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Glossary · VoIP and call quality

VoIP

In one sentence

VoIP (voice over internet protocol) carries phone calls as packets of data over an IP network, such as the internet or an office LAN, instead of over the copper lines of the traditional phone network.

Also known as : voice over IP · voice over internet protocol · IP telephony · internet telephony

How your voice travels in a VoIP callA four-step flow. 1: the voice is picked up by the microphone of the IP phone or softphone. 2: the codec digitises and compresses it, for example with G.711 or Opus. 3: it travels as small IP packets, around fifty a second, across the network. 4: at the other end, the packets are put back in order and turned back into sound.1 · Voice picked upIP phone orsoftphone microphone2 · CodecDigitises and compresses(G.711, G.722, Opus…)3 · IP packetsSent every 20 ms overthe internet or LAN4 · PlaybackPackets reordered andturned back into sound
VoIP call quality depends on this journey: packets that arrive late or go missing are what you hear as breaks.

What is VoIP?

VoIP, voice over internet protocol, carries your voice the same way the network carries any other data: in small packets over an IP network. For a century a phone call held an electrical circuit open end to end; with VoIP, a conversation shares the same cable as email and web browsing.

In the UK this is no longer optional. Openreach is withdrawing the traditional PSTN and ISDN network, and every remaining analogue or ISDN line has to move to an IP-based service. Ireland’s operators are on a similar path as copper is retired.

How it works

A VoIP call is made up of two streams:

  • Signalling, which sets up, changes and ends the call. In business telephony this is almost always SIP.
  • Media, the voice itself. A codec digitises and compresses it, and it travels as RTP packets, typically one every 20 milliseconds.

At the far end a buffer puts packets back in order and plays them out. On a busy network, packets arrive late or go missing: that is what latency, jitter and packet loss measure. On an office network, QoS settings give voice priority over downloads and backups.

To reach ordinary landlines and mobiles, the VoIP provider interconnects with the public network, so you keep geographic numbers such as 020 or 0161 and nobody calling you notices any difference.

Example

A family-run hotel in the Lake District receives notice that its analogue lines will be migrated as part of the switch-off. Its old PBX also runs the room phones. The hotel keeps its 015394 number by porting it, moves reception and back-office staff to a cloud phone system, and connects the room phones through an analogue adapter while it budgets for replacements. Calls between rooms and reception now run over the hotel’s own network, and when the manager is away the main number follows her to the mobile app instead of ringing an empty desk.

VoIP at Universal Telecom

All our voice services run on VoIP: the virtual PBX for businesses that want no equipment, the SIP trunk for those keeping their own PBX, and global numbering for local numbers in other countries.

Frequently asked questions

How much bandwidth does VoIP need?

Roughly 30 to 100 kbps per call in each direction, depending on the codec. Ten simultaneous calls fit within 1 Mbps; a stable, uncongested connection matters far more than headline speed.

What happens to VoIP if the internet goes down?

Office handsets can no longer reach the provider, but a hosted phone system keeps receiving calls and can divert them automatically to mobiles until the connection returns.

Does the UK PSTN switch-off mean I need VoIP?

Yes. Openreach is retiring the analogue PSTN and ISDN network, with completion currently planned for January 2027. Remaining lines need to move to an IP-based service such as a SIP trunk or cloud phone system.

Related terms

  • SIP

    SIP (Session Initiation Protocol) is the standard signalling protocol that sets up, changes and ends VoIP calls. It handles ringing, answering and hanging up; the voice itself travels separately.

  • SIP trunk

    A SIP trunk is a voice-over-IP connection between a business phone system and a carrier. It replaces ISDN lines and carries calls and numbers over your internet connection.

  • Codec

    A codec is the algorithm that digitises and compresses your voice during a VoIP call and turns it back into sound at the other end. It sets both the audio quality and the bandwidth used.

  • QoS (quality of service)

    QoS (quality of service) is the set of network settings that give time-sensitive traffic, such as voice, priority over other data, so calls avoid delay, jitter and packet loss when the link is busy.

  • Cloud PBX

    A cloud PBX is a business phone system run by a provider in its data centres. Extensions, IVR menus, queues and call routing are managed from a web portal, with no PBX hardware on site.

  • Latency

    Latency is the time your voice takes to travel from your mouth to the other person's ear. In VoIP, once one-way delay passes about 150 ms, conversations start to feel stilted.

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