SIP Trunks: ISDN30 Replacement for Multi-Site UK PBXs

Executive Summary: The 2027 Connectivity Mandate

The telecommunications landscape of the United Kingdom is undergoing its most radical transformation since the digitalization of the exchange network in the 1980s. The Openreach PSTN (Public Switched Telephone Network) and ISDN (Integrated Services Digital Network) switch-off, now officially scheduled for 31 January 2027, represents a critical operational deadline for every British business.

For Small and Medium Enterprises (SMEs), the path is often straightforward: a move to cloud-based “Hosted VoIP.” However, for mid-market and enterprise organizations operating multi-site estates with established IP-PBX systems (such as Avaya IP Office, Mitel MiVoice, Cisco Unified CM, or 3CX), the “rip and replace” model of hosted VoIP is often operationally disruptive and financially illogical.

These businesses face a clear choice: migrate to a modern cloud phone system or implement SIP Trunking. While the market often pushes hosted solutions, SIP Trunking remains the “Gold Standard” technical replacement for ISDN30. It offers the stability, control, and scalability required by complex organizations without necessitating a complete overhaul of internal hardware.

This guide serves as the definitive technical blueprint for IT Directors, Network Architects, and Operations Managers. It details how to navigate the transition from copper to cloud-native telephony without losing functionality, compromising voice quality, or disrupting site-to-site continuity.

Understand the full timeline and drivers behind this change in our article on the technical history of the PSTN switch‑off.

What is the Best Replacement for ISDN30?

The most direct, low-risk replacement for ISDN30 is SIP Trunking.

To understand why, we must first define the problem. ISDN30 (Primary Rate Interface or PRI) has been the backbone of corporate telephony for decades. It delivers 30 voice channels over a physical copper or fibre circuit. It is reliable, but it is inflexible, location-dependent, and increasingly expensive.

SIP (Session Initiation Protocol) changes the paradigm. Instead of physical wires carrying voice, SIP acts as a virtual “trunk” that connects your existing on-premise PBX to the PSTN via your data connection (internet or leased line). It emulates the behavior of ISDN but removes the physical constraints.

The Technical Evolution: From Circuit Switching to Packet Switching

The move from ISDN to SIP is a move from “Circuit Switching” (where a physical path is reserved for the duration of a call) to “Packet Switching” (where voice is broken into data packets and routed over IP).

For a Network Manager, this shift unlocks immediate benefits:

1. Efficiency: Bandwidth is only consumed when a call is active.

2. Convergence: Voice and data share the same pipe, eliminating the need for separate physical networks.

3. Flexibility: Channels are virtual software licenses, not physical line cards.

Detailed Comparison: SIP Trunking vs. ISDN30 (PRI)

For a deeper technical breakdown of legacy interfaces, read our ISDN Technical Guide: 2027 Retirement.

Why Multi-Site Businesses Should Choose SIP Trunking Over Hosted VoIP

In the rush toward the 2027 deadline, marketing campaigns often present “Hosted VoIP” (Cloud PBX) as the only solution. While excellent for smaller setups, Hosted VoIP can be a strategic error for multi-site enterprises with complex call flows.

SIP Trunking is the bridge that allows you to modernize your connectivity while preserving your customized internal infrastructure.

1. Retention of Advanced PBX Logic & Investments

Large organizations have often invested heavily in their PBX environment. You may have:

• Complex IVR Trees: “Press 1 for Sales, 2 for Support…” routed through intricate skill-based hunt groups.

• Call Centre Integrations: Screen-popping CRM data (Salesforce, Dynamics) when a call arrives.

• Proprietary Hardware: DECT systems in warehouses or specialized conference phones that are not compatible with generic hosted platforms.

Migrating to Hosted VoIP often requires rebuilding these flows from scratch. SIP Trunking allows you to keep the “Brain” (PBX) and simply replace the “Mouth” (Lines). The user experience for your staff remains identical; only the backend connectivity changes.

2. Multi-Site Efficiency: The “Centralised Trunking” Model

This is the unique value proposition for multi-site businesses that Hosted VoIP cannot easily replicate without high per-seat costs.

The Old ISDN Way:

• HQ (London): 2x ISDN30 (60 channels).

• Branch A (Manchester): 1x ISDN30 (30 channels).

• Branch B (Birmingham): 1x ISDN30 (30 channels).

• Total: 120 Channels paid for monthly, regardless of usage.

The New SIP Way:

• Data Centre / HQ: One single SIP Trunk with 70 Channels.

