PSTN Switch-Off 2027: Your ISDN to VoIP Migration Guide

The clock is ticking for every UK business still relying on traditional telephone lines. The PSTN switch off 2027 UK deadline is no longer a distant event on the horizon — it is an imminent reality that will affect millions of businesses, from one-person consultancies in central London to multi-site enterprises across the United Kingdom. Openreach, the infrastructure arm of BT Group, has confirmed that the Public Switched Telephone Network and all Integrated Services Digital Network (ISDN) services will be permanently decommissioned by January 2027. After that date, no new ISDN lines will be available, no repairs will be carried out on legacy circuits, and every voice call in the UK will travel over Internet Protocol networks.

If your organisation still depends on ISDN lines, analogue phone lines, or any PSTN-based telephony, the time for your ISDN to VoIP migration UK project is now. This comprehensive guide from Cloudswitched, London's trusted IT managed service provider, walks you through every aspect of the transition — from understanding the technical background and the official switch-off timeline to selecting the right SIP trunking providers UK businesses depend on, planning your migration, porting your numbers, testing your new system, and ensuring business continuity throughout. Whether you need VoIP migration services for a single office or a nationwide rollout, this guide gives you the knowledge to make confident, informed decisions.

Understanding the PSTN Switch-Off: What Is Actually Happening?

To appreciate the urgency of the PSTN switch off 2027 UK deadline, it helps to understand what the Public Switched Telephone Network actually is and why it is being retired. The PSTN is the traditional circuit-switched telephone network that has served the United Kingdom since the late nineteenth century. It relies on a vast infrastructure of copper cables, telephone exchanges, and switching equipment to establish dedicated circuits between callers. For over a hundred years, this network has been remarkably reliable — but it is now reaching the end of its operational and economic life.

The copper cables that form the backbone of the PSTN are ageing, expensive to maintain, and increasingly difficult to source replacement parts for. The exchange equipment, much of which dates from the 1980s and 1990s, requires specialist engineers whose skills are retiring along with them. Openreach has made a strategic decision: rather than continuing to invest billions of pounds in maintaining infrastructure that was designed for a pre-internet era, it will redirect investment toward full-fibre broadband and IP-based services that serve the needs of modern businesses and consumers.

ISDN — the Integrated Services Digital Network — sits on top of the PSTN infrastructure. ISDN was introduced in the 1980s as an upgrade to basic analogue lines, offering digital connectivity with dedicated channels for voice and data. ISDN2e provides two 64 Kbps channels (commonly used by small businesses), whilst ISDN30 provides up to 30 channels for larger organisations. Both variants rely on the same underlying PSTN switching equipment and copper infrastructure, which means both are being retired together.

The Official Timeline

The PSTN switch off 2027 UK programme has been years in the making, with Openreach and the wider telecoms industry working to a phased timeline. Understanding where we are in this process is critical for planning your ISDN to VoIP migration UK project:

Who Is Affected?

The short answer is: every business in the United Kingdom that uses a telephone. If your organisation has any of the following, your ISDN to VoIP migration UK project is mandatory, not optional:

• ISDN2e lines — commonly used by small and medium businesses for 2–8 voice channels
• ISDN30 circuits — used by larger organisations for 10–30 concurrent call channels
• Analogue PSTN lines — single traditional phone lines, often used for fax machines, door entry systems, alarm panels, lift emergency phones, and EPOS card payment terminals
• Multiline analogue services — hunt groups and DDI ranges delivered over copper
• Any device connected to the copper phone network — including franking machines, building management systems, CCTV with PSTN diallers, and redcare alarm monitoring

It is not just voice calls that are affected. Many businesses have equipment and systems that rely on PSTN connectivity without even realising it. A thorough audit of every device and service connected to your phone lines is an essential first step in planning your migration. Cloudswitched's VoIP migration services include a comprehensive site survey that identifies every PSTN-dependent system in your organisation, ensuring nothing is missed during the transition.

What Is SIP Trunking? The Technology Behind Your Migration

SIP trunking is the technology that replaces ISDN and PSTN lines for business telephony. SIP stands for Session Initiation Protocol — a standardised communications protocol that manages the setup, maintenance, and termination of real-time voice, video, and messaging sessions over IP networks. A SIP trunk is essentially a virtual phone line that connects your telephone system to the public telephone network via your internet connection, rather than via a physical copper circuit.

Think of SIP trunking as the modern equivalent of your ISDN lines. Where an ISDN2e circuit gives you two fixed voice channels over a dedicated copper pair, and an ISDN30 circuit gives you up to 30 channels over a dedicated digital link, a SIP trunk delivers voice channels over your existing broadband or leased line connection. The key difference is flexibility: with SIP trunking, you can scale your channel count up or down in minutes, pay only for what you use, and take advantage of features that were simply impossible on the old ISDN infrastructure.

How SIP Trunking Works — A Technical Overview

When you make a call using SIP trunking, the following process happens in milliseconds:

1. Call Initiation (SIP INVITE): Your IP phone or PBX sends a SIP INVITE message to your SIP trunking provider's Session Border Controller (SBC). This message contains the caller's identity, the destination number, and the proposed media parameters (codecs, ports, etc.).

2. Call Routing: Your SIP trunking provider's platform authenticates the request, checks your account permissions and number ranges, and routes the call to its destination. If the call is to a PSTN number, the provider's gateway converts the SIP signalling to the appropriate format for the terminating network.

3. Media Establishment (RTP): Once the call is answered, a separate Real-time Transport Protocol stream is established between the endpoints. This carries the actual voice data — compressed using codecs like G.711 (uncompressed, 64 Kbps per channel) or G.729 (compressed, 8 Kbps per channel).

4. Call Termination (SIP BYE): When either party hangs up, a SIP BYE message ends the session and releases all associated resources.

The beauty of SIP trunking is that this entire process runs over your standard internet connection. There are no dedicated copper circuits to install, no engineer visits to add capacity, and no physical limits on how many concurrent calls you can handle — beyond your available bandwidth and your provider's licensing terms.

SIP Trunking vs ISDN — A Detailed Comparison

For businesses planning their ISDN to VoIP migration UK, understanding exactly how SIP trunking compares to your existing ISDN service is essential. The differences span cost, flexibility, features, reliability, and future-readiness:

The Business Case for SIP Trunking: Cost Analysis

One of the most compelling reasons to accelerate your ISDN to VoIP migration UK project is the significant cost savings that SIP trunking delivers. The financial case is clear at every scale, from a small business with a handful of lines to an enterprise with hundreds of concurrent call channels.

