change capacity in minutes. It's cheaper and more flexible than PRI, works with nearly every modern phone system, and — with sensible security — is the default way businesses connect to the phone network in 2026.

If you'd like to try it, open a prepaid account or check our rates by country.