If you already own a PBX you like, paying to rip it out and replace it with a hosted system is wasted money. SIP trunking is the alternative — you keep the PBX, you swap the copper lines for an internet-delivered trunk, and your phone bill usually drops by 40% to 60%. Here's how it actually works and what we charge for it.
We're VoIP International, an operator out of Ocoee, Florida. We sell SIP trunking direct, no reseller in the middle, and we'll explain the parts honestly.
What a SIP trunk is, without the jargon
SIP stands for Session Initiation Protocol. A SIP trunk is a virtual phone line delivered over your internet connection instead of a physical copper pair or a T1. Your PBX — whether it's a 3CX, FreePBX, Avaya, Mitel, or anything else that speaks SIP — points at our trunk, and calls travel as data packets to the public phone network.
Two things to know:
- A channel is one simultaneous call. Buy four channels, you can have four people on calls at once. Five would get a busy signal. Most businesses need fewer channels than they think — count peak concurrent calls, not total employees.
- DIDs are the phone numbers. They're separate from channels. You can have 50 numbers ringing into 8 channels, no problem.
What "replacing copper" actually means
The physical copper line from your office to the telephone company is called a POTS line, or in higher-volume installs, a PRI (Primary Rate Interface) which carries 23 voice channels on a T1. Both are end-of-life products. POTS lines are being repriced upward by every regional carrier as part of FCC sunset rules, and PRI service contracts often run $400 to $1,200 a month before any usage. SIP trunking does the same job — delivers PSTN access to your PBX — over an internet connection you already have, at a fraction of the cost.
How to size your channels honestly
This is where most providers overcharge customers. A 30-person office doesn't need 30 channels. It needs enough channels to cover the busiest hour. A standard ratio for a normal office is one channel per three to five users. A 30-person business is usually fine with 8 to 10 channels, and a contact center with 30 agents on the phone all day needs closer to 30.
How to figure it out:
- Pull a month of call detail records from your current provider.
- Look at the busiest 15-minute window.
- That's roughly your channel count, plus a small buffer.
If you don't have CDRs, count phones that are likely to be on a call at the same time. We'll help you size it on a 15-minute call — we lose nothing by selling fewer channels and we gain a customer who isn't paying for capacity they don't use.
The metered alternative
If your call volume is unpredictable, or you have seasonal spikes, you can skip channel pricing and pay per-minute instead. Our metered rate is $0.015 outbound, $0.005 inbound. For a business doing a few hundred outbound minutes a day, metered is often cheaper than channels. We'll model both on a quote so you can see which way the math goes.
Our pricing, posted publicly
- $15 per channel per month. Unlimited talk inside that channel.
- $0.015 per minute outbound if you'd rather meter than buy unlimited channels.
- $0.005 per minute inbound.
- $15 per number to port existing DIDs in. One-time.
- E911 included. A misdialed 911 from a bad address record is $150 — set the addresses right.
A typical 25-user office with 8 concurrent channels runs $120 per month for the trunk, plus traffic. Compared to a PRI at $400 to $700, the math works fast. Full breakdown on the pricing page.
What's not in the per-channel price
To be upfront: the per-channel rate doesn't include international termination beyond standard US/Canada, doesn't include toll-free numbers (those are $2/month per number plus per-minute), and doesn't include vanity number reservations. None of these are unusual carve-outs, but ask any provider you compare against whether their advertised channel rate includes them.
When SIP trunking is the right move (and when it isn't)
Use SIP trunking when:
- You already own a PBX that's working and paid for.
- You have a contact center with a dialer that expects SIP.
- You're running multiple offices and want to consolidate trunks at HQ.
- You need a backup trunk for disaster recovery on your primary PSTN service.
- Your IT team is comfortable with SIP signaling and can configure the PBX side.
- You want to keep a specific compliance setup that already lives on your PBX.
Skip SIP and go with hosted phone service when:
- Your PBX is end-of-life and you'd be replacing it anyway.
- You have a small team (under 10) and don't want hardware in a closet.
- You want mobile apps, integrations, and call recording without engineering them into an on-prem box.
- You don't have an IT person who wants to own the PBX.
- Your PBX vendor is no longer supporting your model.
What you need on your end
Two things make SIP trunks behave well: a stable internet connection with enough upstream bandwidth (about 100 Kbps per concurrent call on G.711, less on G.729), and a router or session border controller that can handle SIP and prioritize voice traffic. A dedicated internet line for voice isn't required, but on saturated business broadband it's worth considering — it keeps calls clean during big file uploads or backups.
If your router doesn't support SIP ALG correctly (and many don't, even ones that claim to), turn it off. SIP ALG is one of the top sources of one-way audio and dropped calls on small business networks. We'll tell you which routers we've seen behave well with our trunks.
Codecs and bandwidth
Most SIP trunk deployments use G.711 (about 87 Kbps per call including overhead) or G.729 (about 32 Kbps). G.711 sounds better; G.729 saves bandwidth. We default to G.711 because most modern broadband can handle it without anyone noticing. If you're remote and on satellite or LTE, we'll set you to G.729. Either way, calls are encrypted in transit (TLS for signaling, SRTP for media) on supported PBX endpoints.
Failover and disaster recovery
This is one of the reasons businesses move to SIP even if they liked their old PRI. A SIP trunk can fail over in seconds — to a mobile app, to another office, to a backup SIP provider, or even to a list of cell phones. If your PBX dies, your fiber gets cut, or the building floods, calls keep ringing somewhere.