Let us examine the typical cost comparison for a mid-sized UK business with 30 concurrent call channels — a common configuration for companies with 50–150 employees:

Cost Category	ISDN30 (Legacy)	SIP Trunking	Annual Saving
Line rental (30 channels)	£450–750/month	£90–240/month	£2,520–6,120
UK landline calls	2–5p/min	0.5–1p/min or inclusive	£1,200–3,600
UK mobile calls	8–12p/min	2–4p/min or inclusive	£2,400–4,800
International calls	15–50p/min	1–10p/min	£1,000–5,000+
Maintenance & support	£100–200/month	Often included	£1,200–2,400
Scaling costs (adding channels)	£150–300 per channel + engineer	£0 — instant self-service	Variable
Estimated Total Annual Cost	£12,000–22,000	£4,800–9,600	£7,200–12,400

These figures represent a 40–60% reduction in telephony costs — a saving that goes straight to your bottom line, year after year. For many businesses, the cost of the entire ISDN to VoIP migration UK project is recouped within the first six to twelve months through reduced line rental and call charges alone.

Beyond direct cost savings, SIP trunking delivers additional financial benefits that are harder to quantify but equally significant. The elimination of costly engineer call-outs for capacity changes, the ability to consolidate multiple ISDN circuits at different sites into a single SIP trunking arrangement, and the reduced infrastructure footprint in your server room all contribute to a lower total cost of ownership. Many SIP trunking providers UK businesses work with also offer bundled call packages that include unlimited UK landline and mobile calls for a fixed monthly fee, making budgeting simpler and more predictable.

Planning Your ISDN to VoIP Migration: A Step-by-Step Framework

A successful ISDN to VoIP migration UK project requires careful planning, methodical execution, and expert guidance. Rushing the process risks service disruption, lost telephone numbers, and frustrated customers. The following framework, developed by Cloudswitched through hundreds of successful migrations for London and UK-wide businesses, provides a proven roadmap for a smooth transition.

Phase 1: Discovery and Audit

The foundation of every successful migration is a thorough understanding of your current telephony environment. This discovery phase should identify every PSTN-dependent system, every telephone number, every call flow, and every integration point in your organisation.

Inventory your current lines and services: Compile a complete list of every ISDN circuit, analogue line, and PSTN service in your organisation. For each, document the circuit reference number, the telephone numbers associated with it, the monthly cost, the contract end date, and what it is used for. Many businesses are surprised to discover they have more lines than they realised — particularly analogue lines serving legacy equipment in server rooms, lift shafts, and building plant areas.

Map your telephone numbers: Create a comprehensive register of every telephone number your business uses. This includes main business numbers, direct dial-in (DDI) ranges, fax numbers, premium-rate numbers, freephone numbers, and any numbers published in marketing materials, on business cards, or in online directories. Every one of these numbers will need to be ported to your new SIP trunking provider — and missing even one can cause significant business disruption.

Audit non-voice PSTN connections: As mentioned earlier, many business-critical systems rely on PSTN connectivity beyond voice calls. Conduct a physical walk-through of your premises to identify alarm panels, lift emergency phones, door entry systems, EPOS card terminals, franking machines, fax machines, CCTV remote monitoring, and building management systems. Each of these needs a specific migration path to an IP-based or cellular alternative.

Assess your network readiness: Your existing network infrastructure must be capable of supporting voice-quality IP traffic. This means adequate bandwidth (approximately 100 Kbps per concurrent call using G.711 codec), low latency (under 150ms one-way), minimal jitter (under 30ms), and negligible packet loss (under 1%). A professional network assessment, which Cloudswitched includes as part of our VoIP migration services, will identify any upgrades needed before your SIP trunking service goes live.

Phase 2: Solution Design and Provider Selection

With a clear picture of your current environment, the next step is to design your target VoIP solution and select the right SIP trunking providers UK businesses can trust. This phase involves several key decisions:

Hosted PBX vs On-Premise PBX with SIP Trunks: The first architectural decision is whether to use a fully hosted (cloud) PBX service or to deploy an on-premise PBX connected via SIP trunking. A hosted PBX moves everything to the cloud — the provider manages the PBX, and you simply connect endpoints. An on-premise PBX (whether a hardware appliance or a software platform like 3CX, FreePBX, or Asterisk) gives you more control but requires you to manage and maintain the system. For many UK businesses, particularly those with 10–200 users, a hosted PBX with integrated SIP trunking offers the best balance of features, cost, and simplicity.

Channel capacity planning: Unlike ISDN, where you purchase fixed channel blocks, SIP trunking allows flexible capacity. Analyse your call data reports from the past 12 months to determine your peak concurrent call count. Most businesses find that their peak concurrent calls equal approximately 30–40% of their total extension count. Add a 20% headroom buffer for growth and seasonal peaks.

Codec selection: The choice of voice codec affects both call quality and bandwidth consumption. G.711 (also known as PCMA/PCMU) provides toll-quality audio at 64 Kbps per call and is the safest choice for maximum compatibility. G.722 provides wideband HD audio at the same 64 Kbps but with noticeably richer sound. G.729 compresses audio to just 8 Kbps per call — ideal for bandwidth-constrained environments but with slightly reduced quality. Most SIP trunking providers UK businesses use will support all three codecs.

Resilience and redundancy: Design your VoIP solution with resilience in mind. Consider dual internet connections from different providers, automatic failover to mobile devices if the internet fails, and geographically diverse SIP trunk registrations. A well-designed SIP trunking deployment can actually be more reliable than ISDN, which has a single point of failure at the copper circuit level.

Choosing the Right SIP Trunking Providers UK Businesses Can Trust

The UK market for SIP trunking providers UK businesses use is mature and competitive, with options ranging from large carriers to specialist VoIP operators. When evaluating providers for your ISDN to VoIP migration UK project, consider the following criteria:

Network infrastructure and peering: The best SIP trunking providers UK companies rely on operate their own core voice network with multiple points of presence across the UK. Look for providers that peer directly with all major UK carriers (BT, Virgin Media, TalkTalk, Sky) to ensure the shortest, most reliable call paths. Providers that rely on third-party wholesale termination may offer lower prices but at the cost of call quality and reliability.

Number porting capability: Your provider must be able to port all of your existing telephone numbers — including geographic numbers (01/02), non-geographic numbers (03/08), and DDI ranges. Ask specifically about their porting success rate, average porting timescale (typically 5–10 working days for standard ports, longer for complex multi-range ports), and their process for handling porting issues.

SLA and uptime guarantees: Business-grade SIP trunking should come with a formal Service Level Agreement guaranteeing at least 99.9% uptime — and ideally 99.99%. Ensure the SLA includes meaningful financial penalties for breaches and covers the entire service chain, not just the provider's core platform.

Technical support quality: When your phone system stops working, you need fast, expert support. Evaluate the provider's support hours (ideally 24/7 for business-critical telephony), response time SLAs, escalation procedures, and whether support is UK-based. Ask for customer references from businesses of similar size and complexity to yours.

