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SIP trunks, hosted PBX and softphones: VoIP basics for small businesses

SIP trunk vs hosted PBX vs softphone for small businesses: what each one is, how they connect, what drives the cost and how to choose.

Published 11 min read

A SIP trunk is a connection that carries your calls to the public phone network over the internet, and a hosted PBX is a phone system that a provider runs for you. A softphone is software that acts as one of your extensions, so it can replace a desk phone. The practical choice is who runs the phone system: you, with an IP PBX and a SIP trunk for outside calls, or a provider, with a hosted PBX.

The three pieces

SIP trunk

A SIP trunk is a SIP connection between your phone system and a service provider. The provider supplies your phone numbers, usually called DIDs (direct inward dialing numbers), and carries calls to and from the public network. The SIP Forum's SIPconnect 2.0 recommendation calls itself a profile of SIP that "enables direct connectivity between a SIP-enabled Service Provider Network and a SIP-enabled Enterprise Network."

The trunk is not a phone. It is a set of rules: which server to talk to, how to authenticate, which codecs to offer and how many calls can run at once. If the provider gives you a domain name instead of an IP address, your PBX resolves it with the NAPTR and SRV lookups in RFC 3263 (sections 4.1 and 4.2), which also choose the transport. If you also enter a port, the PBX skips those lookups and does an A or AAAA lookup instead.

IP PBX

An IP PBX (PBX stands for private branch exchange) decides what happens to each call. It answers inbound calls, picks the extension that rings, plays the greeting, sends callers to a queue and stores voicemail. The PBX can run on a server in your office, which is called on-premise, or in a provider's data center, which is called hosted.

Asterisk is an open source PBX, distributed under the GNU GPL version 2, if you want to run your own. Start with its documentation at docs.asterisk.org before you build anything.

Softphone

A softphone is a SIP user agent in software. RFC 3261 defines a user agent as a logical entity that can act as both a user agent client and a user agent server (section 6), which is the role a SIP phone plays. A hardware desk phone and a laptop app do the same job at the protocol level. Both need a SIP server to talk to. For a registered account, you also need a username, a password and a transport.

How the pieces connect

A typical small office with one PBX and one trunk looks like this:

Desk phone ──┐
             ├── IP PBX ──── SIP trunk ──── provider ──── public phone network
Softphone ───┘

Each phone registers with the PBX. The PBX sends outbound calls through the trunk, and it accepts inbound calls that arrive on the trunk for your numbers and rings the right extension. In a hosted setup the PBX sits in the provider's network, so you do not run the trunk side yourself, but desk phones and softphones register to it in the same way.

Registration is the part that confuses people. Before a phone can receive a call, it has to tell the PBX where it can be reached, and it does that with a REGISTER request (RFC 3261, section 10). A minimal request looks like this:

REGISTER sip:pbx.example.com SIP/2.0
Via: SIP/2.0/UDP 192.0.2.10:5060;branch=z9hG4bK776asdhds
Max-Forwards: 70
From: <sip:1001@pbx.example.com>;tag=49583
To: <sip:1001@pbx.example.com>
Call-ID: 843817637684230@192.0.2.10
CSeq: 1 REGISTER
Contact: <sip:1001@192.0.2.10:5060>
Expires: 3600
Content-Length: 0

If the server requires a password, it replies 401 Unauthorized with a digest challenge (RFC 3261, sections 21.4.2 and 22.2). The phone sends the same request again with an Authorization header and a higher CSeq. RFC 3261 requires that increment (section 22.2). When the server accepts it, the reply is 200 OK. The registration lasts until the Expires time runs out, so the phone registers again before then. When registration fails, the response code tells you why, and the SIP response codes guide lists the common ones.

Hosted PBX vs on-premise PBX

The choice comes down to who runs the server and who gets the call when it breaks.

QuestionOn-premise PBXHosted PBX
Who runs the serverYour team or a contractorThe provider
Upfront costServer or appliance, plus setup timeUsually little or no hardware
Updates and security patchesYour jobThe provider's job, as the contract says
CustomizationMost control over call flows and integrationsLimited to what the provider exposes
What failsYour hardware, power and backupsThe provider's data center and your internet link
Good fitTeams with IT skills and special routing needsTeams that want less administration

A hosted PBX depends on your internet link. An on-premise PBX depends on it only when it reaches the phone network over a SIP trunk. The main difference is who is on call when something breaks, and how much of the stack you have to run.

SIP trunk or hosted PBX: which one are you buying?

A SIP trunk is a connection with a set of numbers. A hosted PBX is the phone system. Some providers sell both, and some sell only one. Read the quote carefully: a trunk usually does not include extensions, voicemail or a call queue, and a hosted PBX may include those features but charge per user.