We build the failover plan with you when we provision the trunk. Standard configuration includes:
- Automatic rerouting to a backup destination if the primary trunk endpoint stops responding
- Override-via-portal so you can change the destination from a phone if the PBX is unreachable
- Trunk reporting that flags packet loss or jitter so you see problems before users complain
- Optional SMS notification when the trunk goes down or comes back up
What the failover looks like in practice
A typical setup: primary trunk to the PBX at the main office. Secondary trunk to a backup destination — could be a hot-standby PBX at another office, could be a list of mobile numbers, could be a hosted IVR that takes a message and emails it to your team. If the main trunk doesn't respond to a SIP INVITE inside three seconds, the call routes to secondary. Callers never hear the transition. Most disaster events take less than 10 seconds to fully cut over.
If you want UC without replacing the PBX
One option we set up often: keep the PBX, layer our Microsoft Teams integration on top. You can run Teams as the front end for users who live there, and route calls through the trunk you already have. New users get Teams, old users keep their desk phones. No forklift upgrade. This is the cleanest path for organizations that have invested in PBX hardware recently but want to give mobile and remote workers a softer phone experience without provisioning new hardware.
Real-world cost comparisons
A few scenarios we see often:
25-person professional services firm
Current: PRI at $580/month, plus $0.04/min outbound long distance. Average bill: $720/month. Move to SIP: 8 channels at $15 = $120/month. Outbound at $0.015/min averages $90/month. New bill: $210. Annual savings: about $6,100. Payback on porting and minor PBX configuration: under one month.
3-location retail group
Current: 3 POTS lines per store at $55 each, no consolidation. Total: $495/month for 9 lines. Move to centralized SIP trunk at HQ with cross-location routing: 6 channels at $15 = $90/month. Savings: $405/month, plus the operational win of consolidated reporting and one auto-attendant. See multi-location for the rollout pattern.
Inbound call center
Current: 24-channel PRI at $1,100/month plus heavy inbound minutes. Move to SIP: 24 channels at $15 = $360/month, inbound at $0.005/min. Savings often exceed $1,000/month after minutes.
Common mistakes
- Buying too many channels. Count concurrent calls, not employees.
- Skipping QoS on the network. Voice needs priority over file transfers. If your router can't tag DSCP or shape traffic, calls will get choppy when someone backs up to the cloud.
- Leaving SIP ALG on. Disable it on Cisco, Sonicwall, Meraki, and any consumer router. The exceptions are rare.
- Not testing failover. Unplug the PBX once a quarter and see if the failover works. Find out now, not the day the building loses power.
- Forgetting to update E911 addresses when the office moves or a remote rep changes home addresses.
- Underestimating bandwidth. If your office shares a single 50 Mbps cable line and you peak at 12 concurrent calls plus heavy uploads, you'll have call quality complaints. Look at upstream first.
- Not verifying STIR/SHAKEN attestation. If your outbound calls are flagged "Spam Likely," answer rates collapse. Ask your provider about attestation levels.
What to ask a SIP provider
- Is the trunk delivered from your own switching, or are you reselling another carrier?
- What's the published per-channel and per-minute rate? Any minimums?
- Are you fully STIR/SHAKEN attesting so my outbound calls don't get flagged as spam?
- What's the porting timeline, and what's the per-number fee?
- How do I configure failover, and what does the portal expose?
- Who answers if my trunk goes down at 2 a.m.?
- What's your stance on international destinations and abuse handling?
- Can I get a test trunk to validate against my PBX before porting?
What we don't do
We don't sell SIP trunks to gray-market international call termination businesses. We don't help with carrier fraud workarounds. We don't promise unlimited international minutes at suspiciously low rates. We're a US-based operator and we treat compliance and abuse seriously, because regulators and downstream carriers do too. We'd rather decline a customer than be the carrier of last resort for someone running a fraud operation.
PBX compatibility, in practice
Most modern PBX platforms work with our SIP trunks without custom development. We see the following most often:
3CX
3CX has a built-in SIP trunk template wizard. We provide credentials and registrar info; you fill in the wizard and the trunk comes up in 10 minutes. Direct outbound and inbound work immediately. Failover and codec settings are exposed in the GUI.
FreePBX / Asterisk
The classic open-source PBX. Slightly more configuration on the PJSIP trunk side, but every option we need is exposed. Customers with experienced FreePBX admins can configure and test in under an hour.
Avaya IP Office
Common for small and mid-market businesses that bought systems 5 to 10 years ago. SIP trunk profile is configurable; the catch is making sure the system's licensing covers SIP rather than legacy TDM channels. We've helped customers identify what licensing they need before they buy.
Mitel MiVoice
Multiple flavors of Mitel exist; SIP trunk support varies by version. We typically run a proof-of-concept call before committing to a port to confirm the specific platform behaves.
Cisco UCM / CUCM
The enterprise voice platform. SIP trunk integration is straightforward; the configuration involves more steps than a small business PBX, but Cisco voice engineers know the routine.
Microsoft Teams (Direct Routing)
For customers who want Teams as the front end without buying Microsoft's calling plans, we provide Direct Routing-compatible SIP trunks. See Microsoft Teams for the integration overview.
If your PBX isn't on this list, ask. We've worked with platforms that no one's heard of in 10 years. The honest answer in some cases is "your PBX is end-of-life, you're better off moving to hosted" — and we'll tell you that rather than sell you a trunk for a dead box.
Where to start
Tell us your PBX type, how many concurrent calls you actually peak at, and how many DIDs you want to keep. We'll quote the build, port your numbers, and stand it up. We can provide a test trunk first if you want to validate against your PBX before committing. If you're comparing against bigger names, our vs RingCentral page covers the differences. Contact us, get started, or read the FAQ. More background on who we are.