Security credentials: Your voice traffic carries sensitive business conversations. Look for providers that offer TLS encryption for SIP signalling and SRTP encryption for voice media, IP-based access control lists, fraud detection and prevention systems, and compliance certifications relevant to your industry (such as ISO 27001 or Cyber Essentials Plus).

Integration capabilities: Modern SIP trunking should integrate seamlessly with your existing business systems. Look for providers that offer CRM integration (Salesforce, HubSpot, Dynamics 365), Microsoft Teams Direct Routing compatibility, call recording with PCI-DSS compliance for payment environments, and comprehensive APIs for custom integration.

Number Porting: Keeping Your Business Numbers During Migration

For most businesses, their telephone numbers are valuable assets — published on websites, business cards, signage, marketing materials, and embedded in customer contact records. A critical element of any ISDN to VoIP migration UK project is ensuring that every one of your numbers transfers seamlessly to your new SIP trunking service. This process is known as number porting.

Number porting in the UK is governed by Ofcom regulations and facilitated by industry processes managed through the Number Portability Hub. When you port a number from your existing ISDN provider to your new SIP trunking provider, the underlying routing of calls to that number is updated in the national numbering database. The number itself does not change — your customers, suppliers, and contacts continue to dial the same number, and their calls are transparently routed to your new VoIP system.

The Number Porting Process

Step 1 — Provide your number inventory: Give your new SIP trunking provider a complete list of all numbers to be ported, including the full DDI ranges, the current provider name, and the circuit references (known as the "losing provider" details).

Step 2 — Authorisation: You will need to sign a Letter of Authority (LOA) authorising your new provider to request the port on your behalf. This is a regulatory requirement that prevents unauthorised number transfers.

Step 3 — Port request submission: Your new provider submits the port request to the losing provider through the industry's automated porting system. The losing provider has a defined timeframe to validate and accept the request.

Step 4 — Port scheduling: Once accepted, a porting date is agreed — typically 5–10 working days for standard ports. Complex ports involving large DDI ranges or multiple providers may take longer. Your new provider should keep you informed throughout.

Step 5 — Port execution: On the agreed date, the number routing is switched over. Calls to your numbers are now delivered to your new SIP trunking service. There is typically a brief interruption of 1–2 minutes during the switchover, which most providers schedule during off-peak hours.

Step 6 — Verification: Test all ported numbers immediately after the switch to confirm calls are being received correctly. Check inbound calls from landlines, mobiles, and international origins. Verify that DDI routing, hunt groups, and auto-attendant configurations are working as expected.

Common Number Porting Pitfalls to Avoid

Number porting is one of the most sensitive aspects of any ISDN to VoIP migration UK project. Common issues that can derail the process include:

• Incomplete number lists: Missing numbers from your porting request means those numbers remain with the old provider. Conduct a thorough audit before starting.
• Contract disputes: Your existing provider may charge early termination fees or delay the port if there are outstanding contractual issues. Review your current contract terms before initiating the migration.
• Mismatched records: The porting system requires exact matches between the number ranges you request and the records held by the losing provider. Discrepancies in account names, addresses, or circuit references can cause port rejections.
• Premature ISDN cancellation: Never cancel your ISDN circuits before the number port is complete. If you cancel the circuits, the numbers may be released back into the national pool and become permanently unrecoverable.
• Complex DDI ranges: Large DDI ranges that span multiple exchange areas or that have been partially ported in the past can require special handling. Experienced SIP trunking providers UK businesses use will have processes for managing these complexities.

Network Preparation: Getting Your Infrastructure Ready for SIP Trunking

The quality and reliability of your SIP trunking service is directly dependent on the quality of your underlying network infrastructure. Unlike ISDN, where voice quality is guaranteed by the dedicated circuit, SIP trunk calls share your internet connection with all other business traffic. Proper network preparation is therefore essential for a successful ISDN to VoIP migration UK.

Bandwidth Requirements

Each concurrent SIP trunk call requires a specific amount of bandwidth, depending on the codec used:

Codec	Audio Quality	Bandwidth per Call	10 Concurrent Calls	30 Concurrent Calls
G.711 (PCMA/PCMU)	Toll quality (narrowband)	87 Kbps	870 Kbps	2.6 Mbps
G.722	HD wideband	87 Kbps	870 Kbps	2.6 Mbps
G.729	Good (compressed)	32 Kbps	320 Kbps	960 Kbps
Opus	Excellent (adaptive)	16–128 Kbps	160 Kbps–1.28 Mbps	480 Kbps–3.84 Mbps

As a rule of thumb, ensure your internet connection has at least 50% more upload bandwidth than your peak voice traffic requirement. For a business with 30 concurrent calls using G.711, this means at least 4 Mbps of dedicated upload bandwidth for voice alone. Remember that upload bandwidth is typically the bottleneck, especially on asymmetric connections like FTTC (fibre-to-the-cabinet).

Quality of Service (QoS) Configuration

Quality of Service settings on your network ensure that voice packets are prioritised over other traffic types. Without QoS, a large file download or a cloud backup running in the background could cause voice quality to degrade noticeably — resulting in choppy audio, delays, or dropped calls.

Key QoS configurations for SIP trunking include:

• DSCP marking: Tag SIP signalling packets with DSCP CS3 (or AF31) and RTP voice packets with DSCP EF (Expedited Forwarding). This ensures your router and switches give voice traffic absolute priority.
• Traffic shaping: Configure your router to reserve a minimum bandwidth allocation for voice traffic, even under heavy network load.
• VLAN separation: Place voice endpoints on a dedicated VLAN, separate from data traffic. This provides both performance isolation and security benefits.
• Jitter buffer configuration: Ensure your IP phones and PBX have appropriately sized jitter buffers (typically 40–80ms) to smooth out packet arrival timing variations.

A score of 90% or above on a professional VoIP readiness assessment indicates your network is well-prepared for SIP trunking deployment. Cloudswitched's VoIP migration services include a comprehensive network assessment using industry-standard tools to measure latency, jitter, packet loss, and mean opinion score (MOS) across your network.

Firewall and Security Configuration

Your firewall must be configured to allow SIP trunking traffic whilst maintaining security. This is a common area where migrations encounter issues, as many firewalls are not natively optimised for real-time voice traffic.

SIP ALG (Application Layer Gateway): Most business firewalls include a feature called SIP ALG, which attempts to inspect and modify SIP packets as they pass through. In practice, SIP ALG causes far more problems than it solves — mangling SIP headers, breaking NAT traversal, and causing one-way audio issues. The strong recommendation from virtually every SIP trunking provider is to disable SIP ALG on your firewall.

