IPComms
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SIP Trunking Service.
$0.009/min. No Contracts.

Low-cost SIP trunking with no per-channel fees. Connect your Asterisk, FreePBX, 3CX, or Teams PBX in minutes. Inbound and outbound trunks with TLS/SRTP encryption and real human support.

23+
Years in Business
99.99%
Uptime SLA
$0
Per-Channel Fees
8+
PBX Platforms Supported

What is SIP Trunking?

SIP trunking replaces traditional phone lines by connecting your PBX to the telephone network over the internet using the Session Initiation Protocol (SIP). Instead of paying for expensive PRI circuits or analog lines, your voice traffic travels over your existing broadband connection — cutting costs by 40–60% while adding flexibility and scalability.

Traditional Phone Lines

Expensive PRI circuits ($300–$800/month), limited channels, long provisioning times, and inflexible capacity. You pay for lines whether you use them or not.

SIP Trunking with IPComms

$0 per channel — no trunk fees. Pay only for usage ($0.009/min) and numbers ($1.50/mo). Instant provisioning, unlimited scalability, and works with your existing PBX.

Enterprise-Grade Security

TLS signaling and SRTP media encryption on every call. E911 compliant, CNAM support, and redundant infrastructure across multiple data centers.

AI-Powered Migration

Switching to SIP
shouldn't be complicated.

Our AI assistant knows your PBX inside and out. Tell it what system you're running and it generates your exact configuration — credentials, codecs, TLS settings, and all. No guesswork, no support tickets, no waiting.

PBX Config Generator

Copy-paste configs for Asterisk, FreePBX, 3CX, and more

Migration Checklist

Step-by-step walkthrough for your specific setup

Call Troubleshooting

Diagnose one-way audio, registration, and codec issues

Number Porting Help

Documents, timelines, and zero-downtime cutover planning

Try the AI Assistant
IPComms AI
Online

I'm migrating from a PRI to SIP. I run Asterisk 20 with PJSIP.

Here's your PJSIP trunk config:

[trunk-ipcomms]
type = registration
transport = transport-tls
server_uri = sip:sip.ipcomms.net
...

I've enabled TLS 1.3 + SRTP. Want me to generate your dialplan too?

Why Businesses Choose IPComms SIP Trunking

Low Cost, No Compromise

$0.009/min

Enterprise-grade trunking at a fraction of the cost. Twilio charges $0.014/min, traditional telcos $0.03–0.05/min. IPComms includes TLS 1.3 and SRTP encryption on every call at no extra charge — no per-channel fees, no monthly minimums, no contracts.

Inbound Trunks + Local Presence

300+ area codes

Local DIDs from $1.50/mo, toll-free from $2.00/mo. Every inbound trunk includes CNAM caller ID, E911 compliance, and failover routing. Need to port existing numbers? Free for 100+ numbers.

Scales With Your Business

1,000+ concurrent calls

No channel limits — unlimited concurrent calls bounded only by your bandwidth. Each G.711 call uses ~100 Kbps. Call centers and healthcare orgs trust our geo-redundant infrastructure.

How SIP Trunking Works

Replace expensive PRIs and analog lines with flexible, cost-effective SIP trunks.

1

Connect Your PBX

Point your existing PBX (Asterisk, FreePBX, 3CX, or any SIP-compatible system) to our SIP servers. We provide clear configuration guides for every major platform.

2

Get Phone Numbers

Choose local or toll-free numbers from our inventory, or port your existing numbers. Numbers are provisioned instantly through our customer portal.

3

Start Making Calls

Your calls route over the internet to our carrier network. You get crystal-clear voice quality, detailed CDRs, and pay only for what you use.

Pricing

No channel fees. No contracts. Pay only for what you use.

SIP Trunks

$0 /channel/mo
  • Unlimited concurrent calls
  • TLS/SRTP encryption included
  • No monthly minimums

Calling Rates

Outbound USA/CA $0.010/min
Inbound Local $0.009/min
Inbound Toll-Free $0.0185/min
Local DID $1.50/mo
Toll-Free DID $2.00/mo
E911 Service $2.50/mo

Get Started

  • $0/channel — pay only for usage
  • Pick a number & go live in minutes
  • Cancel anytime

Everything You Need

Enterprise features included with every SIP trunk.

Caller ID

Set any of your DIDs as outbound caller ID. Full CNAM support.

Failover Routing

Configure backup destinations. Calls automatically reroute if your primary PBX is unreachable.

Detailed CDRs

Complete call detail records with duration, cost, and routing info. Export anytime.

E911 Compliant

Full E911 support with location registration. Keep your team safe.

Real-Time Portal

Self-service portal for DIDs, routing, billing, and usage monitoring.

API Access

RESTful API for provisioning, CDR retrieval, and account management.

SIP Trunking FAQ

Common questions about SIP trunking and how IPComms works.

SIP trunking is a method of sending voice and other communications over the internet using the Session Initiation Protocol (SIP). It replaces traditional analog phone lines and expensive PRI circuits by connecting your PBX phone system to the public telephone network (PSTN) through a broadband internet connection. This eliminates the need for physical phone lines and typically reduces calling costs by 40–60%.

IPComms SIP trunks are free — there are no per-channel or per-trunk fees. You only pay for usage: outbound calls to the USA/Canada cost $0.010/min, inbound local calls cost $0.009/min, and local DIDs are $1.50/month. There are no contracts and no monthly minimums.

SIP trunking works with any SIP-compatible PBX system including Asterisk, FreePBX, 3CX, Microsoft Teams (Direct Routing), Cisco CUBE/CUCM, Avaya, Genesys Cloud, FreeSWITCH, and most other modern IP-PBX platforms.

VoIP (Voice over Internet Protocol) is the broad technology for making phone calls over the internet. SIP trunking is a specific application of VoIP that connects an on-premises or cloud PBX to the telephone network. Think of VoIP as the umbrella term, and SIP trunking as the bridge between your business phone system and the outside world.

With IPComms, you don't need to worry about trunk counts. Our service has no per-channel fees — you can make as many simultaneous calls as your internet bandwidth supports. A general rule of thumb is that each concurrent call requires about 100 Kbps of bandwidth.

Yes. IPComms supports number porting for local and toll-free numbers across the United States. The porting process typically takes 7–14 business days for local numbers and 5–10 business days for toll-free numbers. You can start using IPComms immediately with new numbers while your existing numbers are being ported.

Ready to upgrade your voice infrastructure?

Get started in minutes with instant provisioning. No contracts, no commitments.