Business VoIP

What Is a SIP Trunk? Explained for UK Business Owners

A SIP trunk replaces traditional ISDN phone lines, connecting your existing on-premise PBX system to the public telephone network over your internet connection. This guide explains how SIP trunking works, when it makes sense, and how it compares to full hosted VoIP.

SIP Trunks: The ISDN Replacement

SIP trunks replace ISDN lines by carrying business phone calls over an IP connection — keeping your existing PBX whilst eliminating expensive ISDN line rental. With BT's PSTN switch-off deadline of 31 January 2027, businesses with ISDN connections must migrate. SIP trunking is typically 30–50% cheaper than ISDN and can be deployed without replacing the existing phone system.

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What Is a SIP Trunk and Why Does It Matter?

As part of our comprehensive business VoIP resource centre, this guide explains SIP trunking in detail — what it is, how it works, when it is the right choice, and how it compares to the alternatives. A SIP (Session Initiation Protocol) trunk is a virtual telephone line that connects a business's on-premise PBX (Private Branch Exchange) phone system to the public telephone network over an internet or dedicated data connection. Where a traditional ISDN line carries voice calls over a physical copper circuit, a SIP trunk carries those same calls as digital data packets over an IP connection.

In practical terms, SIP trunking allows a business to keep its existing PBX hardware and internal phone extensions whilst replacing the expensive ISDN lines that connect the system to the outside world. This makes SIP trunking the most cost-effective migration route for businesses with a relatively modern, SIP-compatible PBX that they do not want to replace as part of the PSTN switch-off. With the UK PSTN switch-off scheduled for completion by January 2027 (Openreach/BT), and over 2 million UK businesses still relying on PSTN/ISDN lines (Ofcom estimates), SIP trunking represents one of the two principal migration paths available.

How SIP Trunking Works: A Technical Overview

Understanding how SIP trunking works requires understanding two key protocols. SIP (Session Initiation Protocol) is responsible for establishing, managing, and terminating voice calls — it handles the signalling that sets up a call between two parties, manages any changes during the call (such as hold or transfer), and terminates the session when the call ends. RTP (Real-time Transport Protocol) handles the actual voice media — the audio data that constitutes the conversation.

A SIP trunk provider allocates a set of virtual channels to your business — each channel supports one simultaneous call. When a member of staff dials an external number, the PBX sends a SIP INVITE message over the data connection to the SIP provider's network. The provider establishes the call to the public telephone network, and the voice audio flows via RTP. Incoming calls to your business phone numbers arrive at the SIP provider and are routed to your PBX over the same data connection.

The PBX continues to handle all internal call routing, hunt groups, auto-attendants, voicemail, and extension management exactly as it always has. The SIP trunk simply replaces the ISDN connection to the external network — the internal operation of your phone system does not change. This is one of the key advantages of SIP trunking: minimal disruption to staff, who continue to use the same phones, the same extensions, and the same features they are accustomed to.

PBX Compatibility

Most modern PBX systems from manufacturers including Avaya, Cisco, 3CX, Mitel, Panasonic, and NEC support SIP trunking natively or with a straightforward software or firmware upgrade. Older PBX systems that only support ISDN interfaces can often be connected to SIP trunks using a media gateway — a device that converts between SIP/IP and the ISDN interface the PBX expects. AMVIA assesses PBX compatibility as part of the migration planning process and recommends the most appropriate approach based on your specific hardware.

SIP Trunking vs ISDN: A Direct Comparison

To understand the benefits of SIP trunking, it helps to compare it directly with the ISDN connections it replaces.

