If your business still pays for PRI lines or a bundle of traditional phone lines from a carrier, you are almost certainly overpaying. SIP trunking is the technology that replaces those physical connections with internet-based voice - and it has been the default choice for businesses with on-premise phone systems for the better part of a decade.

This is not a niche technical topic. The global SIP trunking market is projected to reach $33 billion by 2032, and the ISDN switch-off programmes already underway across Europe mean the decision is no longer optional for many businesses. If you have a PBX and you have not yet moved to SIP, you will need to eventually.

Here is what it is, what it costs, and how to decide whether it is right for your setup.

Key Takeaways

  • SIP trunking replaces physical phone lines with internet-based voice connections to your existing PBX
  • Cost savings typically run 25-65% compared to traditional PRI or PSTN lines
  • Pricing ranges from $0.01/min metered to $15-$25 per channel per month unlimited
  • You keep your PBX - SIP trunking is not a full phone system replacement, just the line side
  • Bandwidth is low - each concurrent call needs just 85-100 kbps, so standard business broadband handles it easily
  • Hosted VoIP (like RingCentral or 8x8 full UCaaS) is the better choice if you do not have a PBX or are ready to replace it entirely

What SIP Trunking Actually Is

SIP stands for Session Initiation Protocol - the signalling standard that sets up, manages, and ends voice and video calls over IP networks. A SIP trunk is a virtual connection between your on-premise phone system (PBX) and the public telephone network (PSTN), carried over your internet connection instead of physical copper.

Think of it as replacing a bundle of physical phone lines with a single internet-based pipe that can handle as many simultaneous calls as you configure. Your PBX still handles internal routing; the SIP trunk handles all external calls.

How It Works

When someone at your office makes an outbound call, the PBX converts the audio into data packets and sends them to your SIP provider via the internet. The provider connects that call to the PSTN and routes it to the destination number. Inbound calls work in reverse.

Each concurrent call uses approximately 85-100 kbps of bandwidth. A 100 Mbps business broadband connection can theoretically support hundreds of simultaneous calls, so bandwidth is rarely the limiting factor.

Session Border Controllers

Most enterprise SIP setups include a session border controller (SBC) - a device or virtual appliance that sits between your PBX and the internet. It handles security (SIP-specific firewall), codec transcoding, call routing redundancy, and toll fraud prevention. For smaller businesses, many SIP providers include SBC functionality in their service.

SIP Trunking vs. What You Might Already Have

The most common legacy alternatives are PRI lines (Primary Rate Interface, which carry 23 or 30 B-channels) and analogue PSTN lines. Both require physical installation, fixed capacity, and long lead times to scale.

Feature SIP Trunking PRI Lines Hosted VoIP (UCaaS)
Requires on-premise PBX Yes Yes No
Scalability Instant (add channels online) Weeks (physical install) Instant
Cost vs PRI 25-65% cheaper Baseline Similar or higher
International calls Low-cost via provider Expensive Varies
Number portability Yes Yes Yes
Setup time Hours to 24 hours Days to weeks Hours
Hardware needed PBX + internet PBX + physical lines None

SIP trunking is not the same as hosted VoIP. With SIP trunking, you keep your existing PBX and just replace the line connections. With hosted VoIP (platforms like RingCentral, 8x8, or Nextiva's full UCaaS product), the entire phone system lives in the cloud and you typically use software clients or IP handsets.

What SIP Trunking Costs

Pricing follows two main models:

Metered (pay-as-you-go): You pay per minute of call time. Twilio charges from $0.013/min for US calls; Plivo from $0.012/min. This works well for businesses with unpredictable or low call volumes.

Unlimited per channel: You pay a flat monthly fee per concurrent call channel. Typical range is $15-$25 per channel per month. A business running 10 simultaneous calls at peak would need 10 channels. At 50+ channels, expect volume discounts of 15-20%.

Setup fees range from $0 to a few hundred dollars - many providers waive these for longer contracts or higher-volume accounts. Number porting is often free, though budget providers may charge $5-$15 per number.