• Connectivity: All branches connect to the HQ PBX via MPLS, SD-WAN, or VPN.

• Result: You pool your capacity. It is statistically unlikely all three sites will be at peak capacity simultaneously. You pay for 70 channels instead of 120, immediately slashing fixed costs by ~40%.

3. Granular Control Over Numbering (DDI Management)

With SIP, your phone numbers (Direct Dialling Inwards – DDIs) are no longer tied to a specific telephone exchange. This allows for Centralised DDI Management. You can manage the numbering plan for your entire UK estate from a single web portal.

• Moving the Manchester office? No need to change numbers or pay for forwarding.

• Opening a virtual sales office in Bristol? Provision local 0117 numbers instantly and route them to your London call centre.

Still unsure? Compare the pros and cons in our VoIP vs SIP Migration FAQ.

The Technical Architecture: Integrating SIP into Legacy Environments

One of the most persistent myths is that older PBX systems cannot handle SIP. In reality, almost any business telephone system currently in operation can be converted to SIP, although the method varies.

Scenario A: Native SIP Integration (The Modern Path)

If your PBX is a modern IP-PBX (e.g., Avaya IP Office R10+, Mitel MiVoice, 3CX, Asterisk), it likely speaks SIP natively.

• Requirement: You will need to purchase “SIP Trunk Licenses” from your PBX vendor to “unlock” the capability.

• Configuration: You input the credentials provided by Stride (IP address, username, password) directly into the PBX network settings.

• Firewall: You must configure your firewall (or use an SBC) to allow traffic on UDP port 5060 (Signalling) and UDP ports 10000-20000 (RTP/Audio).

Scenario B: The Media Gateway (The Legacy Bridge)

If your PBX is older (TDM-based) and only has physical ports for ISDN cables, you do not need to throw it away.

• The Solution: An Enterprise Session Border Controller (E-SBC) or Media Gateway.

• How it Works: This hardware device sits between your internet connection and your PBX. It connects to the internet via Ethernet (receiving SIP) and connects to your PBX via a physical ISDN cable.

• The Conversion: It actively translates SIP data packets into ISDN electrical signals. To your PBX, it “looks” exactly like a traditional BT line. This extends the life of legacy hardware by 5-10 years.

5-Step Migration Plan: From ISDN30/PSTN to SIP Trunks

Successful migration is 80% planning and 20% execution. To ensure zero downtime, follow this industry-standard migration framework.

Step 1: The Estate Audit & Discovery

You cannot replace what you haven’t documented. ISDN invoicing is notoriously complex; do not rely on billing alone.

• Physical Audit: Trace cables. Identify every device plugged into a phone line. Watch out for “hidden” analogue lines used for Redcare alarms, lift emergency phones, franking machines, and PDQ terminals. These cannot run on SIP and require specialized 4G/GSM converters.

• Capacity Analysis: Request a “Traffic Report” from your current provider. Check your “High Water Mark”—the maximum number of concurrent calls you have made in the last 12 months. This determines how many SIP channels you actually need.

Step 2: Network Readiness Assessment (The QoS Check)

SIP Trunking is dependent on your internet connection. If your internet is slow or unstable, your voice calls will be robotic or drop completely.

• Latency & Jitter: Voice is intolerant of delay. Latency should be under 150ms; jitter under 30ms.

• Bandwidth Math: Use the G.711 codec standard. Allow 100Kbps (up and down) for every concurrent call. A 30-channel trunk requires a dedicated 3Mbps symmetric throughput.

• Segregation: For estates with >10 channels, we strongly recommend a dedicated internet line for voice, or a partitioned VLAN with strict Quality of Service (QoS) rules on the firewall.

Step 3: Architecture Design & Number Porting

This is the most critical logistical step. Utilizing a professional number porting service is essential to ensure your DDI ranges move seamlessly without downtime.

• Porting Lead Times: Moving numbers from the BT Openreach network to a SIP network is a regulated process. Category A (Simple) ports take ~10 days. Category C (Complex/Multi-site) ports can take 30+ days. Start this process early.

• Disaster Recovery Plan: Configure your “Failover Routing” before the port happens. Where should calls go if your office loses power? (e.g., “Divert all to mobile 07700…” or “Divert to backup site B”).

Step 4: Pilot Testing (The “Parallel Run”)

Do not cut over on a Monday morning without testing.

• Inbound Test: Port a single, non-critical DDI number first to verify routing.

• Outbound Test: Configure the PBX to route specific outbound calls via the SIP trunk while keeping ISDN for inbound.

