Infrastructure for the Modern Enterprise

SIP Trunking
Explained Simply

Replace your obsolete phone lines with a scalable, cloud-based connection. SIP Trunking connects your business to the world over the internet, cutting costs and boosting reliability.

Your PBX

On-premise or Cloud Phone System

SIP TRUNK
VoIP / Internet Connection
The World

PSTN / Mobile / Landlines

What is a SIP Trunk?

SIP (Session Initiation Protocol) is the standard technology for establishing voice and video connections over a data network. A SIP Trunk is essentially a virtual version of an analog phone line.

Unlike traditional ISDN/PRI lines that require physical copper wires to be installed at your office, SIP trunks work over your existing internet connection. This allows you to connect your Private Branch Exchange (PBX) to the Public Switched Telephone Network (PSTN) without hardware restrictions.

Two-Way Communication

Handles both inbound and outbound calls simultaneously.

Elastic Scalability

Add unlimited channels instantly during high traffic periods.

Secure Transmission

Calls are encrypted using TLS and SRTP protocols.

Hardware Independent

Works with IP-PBX, softphones, and legacy systems (via gateways).

SIP Trunking vs. Traditional ISDN/PRI

Why businesses are switching to SIP en masse.

FeatureSIP TrunkingISDN / PRI
Setup TimeInstant (Minutes)Weeks or Months
Cost StructurePay-per-channel or minuteExpensive fixed monthly rental
ScalabilityUnlimited, InstantLimited to 23/30 channels per wire
HardwareNone (Virtual)Physical Cards & Wiring
LocationWork from anywhereTied to physical office
RedundancyAuto-failover to mobile/cloudLine down = No calls

Why Your Business Needs
SIP Trunking

It's not just about cheaper calls. It's about building a communication infrastructure that grows with your business.

  • Cost Savings up to 60%

    Eliminate expensive line rentals and reduce international call rates significantly.

  • Global Presence

    provision US/Canada digital numbers instantly, without physical offices.

  • Unified Communications

    Support voice, video, and messaging on a single network backbone.

StartACall SIP Features

Instant Provisioning
Real-time Call Analytics
Fraud Detection & Blocking
HD Voice (Opus/G.722)
E.164 Number Formatting
TLS/SRTP Encryption

Common Questions about SIP

Do I need to replace my current PBX?

Not usually. If you have a modern IP-PBX (like Asterisk, 3CX, Avaya IP Office), it supports SIP natively. If you have an older analog PBX, you can use a VoIP Gateway device to convert the SIP signal to analog, extending the life of your hardware.

How much bandwidth does a SIP trunk require?

A standard non-compressed call (G.711 codec) uses about 85-100 kbps (upload and download). We also support the Opus codec, which provides HD quality at lower bitrates, adapting to your network conditions.

Is SIP Trunking secure?

Yes, when configured correctly. StartACall uses TLS (Transport Layer Security) to encrypt signaling and SRTP (Secure Real-time Transport Protocol) to encrypt the audio media, preventing eavesdropping.

What happens if my internet goes down?

This is a major advantage of SIP. We can configure automatic failover forwarding. If your office internet drops, calls can be instantly rerouted to mobile phones, a different office, or a voicemail system in the cloud.

Ready to modernize your phone system?

Get enterprise-grade SIP trunking with pay-as-you-go pricing.

Sizing SIP Channels and Knowing When You Do Not Need a Trunk

In short

Once the concept of SIP trunking is clear, two practical questions remain: how many channels to buy, and whether your business needs a trunk at all. This section provides sizing rules of thumb, contrasts trunks with hosted PBX and plain browser calling for small teams, lists the network checks worth running first, and flags the migration traps that catch offices leaving ISDN.

Channels, Not Lines: Working Out How Many

A channel supports one concurrent call, so the question is never how many employees you have but how many conversations happen at once at your busiest moment. General offices commonly provision one channel for every three to five staff, while sales floors and support desks trend toward one per person.

Measure before you buy. A week of call logs shows your true peak concurrency, and most providers let channel counts move up or down monthly, so start slightly above the observed peak rather than paying for theoretical maximums that legacy circuits forced on buyers. Seasonal businesses should measure their busy season rather than an average week.

Keep channels and phone numbers separate in your head, because they are bought separately. Numbers are addresses while channels are capacity, and a company can point one hundred numbers at a trunk carrying ten channels as long as no more than ten conversations ever overlap.

SIP Trunk, Hosted PBX or Just a Browser

A SIP trunk assumes you own and run a PBX and only need carriage to the outside world. A hosted PBX moves the switching brain to a provider's cloud, trading control for convenience. Both models still exist to serve desk phones and call routing rules.

Small teams increasingly need neither. Browser-based WebRTC calling gives each person a dialer in Chrome, Safari, Firefox or Edge with no PBX, no trunk contract and no hardware, and StartACall prices this per minute on pay-as-you-go credit with no subscription.

The honest dividing line is call flow complexity. Businesses needing ring groups, IVR menus and wallboards belong on PBX infrastructure, while a five person consultancy making direct calls gets identical audio from a browser at a fraction of the administrative weight.

Network Readiness Before You Switch

Voice tolerates far less than web browsing. Aim for one-way latency under 150 milliseconds, jitter under 30 milliseconds and packet loss below one percent on the path to your provider, figures any network engineer can verify in an afternoon of testing.

Wire what you can. Desk phones and PCs handling calls behave far better on Ethernet than on shared Wi-Fi, and routers that support prioritizing voice traffic keep a large file upload from crumpling an important call. Test during your busiest hour, not overnight when results flatter the link.

Leaving ISDN Without Breaking Things

Port numbers in planned batches and run the old and new services in parallel through the transition, since a port that stalls mid process leaves customers ringing a dead line. Update direct dial routing and any published numbers before the final cutover date, not after.

Audit the forgotten devices. Fire alarm panels, lift emergency phones and card payment terminals often sit on analog lines that quietly depend on the circuit being retired, and each needs its own migration plan or adapter before the copper is switched off. Documenting which sockets feed which devices before the migration starts saves frantic tracing later.

Frequently asked questions

How many SIP channels does a small business need?+

Count concurrent calls at your busiest moment, not employees. Ordinary offices typically need one channel per three to five staff, while call-heavy teams approach one per person. A week of call logs gives the real number.

What is the difference between SIP trunking and hosted VoIP?+

A SIP trunk connects a PBX you own to the phone network, while hosted VoIP puts the PBX itself in the provider's cloud. Trunking suits businesses with existing equipment, hosting suits those who want nothing to maintain.

Can a small team skip SIP trunking entirely?+

Often yes. If you need direct calls rather than IVR menus and ring groups, browser-based WebRTC calling delivers carrier-grade audio with no PBX, no trunk and no contract, billed per minute as you use it.

Do fire alarms and lift phones work over SIP?+

Not automatically. These devices usually rely on analog circuits and need dedicated adapters or replacement services before the old lines close. Auditing them early is one of the most important steps in any ISDN migration.

What network quality does SIP calling require?+

As a rule of thumb, one-way latency under 150 milliseconds, jitter under 30 milliseconds and packet loss under one percent. Wired connections and routers that prioritize voice traffic help maintain those figures during busy periods.

Last reviewed June 2026Reviewed by the StartACall calling teamDialing rules cross checked against ITU international dialing procedures
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