What Is a SIP Trunk? Explained for UK Business Owners

What is a SIP trunk and why does it matter?

A SIP (Session Initiation Protocol) trunk carries your business calls as digital data packets over an IP connection instead of a physical copper ISDN circuit. It lets you keep your current PBX hardware and internal extensions while swapping only the expensive lines that connect that system to the outside world.

This makes SIP trunking the most cost-effective migration route for a business with a reasonably modern, SIP-compatible PBX it does not want to replace. With the UK PSTN switch-off scheduled for completion by 31 January 2027 and over 2 million UK businesses still relying on PSTN/ISDN lines (Ofcom estimates), SIP trunking is one of the two main paths off the old network — the other being a full move to a hosted phone system. For the wider picture, see our business VoIP solutions and what the PSTN switch-off means for your phones.

Ofcom's guidance on migrating landlines to digital technology confirms the analogue and ISDN network is being retired industry-wide.

How does SIP trunking work?

SIP trunking uses two protocols. SIP sets up, manages and ends each call — the signalling that connects two parties, handles hold or transfer, and closes the session. RTP (Real-time Transport Protocol) carries the actual voice audio. Together they replace what an ISDN line used to do over copper.

A provider allocates a set of virtual channels to your business, and each channel supports one simultaneous call. When a member of staff dials out, the PBX sends a SIP INVITE over your data connection to the provider, which completes the call to the public network; the audio then flows via RTP. Incoming calls arrive at the provider and route to your PBX over the same link.

Crucially, the PBX still handles all internal routing, hunt groups, auto-attendants, voicemail and extensions exactly as before. Only the external connection changes, so staff keep the same phones, extensions and features. Most modern PBX systems from Avaya, Cisco, 3CX, Mitel, Panasonic and NEC support SIP natively or after a firmware upgrade; older ISDN-only systems can connect through a media gateway. AMVIA checks PBX compatibility as part of migration planning.

SIP trunking vs ISDN: what is the difference?

SIP trunking beats ISDN on cost, scalability, resilience and geographic flexibility, and it is the only one of the two with a future — ISDN is being switched off permanently. ISDN ties calls to a fixed copper circuit; SIP moves them onto an IP connection you can scale and re-route in software.

For a business running 30 ISDN channels, the saving on line rental alone can exceed £500 a month.

SIP trunking vs hosted VoIP: which should you choose?

SIP trunking keeps your on-premise PBX; hosted VoIP (UCaaS) replaces it entirely with a cloud platform. Choose SIP if your PBX is modern, SIP-compatible and has useful life left. Choose hosted VoIP if your system is near end of life or you want cloud flexibility for hybrid teams.

If your PBX is more than seven or eight years old, or carries an expensive maintenance contract, a move to UCaaS for UK SMEs or Microsoft Teams Phone is usually the better long-term call. AMVIA assesses both objectively rather than defaulting to one path.

How many SIP channels does your business need?

The number of SIP channels you need equals your peak concurrent call volume — not your total extensions or headcount. One channel supports one simultaneous call, so a 50-extension office that never exceeds 15 calls at once needs 15 channels, not 50.

Over-provisioning wastes money; under-provisioning gives callers a busy tone at peak times. AMVIA analyses your existing ISDN call data — which your current provider can supply — to size the trunk accurately from day one. Channels can be flexed up or down later as your calling patterns change, without an engineer visit.

What connectivity do you need for SIP trunking?

Each concurrent call uses roughly 80–100 Kbps up and down on the G.711 codec, so 15 calls need only about 1.5 Mbps — well within most FTTC or FTTP broadband. Bandwidth is rarely the limiting factor; connection stability matters more.

Three metrics decide call quality: latency (ideally below 150 ms), jitter (ideally below 30 ms) and packet loss (ideally below 1%). Quality of Service (QoS) settings on your router must prioritise voice over bulk traffic such as backups and downloads, or a single large file transfer can degrade calls. Where call quality is business-critical, a dedicated leased line gives symmetric bandwidth with guaranteed quality, removing the contention inherent in shared broadband. AMVIA assesses your connection before recommending whether it is fit for SIP.

How much does a SIP trunk cost?

SIP trunk pricing has two parts: a per-channel monthly rental and per-minute call charges. Channel rental typically runs £5–£15 per channel per month, against £25–£50 per channel per month for ISDN, and call rates are lower, often with inclusive UK landline and mobile bundles.

A business migrating from a 30-channel ISDN30 circuit with line rental around £1,000–£1,500 a month might pay £150–£450 a month for equivalent SIP capacity — a saving of 50–70% on line rental alone. With lower call charges added in, the total saving often recovers any upfront migration cost within 6 to 12 months. Number porting takes 7 to 14 working days, with no gap in service when managed correctly.

How does AMVIA manage SIP trunk migration?

AMVIA delivers fully managed SIP trunking for UK businesses: PBX compatibility assessment, connectivity evaluation, channel planning from real call data, number porting, QoS configuration and ongoing management. Where a full hosted VoIP or Teams Phone move makes more sense, we say so.

The migration follows a clear sequence:

• PBX compatibility assessment — confirm native SIP support or specify a gateway.
• Provisioning and testing — stand up the trunk on temporary numbers.
• QoS configuration — prioritise voice on your router and network.
• Number porting — transfer your geographic numbers (7–14 working days).
• Cutover and go-live — switch traffic with calls still protected.
• Post-migration monitoring — verify quality and tune as needed.

Because voice now rides your data network, securing it matters: see our approach to VoIP security, aligned with NCSC guidance on telephony and network security. Call 0333 733 8050 to discuss your PSTN migration options.