  • Cost: ISDN line rental typically costs £25–£50 per channel per month, depending on whether ISDN2 or ISDN30 is used. SIP channels from a business provider cost significantly less — often 50–70% lower per channel — with lower per-minute call rates as well. For a business with 30 ISDN channels, the monthly saving on line rental alone can exceed £500.
  • Scalability: Adding ISDN channels requires an engineer visit, physical wiring changes, and a lead time of days or weeks. Adding SIP channels is a software configuration change that takes effect within hours — and channels can be removed just as easily if demand decreases.
  • Resilience: ISDN lines are tied to a physical copper circuit. If that circuit is damaged — by roadworks, flooding, or equipment failure — calls are lost until the circuit is repaired. SIP trunks can be configured with automatic failover to a secondary internet connection (such as 4G/5G backup) or to divert calls to mobile numbers, providing resilience that ISDN cannot match.
  • Geographic flexibility: ISDN lines are provisioned to a specific physical location. SIP trunks are not tied to a location — if your business relocates, the SIP trunk moves with you without needing new line installations. Multiple office locations can share a single SIP trunk service.
  • Future-proofing: ISDN is being permanently switched off in January 2027. SIP trunking is the replacement technology and will continue to be supported and developed for the foreseeable future.

SIP Trunking vs Hosted VoIP: When to Choose Which

SIP trunking and hosted VoIP (also known as UCaaS — Unified Communications as a Service) are both valid migration paths from ISDN, but they suit different circumstances.

SIP trunking retains your on-premise PBX hardware. The trunk provides the external connection, but your existing system still manages everything internally. This is the right choice if your PBX is relatively modern (typically less than five to seven years old), has features your business relies on and staff are trained on, is SIP-compatible (or can be made compatible with a gateway), and has useful remaining life — making replacement premature and wasteful.

Hosted VoIP or UCaaS replaces the entire phone system — PBX hardware and ISDN lines — with a cloud platform. There is no on-premise equipment to manage. Staff access the system via IP desk phones, softphone applications on laptops, or mobile apps. Microsoft Teams Phone is the most widely deployed example for Microsoft 365 users. Hosted VoIP has lower upfront cost (no PBX hardware to purchase), is inherently more flexible for remote and hybrid working, and removes the ongoing responsibility of managing PBX hardware, firmware updates, and maintenance contracts.

For businesses with an ageing PBX that is approaching end of life — particularly systems more than seven or eight years old, or those with expensive maintenance contracts — hosted VoIP is typically the more cost-effective long-term solution. AMVIA assesses both options objectively and recommends the approach that delivers the best value based on your specific situation, rather than defaulting to one path.

Capacity Planning: How Many SIP Channels Do You Need?

One of the most common questions during SIP trunk migration planning is how many channels to provision. The answer depends on your peak concurrent call volume — not your total number of extensions or staff.

A SIP channel supports one simultaneous call. If your business has 50 phone extensions but the maximum number of calls happening at the same time during peak periods is 15, you need 15 SIP channels — not 50. Over-provisioning wastes money; under-provisioning means callers get a busy signal during peak times.

AMVIA analyses your current ISDN call data — which your existing provider can supply — to determine your actual peak concurrent call volume. This data-driven approach ensures you provision the right number of channels from day one, avoiding both the cost of excess capacity and the service impact of insufficient capacity. Channels can be adjusted up or down after deployment if your calling patterns change.

Connectivity Requirements for SIP Trunking

SIP trunks rely on your internet connection to carry voice calls, so connectivity quality directly affects call quality. The bandwidth requirement is modest — each concurrent call uses approximately 80–100 Kbps of upload and download bandwidth when using the G.711 codec. For a business with 15 concurrent calls, that equates to roughly 1.5 Mbps — well within the capacity of most FTTC or FTTP broadband connections.

However, bandwidth alone does not guarantee acceptable call quality. Three additional metrics are critical: 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 be configured to prioritise voice traffic over bulk data transfers such as file downloads, cloud backups, and web browsing. Without QoS, a large file download or cloud sync operation can consume available bandwidth and degrade call quality during peak data usage periods.

For businesses with high call volumes or where call quality is business-critical, a dedicated leased line or Ethernet connection provides symmetric bandwidth with guaranteed quality parameters — eliminating the contention and variability inherent in broadband connections. AMVIA assesses your connectivity as part of the SIP trunk migration process and recommends whether your current connection is suitable or whether an upgrade is advisable.

