What is SIP Trunking?
SIP trunking is how a modern business phone system connects to the outside world. Instead of physical lines running into the building, your phone system talks to a SIP provider over the internet using the Session Initiation Protocol — and that provider hands your calls off to the public phone network. With BT's PSTN switch-off coming, it's now the default route for almost every UK business.
What SIP Trunking Is
SIP trunking is how you connect a phone system to the rest of the world over the internet rather than through physical lines. SIP — Session Initiation Protocol — is the technical standard for setting up, managing, and tearing down voice and video sessions over IP networks. "Trunk" is borrowed from old telephony, where it meant the bundle of physical lines linking your building to the telephone exchange.
So a SIP trunk is a virtual version of that bundle. Your PBX (or cloud phone platform) talks to a SIP provider over your existing broadband, and the SIP provider handles the bit that connects you to everyone else's phone number. Your numbers don't change. The agent's experience doesn't change. The infrastructure underneath does.
How It Works
To see why SIP trunking matters, picture what it replaces. Traditional phone systems connect to the public network through physical lines — either analogue (one call per line) or ISDN circuits carrying multiple calls. Those lines run from the local exchange to a box in your building and you pay for them whether anyone's on a call or not.
The SIP Version
With SIP trunking, your phone system — usually an IP-PBX — connects to a SIP trunking provider over the same internet connection you use for everything else. The SIP protocol handles the signalling: setting calls up, transferring, putting people on hold, joining conferences. The voice audio itself travels as data packets, mixed in alongside your email and web traffic.
One SIP trunk can carry many simultaneous calls. With physical lines you needed one line per concurrent call; with SIP you buy "channels" and scale up or down on demand. If you double your headcount next month you don't need to wait six weeks for BT to install more copper.
Codecs and Bandwidth
Voice has to be converted to digital data, which is what codecs do. The two you'll see most often are G.711 (high quality, around 80–100 Kbps per call) and G.729 (lower quality, around 30 Kbps). G.711 is the default for calls inside your own network where bandwidth is cheap; G.729 still shows up on calls traversing the wider internet where bandwidth is tight.
Why Businesses Moved To It
The shift from ISDN to SIP didn't happen because anyone fell in love with the technology — it happened because the numbers stacked up.
- SIP trunking is typically 30–60 per cent cheaper than equivalent ISDN or analogue, particularly if you do any international calling.
- You can add channels in minutes instead of waiting weeks for BT engineers.
- Your phone numbers come with you when you move premises or change providers.
- A business in Newcastle can have an 020 London number without anybody actually being in London.
- If your office burns down, your numbers can be rerouted to mobiles or another site within minutes.
- BT's PSTN switch-off — originally targeted for the end of 2025, now pushed into 2027 — means traditional ISDN and analogue lines are being withdrawn. SIP is what you're moving to whether you want to or not.
In practice the migration is now well underway across the UK. New offices don't get ISDN installed. SIP is the default.
SIP Trunking and Telephone Payments
For a business taking card payments over the phone, SIP trunking is good news and a new set of problems in equal measure. The good news first. Because the call is already travelling as data, it's straightforward to route it through a secure payment platform — for DTMF masking or IVR card capture — without bolt-on hardware. You're not splicing into a copper line; you're sending SIP packets to one more network destination.
The problem: that same voice traffic is now travelling over the open internet, and card digits keyed during a call are packets that could in theory be intercepted. Anyone running a SIP trunk that carries payment audio should be using TLS for the SIP signalling and SRTP for the voice itself. If your provider doesn't offer both, find a different provider.
There's a quality angle too. If the broadband connection drops a packet mid-keypress, you can lose a digit, and the customer ends up declined for no reason they can see. Pretty soon you've got a frustrated caller and an agent trying to talk them through restarting a payment. The fix is unglamorous: enough bandwidth to handle peak concurrent calls with headroom, plus QoS on your network to prioritise voice over file downloads.
If You're Migrating, Watch For These
- Bandwidth: roughly 80–100 Kbps per concurrent call on G.711. Add up your peak and add at least 30 per cent on top.
- QoS: prioritise voice packets on your LAN and WAN. Without it, a 200MB download will degrade every call in the office.
- Encryption: insist on TLS and SRTP, particularly if you handle payments or any other sensitive content.
- Backup connectivity: if your single internet line dies, every phone in the building dies with it. A second connection from a different carrier is cheap insurance.
- Phased migration: don't cut over every line on a Friday. Pilot with one team, find the problems, then move the rest.
SIP trunking isn't new or experimental anymore. It's the standard way UK and US businesses connect to the phone network in 2026, and the only real question is which provider you pick and how you protect the traffic running across it.
Paytia's PCI DSS Level 1 certified platform incorporates sip trunking as part of its thorough security approach. By processing phone payments through DTMF suppression, Paytia ensures card data is protected at every stage.
Frequently Asked Questions
What is sip trunking?
It's a way of delivering phone service over the internet instead of through physical lines. Your phone system talks SIP — Session Initiation Protocol — to a provider over your existing broadband, and that provider connects your calls to the rest of the phone network.
Why is sip trunking important for PCI DSS?
On its own, SIP trunking doesn't make you compliant — it's just a transport. The PCI angle is that voice on a SIP trunk is data packets on the public internet, so anyone handling payment calls over SIP needs TLS on the signalling and SRTP on the audio, plus a way of keeping card digits out of the agent's environment (which is where DTMF masking comes in).
How does Paytia handle sip trunking?
Paytia plugs into your existing SIP setup — Genesys, Five9, NICE CXone, Amazon Connect, plain SIP, whatever you're already running — at the SIP layer. We don't need to replace your trunking provider; we sit in the audio path long enough to mask the DTMF tones and capture the card digits on a separate encrypted channel to the gateway.
See how Paytia handles sip trunking
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