• Load Testing: Make multiple simultaneous calls to check for “clipping” or echo.

Step 5: The “Green Light” & Decommissioning

Once the numbers port successfully:

1. Update your PBX routing tables to make the SIP Trunk the “Primary Route” for all calls.

2. Monitor for 48 hours.

3. Crucial Step: Formally give notice to cancel your ISDN30 lines. They will not be cancelled automatically, and you will continue to be billed if you do not act.

SIP Trunking FAQs for PSTN/ISDN Migration

Common questions technical managers ask during the 2027 switch-off.

Can SIP Trunking replace my ISDN30 lines directly?

Yes. SIP is the functional equivalent of ISDN30. It provides the same DDI presentation, Caller ID, and audio quality (often better, thanks to HD Voice codecs), but adds flexibility and lowers costs.

How does the 2027 Stop Sell affect my current contract?

The “Stop Sell” (already active in many UK exchanges) means you cannot add new supply. You cannot buy new ISDN channels or modify your existing setup. If you need to expand before 2027, SIP is your only option.

Is SIP secure? What about hacking?

SIP travels over the public internet, so security is vital. Stride Communications implements IP Authentication (locking the trunk to your office IP) and Toll Fraud Protection (automatically capping international spend if suspicious activity is detected). We also recommend using an SBC (Session Border Controller) for encryption.

What happens if my internet goes down?

With ISDN, a line fault meant “dead air.” With SIP, the intelligence is in the network. If your office internet fails, the Stride network detects the timeout and instantly re-routes your inbound calls to a pre-set backup destination (mobile, another branch, or voicemail-to-email).

How does SIP Trunking support 999 Emergency Services?

Ofcom regulations require all VoIP/SIP providers to support access to 999/112. When setting up your SIP trunk, you must register a physical address (Emergency Address Location) for each number. This ensures that if a user dials 999, the operator can see the correct location of the caller, even if the “lines” are virtual.

The Financial Case: ROI of SIP vs. ISDN

For the CFO, the migration is a financial upgrade. The “Total Cost of Ownership” (TCO) for SIP is significantly lower than legacy telephony.

• Line Rental Savings: ISDN30 typically costs £14.00 – £16.00 per channel/month. SIP trunks typically cost £3.00 – £6.00 per channel/month. (Approx. 60-70% saving).

• Call Cost Savings: Legacy ISDN contracts often charge setup fees and per-minute rates for local/national calls. Modern SIP trunks usually include “bundled minutes” covering 01, 02, 03, and 07 numbers.

• Hardware Savings: By centralizing multi-site trunks into one “Head Office” connection, you reduce the need for physical line cards and maintenance contracts at every remote branch.

• Unified Communications (UC) Readiness: Switching to SIP is often the first step toward enable video conferencing, softphones, and advanced VoIP phone systems features that legacy ISDN simply cannot support.

Conclusion: Act Before the “2027 Bottleneck”

The 31 January 2027 deadline is absolute, but the operational risks begin much sooner. As we approach the cut-off, the UK telecommunications industry will face a massive bottleneck of engineering resource. Lead times for number porting will extend from weeks to months.

Waiting until late 2026 to migrate your complex multi-site estate is a high-risk strategy. By moving to SIP Trunking now, you secure your numbering plan, reduce your operational costs immediately, and ensure your business communication remains resilient in the all-IP future.

Ready to Architect Your Solution?

Don’t let the PSTN switch-off become a crisis. Whether you need a technical audit of your multi-site topology or a quote for high-capacity SIP trunks, Stride Communications is your migration partner.

• Explore: Stride SIP Trunking Features & Pricing

• Next Step: Download the Technical Migration Checklist

• Contact Us: Speak to a SIP Solutions Architect today for a free infrastructure consultation.

• “Not an enterprise? If you are a startup or local firm, explore our tailored small business phone systems for a plug-and-play alternative to SIP.”

About the Author Written by: Bhav > Telecommunications Migration Specialist > Bhav is a veteran in the UK telecoms landscape, specializing in multi-site IP-PBX architecture and legacy-to-IP transition strategies. With a deep technical background in ISDN PRI environments, he has successfully led hundreds of UK businesses through the pre-2027 PSTN migration process.

Article Freshness & Accuracy Last Updated: January 03, 2026

Fact-Checked by: Stride Technical Engineering Team

This guide is reviewed quarterly to reflect the latest Openreach “Stop Sell” exchange updates and Ofcom regulatory changes.