Port configuration: SIP signalling typically uses UDP port 5060 (or TCP/TLS port 5061 for encrypted signalling). RTP media uses a dynamic range of UDP ports, typically 10000–20000. Your firewall must allow outbound traffic on these ports to your SIP provider's IP addresses, and must allow the corresponding return traffic.

Session Border Controller (SBC): For larger deployments or environments with strict security requirements, consider deploying a Session Border Controller at your network edge. An SBC provides deep SIP protocol inspection, topology hiding, DoS protection, and encryption termination — acting as a dedicated firewall for your voice traffic.

Testing and Cutover: Ensuring a Smooth Transition

Thorough testing before the final cutover is the single most important factor in a successful ISDN to VoIP migration UK. A well-structured test plan verifies every aspect of your new SIP trunking service before it handles live customer calls.

Pre-Cutover Testing Checklist

Before scheduling your final number port and cutover, complete the following tests on your new SIP trunking platform:

Basic connectivity: Verify that your PBX successfully registers with your SIP trunking provider. Check that registration remains stable over time (no repeated re-registrations that could indicate network issues).

Outbound call testing: Make test calls to UK landlines, UK mobiles, international numbers, premium-rate numbers, freephone numbers, and emergency services (using the test number 112, which connects to the operator). Verify that CLI (Caller Line Identification) is presented correctly on each call — your business number should display, not the underlying SIP trunk number.

Inbound call testing: If your provider has issued temporary numbers, test inbound calls from multiple originating networks. Verify that calls ring the correct extensions, hunt groups, and auto-attendant menus. Test call forwarding, voicemail, and out-of-hours routing.

Call quality testing: Conduct extended calls (15+ minutes) during peak business hours to verify sustained quality under real-world network conditions. Listen for any audio degradation, echo, delay, or choppy audio. Use objective measurement tools that report Mean Opinion Score (MOS) — a score above 4.0 (on a 5-point scale) indicates excellent quality.

Failover testing: If your solution includes disaster recovery features (such as automatic failover to mobile numbers or a secondary site), test these by deliberately simulating failures. Disconnect your primary internet connection and verify that calls are redirected according to your failover rules.

Load testing: If possible, simulate your peak concurrent call count to verify that your network and SIP trunking service can handle the load without quality degradation. This is particularly important for businesses with high call volumes or seasonal peaks.

Integration testing: Test all integrations with business systems — CRM screen pops, call recording, Microsoft Teams Direct Routing, contact centre routing, and any custom applications that interact with your phone system.

Cutover Strategy

The cutover — the moment when your live telephone numbers switch from ISDN to SIP trunking — should be carefully planned and executed. Two main strategies are available:

Big-bang cutover: All numbers are ported simultaneously on a single date. This approach is simpler to manage and avoids the complexity of running parallel systems, but carries higher risk if issues arise. Best suited for smaller organisations or those with a single site.

Phased cutover: Numbers are ported in batches over several days or weeks. This approach allows you to verify each batch before proceeding to the next, reducing risk. It does require temporary parallel operation of both ISDN and SIP systems. Best suited for larger organisations, multi-site deployments, or businesses where telephone service is business-critical.

Regardless of which approach you choose, schedule the cutover during a low-traffic period (typically early morning or over a weekend), have your SIP trunking provider's support team on standby, maintain your ISDN circuits as a fallback until the SIP service is confirmed stable (usually 2–4 weeks post-cutover), and ensure your IT team and any affected staff are briefed on the change and know how to report issues.

Business Continuity During Migration

One of the primary concerns for any business undertaking an ISDN to VoIP migration UK is maintaining uninterrupted telephone service throughout the transition. Your customers, suppliers, and partners should experience no disruption — and ideally should notice an improvement in call quality once the migration is complete.

Risk Mitigation Strategies

Professional VoIP migration services incorporate multiple risk mitigation strategies to protect your business during the transition:

Parallel running: Maintain your existing ISDN service running alongside your new SIP trunking service during the testing and initial cutover period. This provides an immediate fallback if any issues arise with the new system. The additional cost of running both services for 2–4 weeks is minimal compared to the risk of a total telephony outage.

Temporary number diversion: Set up call diverts on your existing ISDN lines to redirect calls to temporary SIP numbers during the transition period. This allows you to test the new system with live calls whilst maintaining the ability to instantly revert to ISDN if needed.

Mobile failover: Configure your SIP trunking service with automatic failover to designated mobile numbers. If your internet connection fails during the migration, calls are seamlessly redirected to key staff members' mobile phones. This provides a safety net that ISDN never offered.

Communication plan: Notify key customers, suppliers, and partners about the migration timeline. Whilst the process should be transparent to them, having a communication plan demonstrates professionalism and provides a point of contact if anyone experiences issues reaching you during the cutover.

Studies consistently show that businesses using professional VoIP migration services achieve a 90%+ first-time success rate for number ports and cutovers, compared to significantly lower success rates for self-managed migrations. The expertise of a specialist managed service provider like Cloudswitched — who handles ISDN to VoIP migration UK projects as a core service — substantially reduces the risk of service disruption.

What Happens If You Don't Migrate?

Some businesses may be tempted to delay their ISDN to VoIP migration UK project, hoping for an extension to the PSTN switch-off deadline or simply struggling to prioritise the work. This is a dangerous strategy. The consequences of failing to migrate before January 2027 are severe and potentially catastrophic for your business.

The Immediate Consequences

Total loss of telephone service: When your local exchange is migrated to all-IP operation — or when the final PSTN switch-off occurs in January 2027 — your ISDN and analogue lines will simply stop working. No incoming calls, no outgoing calls, no fax transmissions, no alarm monitoring. Your business becomes unreachable by phone overnight.

Loss of telephone numbers: If your lines are disconnected without the numbers being ported, those numbers may be released back into the national numbering pool after a quarantine period. Recovering a number that has been released is difficult, time-consuming, and not guaranteed. You could permanently lose telephone numbers that your customers have been using to reach you for years or decades.

Non-voice system failures: Alarm systems connected via PSTN will stop reporting to monitoring centres, potentially invalidating your insurance. Lift emergency phones will stop working, breaching building safety regulations. EPOS terminals using PSTN dial-up connections will stop processing card payments. Fire alarm panels with PSTN diallers will lose their communication path to the fire brigade.

Regulatory and compliance breaches: Depending on your industry, losing telephone service may constitute a breach of regulatory requirements. Financial services firms, healthcare providers, and organisations with public safety responsibilities may face regulatory action if they fail to maintain communications capability.

Competitive disadvantage: While your phone lines are down, your customers are calling your competitors. Every missed call is a missed opportunity — and in many industries, a customer who cannot reach you on the first attempt will not try again. They will simply move on to the next provider they find.

