In short
A VoIP phone system works by registering your phones and apps with a phone system (IP PBX), setting up each call with the SIP signalling protocol, and sending your voice as compressed RTP packets over the network. A SIP trunk, Microsoft Calling Plan, Operator Connect, Direct Routing or a Webex PSTN option connects calls to ordinary numbers.
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A VoIP call, step by step
Think of this as a diagram in words. When someone in your office calls a customer's mobile, this is what happens:
- Registration. When your desk phone, softphone or mobile app starts, it registers with the phone system over the network, which tells the phone system where to send calls for your extension.
- Dialling. You dial the number. Your device sends a SIP request to the phone system asking to set up a call.
- Routing. The phone system checks your rules: who you are, what you are allowed to dial, and which route to use for external calls.
- Hand-off to the phone network. The phone system passes the call to its connection to the public phone network (PSTN), such as a SIP trunk or a Microsoft or Cisco calling service.
- Ringing and answer. The customer's mobile rings. When they answer, SIP messages confirm the call and agree the codec both ends will use.
- Audio flows. Your voice is digitised, compressed by the codec, cut into small packets and sent as RTP (real-time transport protocol) packets. Their voice comes back the same way.
- Hang up. When either side hangs up, a final SIP message closes the call, and the phone system logs it for call history and reporting.
Incoming calls follow the same path in reverse: the call arrives from the phone network, the phone system applies your menu, hunt group or queue, and rings the right devices.
Codecs, packets and RTP
A codec is the method used to turn sound into data and back again. Common ones on business VoIP systems are G.711 (traditional telephone quality), G.722 (wideband or "HD" voice), G.729 (compressed, using less bandwidth) and Opus, which adapts its rate to the network. Microsoft lists SILK, G.711, G.722, G.729 and AMR-WB among the codecs supported for Teams Direct Routing.
The codec decides how much bandwidth each call uses. Cisco puts a G.711 or G.722 call at about 87.2 kbps in each direction and a G.729 call at about 31.2 kbps, including Ethernet overhead at 20 ms packetisation. Each packet carries a small slice of speech, so packets that arrive late, out of order or not at all are heard as delay, choppy audio or gaps. That is why a voice-friendly network matters, and why our VoIP call quality guide focuses on delay, jitter and packet loss.
Audio can be encrypted in transit (SRTP for media and TLS for signalling) where the platform and devices support it, which our VoIP security page covers.
SIP signalling and registration
SIP (session initiation protocol) is the language VoIP devices and systems use to set up, change and end calls. It does not carry your voice; it arranges the call, while RTP carries the audio. SIP messages handle registration, ringing, answering, holding, transferring and hanging up.
Registration is what makes VoIP so flexible. A device proves who it is to the phone system, and from then on calls for that extension go to wherever it registered from: a desk in the office, a laptop at home or a mobile on 4G or 5G. Several devices can register for the same user, so one number can ring all of them at once.
Because SIP is the signalling standard, a SIP phone from one maker can usually work with a phone system from another, if it is on that platform's supported list. Our SIP vs VoIP explainer goes further.
How does VoIP connect to the PSTN?
Your VoIP phone system talks to the public switched telephone network (PSTN) through a provider that interconnects with it. The provider converts between VoIP and the phone network, carries your numbers and passes on caller ID. The route depends on the platform:
| Route | Used with | How it works |
|---|---|---|
| SIP trunk | 3CX and other IP PBXs | An internet-based trunk from a carrier connects your phone system to the PSTN; capacity is set in channels or simultaneous calls. See SIP trunk providers |
| Microsoft Calling Plan | Teams Phone | Microsoft acts as the carrier and provides numbers and calls |
| Operator Connect | Teams Phone | A certified operator provides numbers and calling, enabled in the Teams admin centre |
| Teams Phone Mobile | Teams Phone | A mobile operator links a user's mobile number to Teams |
| Direct Routing | Teams Phone | A certified session border controller (SBC) connects Teams to a carrier or PBX; our Forever Teams service uses this route |
| Cisco Calling Plans | Webex Calling | Cisco provides PSTN service; the UK is listed |
| Cloud Connected PSTN | Webex Calling | A Cisco cloud-connected PSTN provider supplies calling |
| Local gateway | Webex Calling | A gateway or SBC on your premises or hosted by a partner connects to your own trunk |
Microsoft lets these Teams options be mixed in one tenant, and Webex Calling chooses the PSTN option per location. Hosted platforms such as Forever UC handle the PSTN connection for you as part of the service. For the platform detail, see Teams VoIP, 3CX VoIP and Webex VoIP.