Number Porting and Migration Process

Your existing geographic phone numbers — the numbers your clients already know and use — can almost always be ported to a new SIP trunk provider. Number porting is the process of transferring ownership of your numbers from your current ISDN provider to the SIP trunk carrier. The process typically takes 7 to 14 working days, during which calls continue to route through your existing ISDN lines until the port completes — there is no gap in service if the port is managed correctly.

The migration process for SIP trunking typically follows these steps: PBX compatibility assessment; SIP trunk provisioning and testing with temporary numbers; QoS configuration on your router and network; number porting from your current ISDN provider; cutover and go-live; and post-migration monitoring. AMVIA manages this entire process as part of its SIP trunk deployment service, coordinating with your existing provider and ensuring a smooth transition.

SIP Trunk Pricing: What to Expect

SIP trunk pricing for UK businesses typically consists of two components: a per-channel monthly rental and per-minute call charges. Channel rental costs vary by provider but are significantly lower than equivalent ISDN rental — often £5–£15 per channel per month compared to £25–£50 for ISDN. Call charges are also lower, with many SIP providers offering inclusive UK landline and mobile minute bundles.

For a business migrating from a 30-channel ISDN30 circuit with line rental of approximately £1,000–£1,500 per month, the equivalent SIP trunk service might cost £150–£450 per month for the same channel capacity — representing a saving of 50–70% on line rental alone. When lower call charges are factored in, the total saving is often sufficient to recover any upfront migration costs within 6 to 12 months.

Key Considerations for UK SMEs

  • Verify PBX compatibility before committing: Not all PBX systems support SIP natively. Some require a gateway device or firmware upgrade. AMVIA assesses compatibility as part of the planning process — there is no charge for this initial assessment.
  • Test your internet connection under load: Have your broadband tested for bandwidth, jitter, and packet loss during peak usage periods — not just at quiet times. Real-world performance under load determines whether your connection will support SIP calls reliably.
  • Configure QoS before going live: Quality of Service settings on your router are essential, not optional. Without them, call quality will suffer when other applications compete for bandwidth.
  • Plan number porting with adequate lead time: Allow a minimum of two weeks for number porting. As migration demand increases through 2026 ahead of the PSTN switch-off, porting queues may lengthen — plan early.
  • Consider disaster recovery: Configure automatic call divert to mobile numbers if your internet connection fails. For critical operations, a secondary internet connection (4G/5G) providing automatic failover adds an additional layer of resilience.

How AMVIA Can Help

AMVIA provides fully managed SIP trunking services for UK businesses, including PBX compatibility assessment, connectivity evaluation, channel planning based on actual call data, number porting, QoS configuration, and ongoing management. Where a business is also considering a full hosted VoIP or Microsoft Teams Phone migration, AMVIA will assess both options and make an honest recommendation based on your existing system, team size, budget, and long-term requirements. Call 0333 733 8050 to discuss your PSTN migration options.

SIP Trunking: Key Features

What SIP trunks provide compared to traditional ISDN connections.

ISDN Replacement

Drop-in replacement for ISDN2 and ISDN30 lines — keep your existing PBX and phone numbers.

Flexible Channel Scaling

Add or remove simultaneous call channels without engineer visits — scale with your business.

Cost Reduction

Typically 30–50% lower cost than ISDN line rental with lower per-minute call rates.

Number Portability

Retain existing geographic phone numbers — no disruption to clients who already know your number.

SIP Trunk Implementation Checklist

What to confirm before migrating from ISDN to SIP trunking.

PBX compatibility confirmed

Existing PBX supports SIP natively or via a compatible gateway device.

Broadband bandwidth assessed

Upload speed sufficient for maximum expected concurrent calls — typically 1–2Mbps for 10–20 channels.

QoS configured on router

Voice traffic prioritised to prevent call quality issues under load.

Number porting planned

All geographic numbers identified and porting arranged — 7–14 working days lead time.

Channel count determined

Number of simultaneous call channels based on peak usage analysis — not total extensions.

Disaster recovery considered

Failover plan in place if internet connection fails — calls can divert to mobiles automatically.

SIP Trunk FAQs

Migrate from ISDN to SIP Trunks

AMVIA will assess your existing PBX, check compatibility, and manage the full SIP trunk migration — including number porting and QoS configuration.