The Hidden Cost of Waiting

Beyond the catastrophic scenario of missing the deadline entirely, delaying your migration has real costs right now. Every month you continue paying for ISDN is a month where you are overpaying for telephony — the cost differential between ISDN and SIP trunking means you are effectively burning money. Furthermore, as the January 2027 deadline approaches, the demand for VoIP migration services will surge, and the availability of experienced migration engineers will tighten. Businesses that wait until the last six months will face longer lead times, higher costs, and less flexibility in scheduling their cutover.

The message is clear: the PSTN switch off 2027 UK deadline is not going to be extended. Openreach has been unequivocal on this point. The time to begin your ISDN to VoIP migration UK project is now — not next quarter, not next year, and certainly not when the deadline is imminent and options are limited.

SIP Trunking Features That Transform Business Communications

Migrating from ISDN to SIP trunking is not just about avoiding the PSTN switch-off — it is an opportunity to dramatically improve your business communications. Modern SIP trunking services offer features and capabilities that were simply impossible on the legacy ISDN infrastructure.

Advanced Call Routing and Management

Geographic flexibility: With SIP trunking, your telephone numbers are no longer tied to a physical location. A London 020 number can ring on a desk in Manchester, a mobile in Edinburgh, or a laptop in a co-working space in Brighton. This enables truly flexible working arrangements without the complexity of call forwarding or the cost of maintaining multiple ISDN circuits at each location.

Time-based routing: Configure different call handling rules for business hours, lunchtimes, evenings, weekends, and bank holidays. Calls outside business hours can be automatically routed to voicemail, an out-of-hours answering service, or an on-call mobile — all without manual intervention.

Skills-based routing: For businesses with contact centre operations, SIP trunking combined with a modern PBX enables sophisticated call routing based on agent skills, language proficiency, customer tier, or call type. This ensures that every caller reaches the most appropriate person first time.

Automatic failover: If your primary internet connection fails, calls can be automatically redirected to backup numbers, mobile phones, or a secondary site. This level of resilience simply wasn't available with ISDN, where a cable cut or exchange fault meant silence on the line.

Unified Communications Integration

Modern SIP trunking is the foundation for unified communications (UC) — the convergence of voice, video, messaging, and presence into a single integrated platform. Key UC capabilities enabled by SIP trunking include:

Microsoft Teams Direct Routing: Connect your SIP trunking service directly to Microsoft Teams, enabling Teams to become your primary business phone system. Staff can make and receive calls on their business numbers directly from the Teams desktop client, mobile app, or Teams-enabled desk phones — with full PSTN connectivity for calls to external numbers.

Video conferencing integration: Many SIP-based PBX platforms include built-in video conferencing capabilities, allowing you to escalate a voice call to a video meeting with a single click. This reduces reliance on separate video conferencing subscriptions and simplifies your communications stack.

Instant messaging and presence: Know at a glance whether colleagues are available, busy, on a call, or away — and send quick messages without picking up the phone. Presence information shared across your organisation improves response times and reduces unnecessary voicemail.

CRM integration: When a customer calls, their record automatically appears on your screen before you answer. When you make an outbound call, it is logged against the customer record in your CRM. Click-to-dial from your CRM saves time and reduces misdials. These integrations are straightforward with SIP trunking and a modern PBX, but were practically impossible to implement reliably over ISDN.

Analytics and Reporting

SIP trunking platforms generate detailed call data records (CDRs) that provide rich insights into your communications patterns. Modern analytics dashboards can show you call volumes by time of day, average call duration by department, missed call rates, peak hour analysis, and call quality metrics. This data enables evidence-based decisions about staffing, training, and customer service improvements.

Industry-Specific Migration Considerations

Different industries face unique challenges and opportunities when planning their ISDN to VoIP migration UK. Here are some sector-specific considerations that Cloudswitched addresses as part of our VoIP migration services:

Professional Services (Legal, Accounting, Consulting)

Professional services firms handle highly confidential client communications. Key priorities include encrypted SIP trunking (TLS/SRTP), compliant call recording with tamper-evident storage, integration with practice management systems, and the ability to present different CLI numbers for different departments or partner lines. Many legal firms also require the ability to bill telephone time accurately to client matters, which SIP-based PBX platforms support through detailed CDR reporting and billing integration.

Healthcare

Healthcare organisations must ensure their SIP trunking service meets NHS Digital guidelines and maintains clinical safety standards. Key requirements include 999/112 emergency calling with accurate location information, integration with appointment booking and clinical systems, high availability with sub-second failover (for on-call doctor routing), and compliance with the Data Protection Act 2018 and UK GDPR for recorded calls containing patient information.

Retail and Hospitality

Multi-site retail and hospitality businesses benefit enormously from centralised SIP trunking. Instead of maintaining separate ISDN circuits at each branch, a single SIP trunking arrangement can serve all locations. Calls can be intelligently routed between sites, central reservation lines can overflow to available staff at any location, and head office has visibility of call activity across the entire estate.

Financial Services

FCA-regulated financial services firms have specific requirements for call recording, retention, and retrieval. SIP trunking must be combined with PCI-DSS compliant call recording solutions that can pause recording during card payment capture, retain recordings for the required period (typically 5–7 years), and provide rapid search and retrieval for regulatory inquiries. MiFID II compliance requires recording of all communications that relate to transactions, making comprehensive call recording a regulatory necessity.

Education

Schools, colleges, and universities face the PSTN switch-off alongside tight budgets. SIP trunking offers significant cost savings over ISDN, but the migration must account for safeguarding requirements (including robust 999 calling from every location on campus), integration with visitor management and lockdown systems, bell and PA system triggers, and the academic calendar (ideally scheduling the cutover during school holidays).

Choosing Between Hosted VoIP and SIP Trunking with On-Premise PBX

When planning your ISDN to VoIP migration UK, one of the fundamental decisions is whether to adopt a fully hosted VoIP service (where the PBX is in the cloud) or to deploy SIP trunks connected to an on-premise PBX system. Both approaches use SIP trunking as the underlying connectivity method, but they differ significantly in terms of control, cost structure, and management responsibility.

For the majority of UK SMBs undertaking their ISDN to VoIP migration UK, a hosted VoIP solution offers the best combination of simplicity, cost-effectiveness, and feature richness. Larger enterprises, contact centres, and organisations with highly specialised telephony requirements may benefit from the additional control of an on-premise PBX connected via SIP trunking. Cloudswitched advises clients on the optimal approach for their specific situation as part of our VoIP migration services.

Security Considerations for SIP Trunking

As voice communications move from the relatively closed PSTN/ISDN environment to IP networks, security becomes a critical consideration. SIP trunking introduces new attack surfaces that must be addressed through a combination of technical controls, provider selection, and ongoing monitoring.

