In this guide, we’ll walk you through everything you need to know about how to migrate from PRI to SIP successfully. From assessing your current setup and avoiding common pitfalls to choosing the right provider and optimizing your voice infrastructure, this practical step-by-step guide will help you plan, execute, and refine your migration with confidence.
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TL;DR — Key Takeaways for Migrating from PRI to SIP
SIP is replacing PRI as legacy phone lines are phased out globally. It’s more scalable, flexible, and cost-effective.
A successful migration starts with proper planning: assess your current setup, ensure hardware compatibility, and identify legacy dependencies like fax or alarms.
The transition involves parallel testing, careful configuration, and number porting. A phased approach minimizes risk.
SIP improves business communications with lower costs, better redundancy, and global reach.
Main Questions when migrating from PRI to SIP
- Is SIP better than PRI?
Yes — in most modern business environments, SIP trunking outperforms PRI in flexibility, scalability, and cost-efficiency. While PRI is hardware-based and limited to 23 channels per circuit, SIP operates over the internet and can scale easily with your needs, without requiring physical lines.
- Can SIP trunks improve call quality?
Yes. When implemented with proper network setup and QoS (Quality of Service), SIP trunks can deliver crystal-clear voice quality, often better than aging PRI lines.
Why You Should Migrate from PRI to SIP?
The shift from PRI (Primary Rate Interface) to SIP trunking is being driven by both technological evolution and necessity. Globally, telecom operators are decommissioning PSTN and ISDN infrastructure, which means PRI circuits are gradually being phased out. Maintaining legacy systems has become increasingly expensive, with limited availability of replacement parts, higher support costs, and shrinking carrier support.
By the beginning of 2027, BT, along with other big phone companies in the UK, will switch off PSTN and shift completely to digital services such as VoIP.
SIP trunking offers a modern, IP-based alternative that aligns with today’s digital business needs. With SIP, you benefit from cost savings, on-demand scalability, easier integration with cloud platforms, and greater geographic flexibility.
Here’s a side-by-side comparison of PRI vs SIP:
| Feature | PRI | SIP Trunking |
|---|---|---|
| Infrastructure | Physical copper lines | Internet-based (VoIP) |
| Scalability | Fixed (23 channels per circuit) | Virtually unlimited |
| Setup | On-site hardware required | Cloud-ready, minimal hardware |
| Cost | High CAPEX and maintenance | Lower OPEX, pay-as-you-go options |
| Flexibility | Limited | Highly flexible (add/remove channels) |
| Global Reach | Limited to local carrier zones | Global DIDs and routing are possible |
| Compatibility | Tied to legacy systems | Works with modern PBX and UCaaS |
| Availability | Declining | Growing rapidly worldwide |
What You Need to Know Before You Migrate From PRI to SIP
Before you migrate from PRI to SIP, there are five key areas you need to assess: your current telephony setup, PBX compatibility, analog dependencies, network readiness, and security posture. Evaluating these up front will help avoid migration delays, service interruptions, and unexpected costs.
Take Inventory of Your Current Telephony Setup
Start by documenting your existing environment. List all PRI circuits, DID ranges, hunt groups, IVR paths, call queues, and any integration points like CRMs or contact centers. This inventory will act as your baseline and ensure nothing is lost during the transition. It also helps plan number porting and routing rules for the SIP setup.
Check PBX and Hardware Compatibility
Next, evaluate whether your existing PBX supports SIP natively. Many modern systems do, but older PBXs may require a SIP-to-TDM gateway or a Session Border Controller (SBC) to bridge the gap. Confirm compatibility with your provider and identify any upgrades or hardware requirements needed for a smooth integration.
Identify Analog and Legacy Dependencies
PRI lines often support more than voice. Look for devices or systems that rely on analog signaling, like fax machines, alarm systems, building entry intercoms, paging systems, or credit card terminals. Not all of these work well over SIP, and you may need adapters or keep a few analog lines during the transition.
Assess Network & Security Readiness
SIP runs over your IP network, so bandwidth and quality of service (QoS) matter. Assess your internet capacity to ensure it can handle the expected number of concurrent calls without congestion. Implement QoS to prioritize voice traffic and review firewall/NAT policies to allow SIP signaling and RTP audio. For security, ensure TLS and SRTP support, and review your basic VoIP security posture (rate limiting, IP whitelisting, etc.).
How to Migrate from PRI to SIP? Step-by-Step Process
Migrating from PRI to SIP requires a clear and phased approach. By breaking the process down into defined steps, you reduce disruption, minimize risk, and ensure your new SIP setup is optimized from the start. Let’s begin with the first two foundational stages.
Step 1 – Design Your Future Voice Architecture
Map out your numbering plan. Will you port all existing numbers to your new SIP trunk provider, or keep a few PRI lines temporarily for legacy systems? Consider whether you need new number ranges, toll-free numbers, or international DIDs. For larger environments, it’s smart to review how numbers are assigned across teams, departments, and locations.
Don’t overlook redundancy. A reliable SIP deployment should include multiple trunks, geographic routing options, and possibly backup internet links. This is your chance to build a more resilient voice setup than what was possible with fixed PRI lines.
Step 2 – Choose the Right SIP Trunking Provider
The provider you choose plays a major role in the reliability, call quality, and cost-effectiveness of your new voice system. Look for a provider with:
Coverage in all your active and target markets
Transparent pricing models (per channel, per minute, bundles, etc.)
