How VoIP works technically
During a VoIP call, the microphone captures your voice and a codec (software for encoding audio) converts it into digital data packets. These travel across the network using the IP protocol: the same route as other internet traffic. On the recipient's side, the codec decodes the packets back into an audio signal.
The entire transmission depends on several layers:
- Code of Conduct - determines audio quality and bandwidth requirements; the standard G.711 codec requires approximately 80-100 kbps per call
- Signalling protocol - most commonly SIP (Session Initiation Protocol), which establishes, maintains and terminates the call
- Network infrastructure - router, switch and sufficient bandwidth with low latency (delay)
What VoIP is practically used for in a business
Companies are switching to VoIP for several reasons. Calls between branches or remote employees are free over the internet or cost a fraction of traditional tariffs. Numbers are not tied to a specific physical location: an employee can receive business calls from a laptop or mobile app (softphone) from anywhere.
VoIP also opens the door to functions that a traditional line cannot offer:
- call forwarding to a mobile phone or colleague
- voice mailbox with delivery of audio recording by email
- Call recording for quality control
- call queue and voice menu (IVR)
- CRM system integration: customer card opens upon incoming call
What to watch out for when deploying VoIP
Call quality depends on the network environment. Three parameters are key:
- Latency - packet delay; ideally under 150 ms one-way
- Jitter - latency fluctuations; with high jitter, you hear voice choppiness
- Packet loss - even a small percentage causes sound dropouts
To ensure quality, it is advisable to configure QoS (Quality of Service, i.e., traffic prioritisation) on the router so that voice data takes precedence over other traffic, such as downloading updates. Corporate Wi-Fi should be configured to prioritise voice traffic, ideally in the 5 GHz band with low interference.
A cloud-based virtual PBX, such as VirtualniUstredna.cz, hides this complex technology behind a simple web interface. You can configure SIP trunks, numbers and call routing rules without any telecommunications training.
VoIP versus traditional telephone line
A classic analogue line or ISDN operates on a circuit-switching principle, for the duration of a call, the entire communication channel is dedicated to you. VoIP, by contrast, shares network capacity with other traffic, which is cheaper but requires correct configuration.
ISDN networks in the Czech Republic are being phased out as operators switch to IP infrastructure. VoIP is therefore no longer just an economical choice. It is becoming the standard.
Summary of differences
- Acquisition costs: VoIP significantly lower: does not require physical PBXs or special cabling
- Scalability: VoIP: adding a line means clicking in the admin panel, not calling a technician.
- Reliability: dependent on internet quality; backup mobile connection handles outages
Frequently Asked Questions
Do I need a special telephone for VoIP?
Not necessarily. If you have an IP phone with SIP support, simply connect it to the network and register it on the PBX. Alternatively, a softphone can be used: an application on a computer or mobile phone that functions as a telephone without any hardware.
Does VoIP work during an internet outage?
No, VoIP requires a working internet connection. Companies therefore set up call forwarding to a mobile number as a backup, in the event of an internet outage, the switchboard automatically redirects incoming calls to the responsible employee's mobile phone.
Is VoIP secure?
VoIP can be secured through encryption: the SRTP protocol encrypts transmitted voice, while TLS protects signalling. Modern cloud-based PBX systems enable encryption by default. The risks are similar to those of other internet services: weak passwords or an unsecured network pose a greater threat than the protocol itself.