Common SIP Security Threats

Toll fraud: The most financially damaging threat. Attackers compromise your PBX or SIP trunking credentials and use them to make expensive international or premium-rate calls, often generating thousands of pounds in charges within hours. Toll fraud is a global problem costing the telecoms industry billions annually.

Denial of Service (DoS): Attackers flood your SIP infrastructure with bogus requests, overwhelming your PBX or internet connection and preventing legitimate calls from being made or received.

Eavesdropping: Without encryption, SIP signalling and RTP media can be intercepted and recorded by anyone with access to the network path between your PBX and your SIP trunking provider.

Registration hijacking: Attackers attempt to register their own device with your SIP trunking provider using stolen credentials, effectively hijacking your telephone numbers.

Vishing (voice phishing): Attackers use compromised VoIP systems to make calls that appear to originate from your business number, conducting social engineering attacks against your customers or suppliers.

Security Best Practices

Protecting your SIP trunking service requires a layered security approach:

Encryption: Use TLS (Transport Layer Security) for SIP signalling and SRTP (Secure Real-time Transport Protocol) for voice media. This prevents eavesdropping and tampering with call data in transit. All reputable SIP trunking providers UK businesses use should support both TLS and SRTP.

Strong authentication: Use complex, unique passwords for SIP trunk registration. Enable IP-based access control lists (ACLs) to restrict which IP addresses can register with your provider. Consider mutual TLS authentication for the highest security.

Fraud prevention: Configure call spending limits with your SIP trunking provider. Block calls to high-cost international destinations that your business never calls. Set up real-time alerting for unusual call patterns (such as calls outside business hours or sudden spikes in international traffic). Enable your provider's fraud detection systems.

Network security: Deploy a Session Border Controller (SBC) or SIP-aware firewall at your network edge. Keep your PBX software and firmware up to date with security patches. Disable unused SIP features and protocols. Monitor your PBX logs for suspicious activity.

Regular security audits: Conduct periodic security reviews of your SIP trunking configuration, PBX security settings, and network firewall rules. Penetration testing by a qualified security assessor can identify vulnerabilities before attackers exploit them.

Research from UK cybersecurity firms indicates that up to 80% of successful attacks against SIP trunking deployments could have been prevented by implementing basic security controls — encryption, strong authentication, and fraud prevention limits. These are straightforward measures that any competent SIP trunking provider and IT support partner should implement as standard.

Multi-Site and Remote Working: SIP Trunking for the Modern Workforce

The shift to hybrid and remote working has fundamentally changed how UK businesses use their telephone systems. SIP trunking is inherently well-suited to supporting distributed teams, as it decouples telephone numbers from physical locations and enables voice calls to follow users wherever they are.

Centralised SIP Trunking for Multi-Site Businesses

Businesses with multiple offices traditionally needed separate ISDN circuits at each location, each with its own set of telephone numbers and its own phone system (or expensive ISDN tie lines connecting PBX systems). This was costly, complex to manage, and created inconsistent user experiences across sites.

With centralised SIP trunking, a single SIP trunk arrangement can serve all of your sites through a cloud-hosted PBX. Each location connects to the central PBX via its local internet connection. Internal calls between sites are carried over the internet at zero cost. Telephone numbers for any location can ring at any other location. Staff can hot-desk between offices and retain their extension and direct dial number. And the entire estate can be managed from a single administration portal.

The cost savings from consolidating multiple ISDN circuits into a single centralised SIP trunking arrangement can be substantial. A business with five offices, each with an ISDN30 circuit, could save £30,000–£50,000 annually by moving to centralised SIP trunking — on top of the improved features and simplified management.

Supporting Remote and Hybrid Workers

Remote workers using SIP trunking can access the full features of the business phone system from any location with an internet connection. Softphone applications on laptops and mobile apps on smartphones connect to the central PBX via SIP, enabling remote staff to make and receive calls on their business extension, present the business CLI to external callers, access the company directory and presence information, transfer calls to colleagues, and use voicemail, call recording, and all other PBX features.

From the caller's perspective, there is no difference between calling a desk-based employee and a remote worker. The business number rings, the call is answered, and the quality is indistinguishable from an in-office call. This transparency is essential for maintaining professional standards and customer confidence in a hybrid working environment.

Frequently Asked Questions About the PSTN Switch-Off and SIP Trunking

Based on the hundreds of ISDN to VoIP migration UK projects Cloudswitched has delivered, here are the questions we hear most frequently from businesses planning their migration:

Q: Can the PSTN switch-off deadline be extended?
A: No. Openreach has been unequivocal that the PSTN switch off 2027 UK deadline is fixed. The programme is well advanced, with many exchanges already migrated. There is no precedent or mechanism for an industry-wide extension, and the economic case for maintaining ageing PSTN infrastructure does not exist. Plan on the January 2027 deadline being absolute.

Q: Will my phone numbers change?
A: No. All of your existing telephone numbers — geographic, non-geographic, freephone, and DDI ranges — can be ported to your new SIP trunking service. Your customers continue to dial the same numbers. The porting process is regulated by Ofcom and is a standard, well-established procedure.

Q: What broadband speed do I need for SIP trunking?
A: Each concurrent call requires approximately 100 Kbps of bandwidth using the G.711 codec. For a business with 10 concurrent calls, you need at least 1.5 Mbps of dedicated upload bandwidth for voice. We recommend having at least 50% headroom above your peak requirement. A fibre-to-the-premises (FTTP) or leased line connection is ideal for businesses with high call volumes.

Q: What about fax machines?
A: Traditional fax machines connected to analogue PSTN lines will stop working after the switch-off. The most common replacement is a Fax-over-IP (FoIP) gateway or, more practically, a cloud-based fax service that sends and receives faxes as email attachments. If fax is critical for your business, discuss the options with your VoIP migration services provider.

Q: How long does the migration take?
A: A typical ISDN to VoIP migration UK project takes 8–14 weeks from initial discovery to completed cutover, depending on complexity. Simple single-site migrations can be faster. Complex multi-site deployments with large number ranges and extensive integration requirements may take longer. Starting early gives you more flexibility and less pressure.

Q: Is VoIP call quality as good as ISDN?
A: On a properly configured network with adequate bandwidth, SIP trunking call quality is equal to or better than ISDN. HD (wideband) codecs available on SIP deliver noticeably richer audio than the narrowband G.711 codec used on ISDN. Network quality issues (jitter, packet loss) can degrade call quality, which is why professional network assessment and QoS configuration are essential parts of any migration.

Q: What happens to my alarm system?
A: Intruder alarms, fire alarms, and other security systems that use PSTN lines for monitoring will need to be migrated to alternative communication paths. Options include cellular (4G/5G) communicators, IP-based alarm signalling over your broadband connection, or dual-path solutions that use both. Speak to your alarm monitoring company well in advance of your migration — they will need to update their receiving equipment to accept signals from your new communication path.