Cost drivers to compare

Compare the rate card, not the headline number. These are the lines that usually move the monthly total:

  • Numbers. Providers often charge a monthly fee for each DID and a one-off fee to port an existing number. Porting lead times and fees depend on the carrier and the country.
  • Outbound minutes by destination. Local, mobile, long distance and international calls are priced differently. Price your real mix from last month's call log, not from an average.
  • Concurrent calls. The maximum number of calls that can run at once, sometimes called channels or call paths. Size it for your busiest minute, plus a margin.
  • Seats or extensions. Hosted plans often bill per user, so the cost grows with headcount.
  • Add-ons. Call recording, analytics, voicemail transcription and integrations are often separate lines.
  • Toll-free numbers. Check their per-minute charges separately from local numbers.

Concurrency is easy to get wrong. As a hypothetical, take an office of eight people where the busiest moment is four calls at once. Four concurrent paths is the floor, and a margin helps when calls overlap in ways you did not plan for.

Bandwidth for G.711 is easy to estimate. RFC 3551 defines PCMU (payload type 0) and PCMA (payload type 8), both with an 8,000 Hz clock and one channel, and it says each sample is 8 bits (section 4.5.14, Table 4). That is 64 kbps of audio. At the default 20 ms packet interval, each packet carries 160 bytes of audio, so there are 50 packets per second. A typical 20-byte IPv4 header (RFC 791), an 8-byte UDP header (RFC 768) and the 12-byte RTP fixed header (RFC 3550, section 5.1) add 40 bytes per packet. At 50 packets per second that is another 16 kbps, so each direction needs about 80 kbps before Ethernet framing. Four simultaneous G.711 calls need roughly 320 kbps each way, before SIP signaling and link-layer overhead.

When a softphone replaces desk phones

A softphone works well when the person has a reliable laptop or phone and a decent headset. Typical cases are remote staff, field sales, managers who travel, part-time staff, and a pilot before you buy hardware. Desk phones still make more sense at a busy reception desk, in a noisy room, or on a network that drops packets. Our Zoiper alternatives roundup compares the apps by platform.

As of October 2026, Darwvin Softphone is in Early Access for Android and as a Linux desktop app. Windows and macOS desktop apps are in development, and there is no iOS app.

Before you switch, check four things:

  1. The headset. The other side hears your microphone first, so the headset matters as much as the network.
  2. Upload bandwidth and packet loss. Test on the path your staff will use, during a busy hour. A quiet morning proves little.
  3. Background behavior on phones. Phone operating systems can suspend background apps. Check how the app rings when the screen is off.
  4. Emergency calls and location. Before you rely on any softphone for emergency calls, confirm how your provider routes them and what location it sends.

A checklist for choosing

  1. Count peak simultaneous calls. Buy concurrent paths for that number, plus a margin.
  2. Decide who runs the PBX. Ask who applies patches, who answers an outage at night and what the service agreement promises.
  3. Get the SIP details in writing. Server name or IP address, transport (UDP, TCP or TLS), port, codec list, and whether the provider uses registration or IP authentication. SIPconnect 2.0 requires TCP on both the PBX and provider sides, and allows UDP to accommodate legacy devices (section 8).
  4. Check the codecs. SIPconnect 2.0 requires any media endpoint that originates or terminates voice traffic to support G.711 mu-law and A-law at 20 ms (section 14.3), so G.711 is the baseline. Ask whether G.722 or Opus is also offered, and read the codec comparison before you decide.
  5. Test behind your real router. NAT and firewall settings decide whether registration and audio work. The SIP and NAT guide covers the cases.
  6. Check number porting. Look at lead time, fees, and whether the current carrier releases numbers on schedule.
  7. Check emergency calling. Confirm routing and location for every device type you will use.
  8. Price with your real call mix. Use last month's call log together with the rate card.
  9. Pilot before you retire desk phones. Run two weeks with a few softphone users, and keep the old phones connected until the pilot is clean.
  10. Check the CRM link. If calls should land on customer records, ask how the phone system integrates with your CRM before you sign.

If a test call fails after setup, start with the one-way audio guide before you change the PBX. For account setup, the Darwvin Softphone setup guide is the place to start, and the Free plan includes two SIP accounts.

FAQ

Is a SIP trunk the same as a hosted PBX?

No. A SIP trunk carries calls between your phone system and the phone network, while a hosted PBX is the phone system itself, run by a provider. Some providers sell both, so compare the parts you will actually receive.

Do I need an IP PBX if I buy a SIP trunk?

Something on your side has to speak SIP. That can be an IP PBX, a SIP gateway or, with some providers, a softphone or desk phone that registers directly. Ask the provider which model it supports, because registration and IP authentication settings differ.

How many simultaneous calls do I need?

Count the people who might be on a call at the same moment, then add a margin for overlap. Calls above your limit can be rejected with a SIP error response such as 486 Busy Here or 503 Service Unavailable (RFC 3261, sections 21.4.24 and 21.5.4), depending on the provider. The response codes guide explains the common ones.

Can a softphone replace desk phones for a whole office?

It can, if the headsets are good, the network is sound and you have checked emergency calling. High-volume reception desks usually work better with desk phones.

Does moving to a hosted PBX mean losing my phone numbers?

Not by itself. Many numbers can be ported to a new provider, but the process depends on the country and the current carrier, and it takes time. Confirm the porting plan before you cancel the old service.

References

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