Realistic Monthly Bills

A business with 20 staff making moderate outbound calls might run 5-8 peak concurrent channels. At $20/channel that is $100-$160/month for unlimited UK/US calling - compared to several hundred pounds for an equivalent PRI circuit.

Which Businesses Should Switch

SIP trunking makes clear sense if you:

  • Have an existing PBX (Asterisk, FreePBX, 3CX, Cisco, Avaya) and want to reduce line costs
  • Have multiple sites and want to consolidate onto a single trunk
  • Have seasonal call volume spikes and need to scale channels up and down
  • Are facing ISDN switch-off and need an alternative
  • Make significant international calls (SIP rates beat traditional carriers substantially)

SIP trunking is probably not the right move if you:

  • Are replacing your phone system anyway - just move to hosted VoIP instead
  • Have fewer than 5 staff with simple phone needs - a UCaaS bundle is simpler and comparably priced
  • Have unreliable or low-bandwidth internet - voice quality depends on your connection

Choosing a SIP Trunk Provider

The community consensus on forums like r/VoIP and the FreePBX community consistently points to a few providers for different profiles:

  • voip.ms - popular with hobbyists and small businesses, low metered rates, very flexible
  • Telnyx - praised for support quality and developer-friendly APIs, competitive pricing
  • Flowroute - reliable, straightforward, good for mid-market
  • Twilio - best if you need programmable voice or are building custom call workflows
  • Nextiva - better for businesses that want hand-holding and a single vendor relationship
  • Vonage - strong choice if you need Microsoft Teams Direct Routing integration
  • Avoxi - worth considering for international-heavy businesses

Evaluate any provider on uptime SLA (99.9% is the minimum to expect), geographic redundancy, fraud monitoring, and whether they offer E911 routing if you are in the US.

What You Need Before You Switch

Before porting numbers and provisioning trunks, check:

  1. PBX compatibility - most systems from the mid-1990s onward support SIP natively; older analogue PBXes need a gateway
  2. Internet connection quality - test for packet loss and jitter, not just raw speed
  3. Firewall configuration - SIP uses UDP ports 5060/5061 and a range of RTP media ports; these need to be open or handled by your SBC
  4. Number porting timeline - allow 2-4 weeks for porting from most UK/US carriers
  5. E911 registration - if you are in the US, register your business address with your provider for emergency call routing

Frequently Asked Questions

Q: Do I need to replace my existing PBX to use SIP trunking? No. SIP trunking is designed to connect to an existing IP-capable PBX. If your PBX is older and analogue-only, you need a VoIP gateway, but replacement is not required.

Q: How many SIP channels do I need? Count the maximum number of calls your business handles simultaneously at peak. A business with 50 staff typically needs 10-20 channels, not 50. Most providers let you add channels instantly, so starting low and scaling up is a sensible approach.

Q: Is SIP trunking secure? SIP trunking carries real fraud risk if misconfigured - attackers probe for open SIP ports and exploit weak credentials to make international calls at your expense. A session border controller, strong credentials, and IP whitelisting mitigate this. Reputable providers also include toll fraud monitoring.

Q: Can I keep my existing phone numbers? Yes. Number porting transfers your existing geographic or non-geographic numbers to your new SIP provider. Most providers handle this free of charge as part of onboarding.

Q: What happens if my internet goes down? Most SIP providers offer failover routing - calls can be redirected to a mobile number or secondary circuit automatically. Configure this before you go live, not after your first outage.

Q: How is call quality with SIP? With a stable, low-latency internet connection, SIP call quality is indistinguishable from PSTN. Problems arise from packet loss, high jitter, or insufficient QoS configuration on your router. Prioritise voice traffic in your router settings using DSCP/QoS tagging.

The Bottom Line

SIP trunking is not a cutting-edge technology - it is mature, widely supported, and significantly cheaper than what most businesses running legacy phone lines are paying today. If you have a PBX and you are still on PRI or PSTN lines, the case for switching is straightforward. The main decision is not whether to move to SIP, but which provider fits your call volume and technical setup. Start with a metered provider like Telnyx or voip.ms on a trial basis, verify call quality on your specific network, then commit.