Q: Can I use SIP trunking with Microsoft Teams?
A: Yes. Microsoft Teams Direct Routing allows you to connect SIP trunking directly to the Teams platform, enabling Teams users to make and receive PSTN calls using your business telephone numbers. This is one of the most popular deployment models for UK businesses that have adopted Microsoft 365. Not all SIP trunking providers UK businesses use support Direct Routing, so verify this capability during your provider selection process.

The Cloudswitched Approach to VoIP Migration Services

As a London-based IT managed service provider with extensive experience in business telephony, Cloudswitched delivers comprehensive VoIP migration services that take the complexity and risk out of your ISDN to VoIP migration UK project. Our approach is designed to ensure a smooth, disruption-free transition that not only meets the PSTN switch off 2027 UK deadline but positions your business to take full advantage of modern communications technology.

Our Migration Methodology

1. Complimentary Assessment: We begin with a no-obligation assessment of your current telephony environment, business requirements, and migration priorities. This includes a review of your existing ISDN/PSTN services, an evaluation of your network readiness, and an initial recommendation on the optimal SIP trunking solution for your organisation.

2. Detailed Design: Our engineers design your target VoIP solution, specifying the PBX platform, SIP trunking configuration, endpoint devices, network upgrades, call flow design, and integration requirements. We present this design to you for review and approval before any implementation begins.

3. Infrastructure Preparation: We configure your network infrastructure for optimal voice quality, including QoS settings, VLAN configuration, firewall rules, and bandwidth upgrades where necessary. We work with your chosen SIP trunking providers UK to provision the trunk and configure all necessary security controls.

4. Number Porting Management: We manage the entire number porting process on your behalf, liaising with your current provider and your new SIP trunking provider to ensure every number is ported correctly and on schedule. Our porting success rate exceeds 99%, with zero number losses across our migration history.

5. Configuration and Testing: We configure your PBX, extensions, call flows, voicemail, auto-attendant, hunt groups, and all other telephony features according to the agreed design. Comprehensive testing — including call quality, failover, and integration testing — is completed before any live traffic is carried.

6. Managed Cutover: We manage the cutover process, typically scheduling it during off-peak hours to minimise any impact. Our engineers monitor the system in real-time during and after the cutover, ensuring that all calls are routing correctly and that quality meets our standards.

7. Ongoing Support: After migration, Cloudswitched provides ongoing managed support for your VoIP infrastructure. This includes monitoring, fault resolution, adds/moves/changes, capacity management, and regular service reviews to ensure your SIP trunking solution continues to meet your evolving business needs.

Your ISDN to VoIP Migration Checklist

To help you plan and track your ISDN to VoIP migration UK project, here is a comprehensive checklist that covers every phase of the transition. Use this alongside your chosen VoIP migration services provider to ensure nothing is missed.

Understanding SIP Trunking Pricing Models in the UK

When evaluating SIP trunking providers UK businesses work with, understanding the different pricing models is essential for making an accurate cost comparison and choosing the right provider for your needs.

Per-Channel Pricing

The most common pricing model for SIP trunking in the UK is per-channel pricing. You pay a fixed monthly fee for each concurrent call channel. Typical UK prices range from £3 to £8 per channel per month, depending on the provider, the volume of channels purchased, and the features included. Call charges are typically billed separately, either per minute or as part of an inclusive call bundle.

Per-User Pricing

Some hosted VoIP providers combine SIP trunking with their PBX platform and charge a per-user monthly fee. This simplifies budgeting but may be more expensive per channel for businesses with a low ratio of concurrent calls to total users. Per-user pricing typically ranges from £8 to £25 per user per month, often including a generous number of inclusive minutes.

Metered vs Unmetered Calls

Metered plans charge for each call on a per-minute basis, whilst unmetered plans include unlimited calls to UK landlines and/or mobiles for a fixed monthly fee. For businesses with high call volumes, an unmetered plan offers cost predictability and usually works out cheaper. For businesses with low call volumes or primarily inbound traffic, a metered plan may be more economical.

Pricing Model	Monthly Cost Range	Best For	Watch Out For
Per-channel (metered calls)	£3–6/channel + call charges	Low-volume callers, inbound-heavy businesses	Call charges can spike unexpectedly
Per-channel (inclusive minutes)	£5–10/channel	Medium-volume callers, predictable usage	Overage charges if you exceed inclusive minutes
Per-user (hosted VoIP)	£8–25/user	SMBs wanting all-inclusive simplicity	Can be expensive for large user counts with low call volumes
Unlimited (all-you-can-call)	£8–15/channel	High-volume outbound callers, contact centres	Fair usage policies may apply; check the small print

Emergency Calling and Regulatory Compliance

A critical aspect of any SIP trunking deployment that is sometimes overlooked during the ISDN to VoIP migration UK planning process is emergency calling. Under Ofcom regulations, any provider of electronic communications services that enable calls to the public telephone network must ensure that their users can call 999 and 112 emergency services.

With ISDN, emergency call location information was automatically determined from the physical address associated with the copper circuit. With SIP trunking, where a number can be used from any location, the responsibility for providing accurate location information shifts to the business and its VoIP provider.

Key requirements for emergency calling on SIP trunking:

• Registered address: Every SIP trunking service must have a registered address associated with it for emergency call routing purposes. This address is provided to the emergency services when a 999/112 call is made.
• Multi-site accuracy: If you have multiple offices, each location should have its own registered address configured in the SIP platform to ensure the correct location is passed to emergency services regardless of which site the call originates from.
• Remote worker considerations: For remote workers using softphones, the registered address may not reflect their actual location. Some providers offer solutions that allow users to update their emergency location dynamically. Businesses should have clear policies about emergency calling procedures for remote staff.
• Fallback mechanisms: Consider what happens if your internet connection fails and a 999 call needs to be made. Mobile phones provide an obvious backup, but this should be explicitly documented in your business continuity plan.

Future-Proofing Your Communications: Beyond the PSTN Switch-Off

While the immediate driver for your ISDN to VoIP migration UK project is the PSTN switch off 2027 UK deadline, the transition to SIP trunking opens the door to a range of emerging communications technologies that will shape business telephony over the coming decade.

AI-powered voice analytics: Modern VoIP platforms can integrate with artificial intelligence and machine learning systems to provide real-time call transcription, sentiment analysis, keyword spotting, and automated quality scoring. These capabilities transform telephone calls from ephemeral conversations into structured, searchable business data.

WebRTC integration: WebRTC (Web Real-Time Communication) enables voice and video calls directly from web browsers without plugins. Combined with SIP trunking, this enables click-to-call functionality on your website, browser-based softphones for staff, and seamless voice integration into web applications and customer portals.