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What the phone system does with an incoming call
The phone system is the brain of a VoIP setup. When a call arrives from the phone network, it works through your rules in order:
- Which number was dialled: the main number, a department number or a person's direct dial
- What time it is: opening hours, lunch, out of hours and holidays can each send calls somewhere different
- Where to send it: an auto attendant menu, a hunt group, a call queue, a single user or voicemail
- Who is available: users signed in on a desk phone, laptop or mobile, and whether they are already on a call
- What happens next: if nobody answers, the call can overflow to another group, a mobile or voicemail, and the voicemail can be emailed
All of this is set in software, so changing a menu or holiday message takes minutes rather than an engineer visit. VoIP phone system features covers the routing tools each platform offers, and multi-line VoIP phone systems explains how many calls can be handled at once.
VoIP system architecture: hosted, on-site and hybrid
The same building blocks can be arranged in three ways:
- Hosted: the phone system runs in the provider's data centre, devices register over the internet and the provider runs the PSTN connection. Forever UC, Webex Calling, Teams Phone and 3CX hosted by 3CX work this way
- Self-hosted or on-site: you run the phone system software, such as 3CX, on a server in your building or in your own cloud account, and connect it to the PSTN with a SIP trunk
- Hybrid: an existing phone system keeps running while users also call through Teams, for example with Dstny's Direct Routing service, which connects Teams users to an existing SIP-capable PBX or SIP trunk with nothing installed on site; each user needs Teams Phone Standard
PBX vs VoIP compares the phone system options in more depth, and Webex Calling, for example, runs from Cisco data centres that include London and Manchester.
Is a VoIP phone system the right fit for you?
We will check it against your systems and requirements and recommend the right option.
How to build a VoIP phone system for your business
Building one is less about technology and more about planning. In practice it means:
- Checking your connection and network against the VoIP requirements checklist.
- Choosing a platform that fits how your people work; see choosing the best VoIP phone system.
- Planning numbers, direct dials and porting your landline number to VoIP.
- Designing call flows: menus, groups, queues and opening hours.
- Choosing devices, from apps to VoIP desk phones for business.
- Installing, testing and training; see VoIP phone system installation.
We can do this with you. Contact us and we will start with a short requirements check.
How Forever Systems helps you choose a VoIP phone system
We supply and support several VoIP platforms, so we can recommend the one that fits how your team takes calls rather than fitting your business around one product.
- Platform advice across hosted 3CX, Microsoft Teams Phone, Webex Calling, Forever UC and Forever Teams, with SIP trunks for phone systems you already own
- A requirements check of your broadband, network and routers before anything goes live
- Numbers and porting: we can help you keep your existing business numbers and plan the move away from analogue and ISDN lines
- Handsets and headsets that suit each platform, bought outright or, in some cases, included in your monthly rental, with the options confirmed at quote
- Set-up and changes: call flows, users and devices configured for you, and help with moves, adds and changes afterwards
- For BT Cloud Voice, see our sister site cloudvoice.cloud, and for hosted and cloud platforms in depth, cloudphone.cloud
We work with businesses, schools, charities and public sector organisations.
Frequently asked questions
How does VoIP connect to the PSTN?
Through a provider that interconnects with the public phone network. A 3CX or other IP PBX uses a SIP trunk; Teams Phone uses a Microsoft Calling Plan, Operator Connect, Teams Phone Mobile or Direct Routing; Webex Calling uses Cisco Calling Plans, Cloud Connected PSTN or a local gateway. The provider converts between VoIP and the phone network.
Does VoIP use SIP?
Most business VoIP does. SIP sets up, changes and ends calls, while RTP carries the audio. Some platforms use their own protocols between their apps and servers but still use SIP to connect to carriers and other systems.
Can a VoIP phone system call mobiles and landlines?
Yes. Once the phone system is connected to the PSTN, users can call any UK landline or mobile, and international numbers if allowed, and anyone can call your numbers as normal.
What happens to VoIP calls if the internet goes down?
Devices at that site lose their link to the phone system until the connection returns. With a hosted system, incoming calls can be diverted to mobiles, another site or voicemail, and mobile apps can carry on over mobile data. A backup connection reduces the risk.
What is the difference between SIP and RTP?
SIP is the signalling protocol that sets up and ends calls. RTP carries the audio once the call is connected. Both can be encrypted, with TLS for SIP and SRTP for the media, where the platform supports it.
What is a codec in VoIP?
A codec turns speech into data and back again. Common VoIP codecs include G.711, G.722 (wideband), G.729 (compressed) and Opus, which adapts to the network. The codec affects call quality and how much bandwidth each call uses.
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