High-quality routing options (standard vs premium)
Support for number porting and regulatory compliance
Security features like TLS/SRTP, fraud protection, and IP authentication
Excellent support and onboarding assistance
Step 3 – Set Up and Configure Your SIP Trunks
The next step is technical configuration. Your SIP trunking provider will issue the necessary credentials, IP ranges, and setup parameters. These need to be entered into your PBX or SBC (Session Border Controller), depending on your setup.
You’ll also configure the following:
Codecs: Use G.711 for high-quality audio or G.729 for bandwidth-limited environments.
Dial plans and routing rules: Adjust for international prefixes, emergency numbers, toll-free destinations, and internal extensions.
Caller ID behavior: Decide how your outbound caller ID is displayed (main line vs. user DID).
Failover logic: In case of trunk failure, calls should automatically reroute to backup trunks or destinations.
Step 4 – Run SIP and PRI in Parallel
Before cutting over completely, it’s best practice to run your new SIP trunk setup alongside your existing PRI lines. This dual-path environment allows for testing and minimizes risk. Start by routing a limited number of calls over SIP while the rest continue over PRI.
During this phase, monitor:
Call quality (jitter, latency, packet loss)
Inbound and outbound routing behavior
Interactions with IVRs, call queues, voicemail, and call recording systems
Firewall and SBC logs for any blocked or malformed SIP traffic
Step 5 – Test Everything Thoroughly
A full round of testing is essential to ensure your SIP trunking is fully functional. This includes verifying inbound and outbound call handling, IVRs, voicemail, call queues, and call recordings. Don’t forget to test edge cases, like international calls, emergency dialing, and any integrations with analog systems such as fax machines or alarm lines.
If you’re using failover or backup trunks, simulate trunk failures to confirm that routing works as expected. Treat this stage like a quality assurance process—log everything, fix inconsistencies, and gather feedback from users who interact with the phone system daily.
Step 6 – Plan and Execute the Cutover
Now you can schedule the final cutover from PRI to SIP. It’s best to do this during low-traffic periods, such as evenings or weekends, to reduce risk. You’ll work with your SIP provider to port over numbers and redirect inbound/outbound traffic from PRI to SIP. During the switchover, actively monitor your call quality, call completion rates, and system performance. Ensure your IT and telephony teams are on standby to address any last-minute configuration issues or routing hiccups.
Common Pitfalls When Migrating from PRI to SIP
The most common pitfalls for businesses when migrating from PRI to SIP come from skipping key steps, underestimating technical needs, or choosing the wrong partner. Here are the most common mistakes, and how to avoid them:
Overlooking Analog Devices and Legacy Lines: Fax machines, alarms, door systems, and paging equipment often still rely on analog connections. Failing to account for these during planning can lead to service disruption or unexpected replacement costs.
Skipping or Rushing the Parallel Testing Phase: Migrating without a proper pilot run is a recipe for outages. Running SIP and PRI side-by-side during the initial stage helps uncover configuration or quality issues in a safe environment.
Ignoring Network QoS Requirements: SIP is only as good as your internet connection. Without proper network configuration (including bandwidth checks and QoS settings) call quality may suffer, and the provider may be wrongly blamed.
Migrating Without a Rollback Plan: Even the best migrations can hit a snag. Not having a clear rollback or failover plan in case things go wrong can turn a temporary issue into a major outage.
Best Practices for a Smooth PRI to SIP Migration
To make your migration from PRI to SIP as seamless and risk-free as possible, it’s essential to follow a few best practices. These steps can prevent downtime, ensure a better user experience, and help you get the most value from your SIP solution:
Start with a Clear Inventory and Call-Flow Map: Know exactly what’s running on your current phone system before making any changes.
Involve Network, Security, and Support Teams Early: SIP touches multiple parts of your infrastructure. Bringing IT, security, and end-user support into the planning process helps avoid surprises and ensures everything is aligned from the start.
Use a Structured Test Plan with Checklists: Define success criteria in advance and test systematically. This includes call routing, failover, voicemail, and fax scenarios — not just basic call quality.
Communicate the Change to Stakeholders and End Users: Keep employees informed about what’s changing, why it matters, and what they might need to do differently. This minimizes confusion and ensures smoother adoption.
Treat SIP as Core Infrastructure, Not Just “Another Telco Contract”: Your SIP trunking service is a critical part of your business communications. Choose a provider that offers visibility, security, scalability, and ongoing support.
How Telxi Makes PRI to SIP Migration Easy
If you’re ready to migrate from PRI to SIP, we make the process fast and simple. Our team provides hands-on planning support, instant SIP trunk activation, and transparent pricing. With 24/7 expert assistance, full compatibility with leading PBX systems, and built-in enterprise-grade security, we can help you move to modern voice infrastructure without disruption.
FAQ: Migrate from PRI to SIP
- How much does a PRI circuit cost?
A single PRI circuit can cost anywhere from $300 to $1,000 per month depending on the provider, location, and service terms. That includes 23 channels — but with SIP, you only pay for what you use, and typically at a much lower rate per channel or per minute.
- How many calls can one SIP trunk handle?
A SIP trunk itself is just the virtual connection — the number of concurrent calls you can handle depends on how many SIP channels are configured. Unlike PRI, which is fixed at 23 channels, SIP allows you to scale up or down easily based on demand.
- What is the 7 5 3 1 rule in SIP?
The 7-5-3-1 rule is a rough network engineering guideline for acceptable VoIP performance:
<7 ms jitter
<5% packet loss
<3 seconds of latency
<1% re-transmission rate
Meeting or exceeding these thresholds helps ensure high-quality SIP calls.