5G and edge computing: As 5G networks expand across the UK, the combination of ultra-low-latency mobile connectivity and edge computing will enable new voice application architectures. Mobile SIP trunking over 5G could provide fixed-line quality voice with the flexibility of mobile connectivity — particularly valuable for field-based workforces.

CPaaS (Communications Platform as a Service): API-driven communications platforms allow businesses to embed voice, video, messaging, and other communication capabilities directly into their applications and workflows. SIP trunking provides the PSTN connectivity layer for CPaaS solutions, enabling programmatic voice interactions such as automated appointment reminders, two-factor authentication calls, and interactive voice response systems.

An estimated 75% of UK businesses have already completed or are well advanced in their migration from ISDN to VoIP and SIP trunking. The remaining 25% — approximately 2 million businesses — face increasing urgency as the January 2027 deadline approaches. Early movers have already been benefiting from lower costs, better features, and improved resilience for years. Businesses that migrate now will still capture significant benefits and have ample time for thorough testing and a smooth transition. Those that wait much longer risk a rushed, costly, and potentially disruptive migration.

Common Migration Mistakes and How to Avoid Them

Drawing on Cloudswitched's extensive experience delivering VoIP migration services across London and the UK, here are the most common mistakes we see businesses make during their ISDN to VoIP migration UK — and how to avoid them:

Mistake 1: Treating it as a like-for-like replacement. Simply replacing ISDN channels with SIP channels without redesigning your call flows and taking advantage of new capabilities is a wasted opportunity. Use the migration as a chance to optimise your communications — improve call routing, add HD voice, implement unified communications, and design for resilience.

Mistake 2: Skipping the network assessment. Deploying SIP trunking on a network that hasn't been tested for voice quality is gambling with your customer experience. A £200 network assessment can prevent thousands of pounds of troubleshooting and reputation damage after go-live.

Mistake 3: Choosing the cheapest SIP trunking provider. The cheapest SIP trunking providers UK market offers may look attractive on paper, but if they don't offer encryption, have poor support, lack fraud prevention, or route calls through low-quality carriers, you'll pay the price in other ways. Evaluate total cost of ownership, not just per-channel pricing.

Mistake 4: Forgetting non-voice PSTN connections. Alarm systems, lifts, EPOS terminals, and building management systems are easily overlooked. A single missed analogue line can mean a failed security system or a non-operational lift emergency phone — with serious safety and compliance implications.

Mistake 5: Not planning for the cutover. A cutover without a detailed plan, rollback procedure, and support escalation path is an accident waiting to happen. Every cutover should have a written plan, assigned responsibilities, and a clear decision point for reverting to the old system if issues arise.

Mistake 6: Cancelling ISDN before migration is stable. Keep your ISDN circuits active for at least 2–4 weeks after cutover as a safety net. The monthly rental cost for this overlap period is negligible compared to the risk of being without telephone service if an issue arises with the new system.

Mistake 7: Going it alone without expert help. SIP trunking migration involves telecoms engineering, network configuration, number porting logistics, security hardening, and project management. Unless your IT team has specific experience in VoIP migration, engaging professional VoIP migration services will save time, reduce risk, and deliver a better outcome.

Glossary of Key Terms

To help you navigate the technical terminology associated with SIP trunking and VoIP migration, here is a glossary of the most important terms:

Term	Definition
PSTN	Public Switched Telephone Network — the traditional circuit-switched telephone network being retired in January 2027
ISDN	Integrated Services Digital Network — a digital telephony standard that runs on PSTN infrastructure, available as ISDN2e (2 channels) or ISDN30 (up to 30 channels)
SIP	Session Initiation Protocol — the standard protocol for establishing, maintaining, and terminating voice, video, and messaging sessions over IP networks
SIP Trunk	A virtual connection between your PBX and the public telephone network, delivered over your internet connection using SIP
VoIP	Voice over Internet Protocol — the technology of transmitting voice calls as data packets over IP networks
PBX	Private Branch Exchange — the telephone switching system that manages internal extensions, call routing, and features within an organisation
RTP	Real-time Transport Protocol — the protocol that carries actual voice data during a SIP call
Codec	Coder-Decoder — software that compresses and decompresses voice audio for transmission (e.g., G.711, G.722, G.729, Opus)
QoS	Quality of Service — network settings that prioritise voice traffic to ensure call quality
SBC	Session Border Controller — a dedicated network device that secures and manages SIP traffic at the network edge
DDI	Direct Dial-In — telephone numbers that route directly to individual extensions or departments within a PBX
CLI	Caller Line Identification — the telephone number displayed to the person you are calling
Number Porting	The process of transferring telephone numbers from one provider to another whilst keeping the same numbers
TLS/SRTP	Transport Layer Security / Secure Real-time Transport Protocol — encryption protocols that protect SIP signalling and voice media
SIP ALG	SIP Application Layer Gateway — a firewall feature that modifies SIP packets (usually recommended to disable)

Conclusion: Act Now to Secure Your Business Communications Future

The PSTN switch off 2027 UK is not a distant possibility — it is a confirmed, imminent change that will affect every business in the United Kingdom. The Integrated Services Digital Network and the Public Switched Telephone Network that have served British businesses for decades are reaching their final months of operation. Every ISDN circuit, every analogue line, and every PSTN-dependent system in your organisation will need to be migrated to an IP-based alternative before January 2027.

The good news is that SIP trunking and modern VoIP technology offer a compelling upgrade — not just a like-for-like replacement. Lower costs, greater flexibility, HD voice quality, unified communications, geographic freedom, built-in resilience, and a wealth of advanced features make the migration from ISDN to VoIP a genuinely positive step for your business. The transition is an opportunity to modernise your communications infrastructure, improve your customer experience, and position your organisation for the all-IP future.

But time is running short. Exchange-level migrations are accelerating across the UK. The availability of experienced migration engineers and project slots is tightening as more businesses compete for the same resources. Businesses that begin their ISDN to VoIP migration UK now will enjoy a well-planned, thoroughly tested, and smoothly executed transition. Those that delay risk a rushed migration with elevated costs and higher risk of disruption.

Cloudswitched has been guiding London and UK businesses through the PSTN switch-off since the programme was first announced. Our comprehensive VoIP migration services cover every aspect of the journey — from initial audit and solution design through provider selection, number porting, network preparation, testing, cutover, and ongoing managed support. With over 500 successful migrations behind us and a 99.8% number porting success rate, we have the experience, expertise, and track record to deliver a migration you can rely on.

Don't wait until the deadline forces your hand. Contact Cloudswitched today to begin your ISDN to VoIP migration UK project and ensure your business is ready for the all-IP future. Your customers — and your bottom line — will thank you.