How VoIP Works —
The Complete Explanation
VoIP converts your voice into digital data packets sent over the internet. But what does that actually mean for your business? This guide explains it clearly — from signal to screen, no jargon required.
How VoIP Works — Step by Step
When you make a VoIP call, your voice goes through a series of fast, invisible steps before reaching the person on the other end. Here’s exactly what happens:
Breaking Down Each Step
Step 1 — Analogue to digital conversion: Your microphone captures sound waves. An ADC (Analogue-to-Digital Converter) samples these waves thousands of times per second and converts them into binary data. Most VoIP systems sample at 8,000 or 16,000 times per second.
Step 2 — Compression via codec: Raw digital audio takes a lot of bandwidth. A codec compresses the audio data — typically by 10–20× — while preserving enough quality for clear speech. The codec choice directly affects both call quality and bandwidth usage.
Step 3 — Packetisation: The compressed audio is split into tiny data packets — typically 20ms of audio each. Each packet is labelled with source, destination, sequence number, and timestamp, allowing out-of-order packets to be correctly reassembled.
Step 4 — Transmission over IP network: Packets travel over your internet connection to your VoIP provider’s servers, then onwards to the recipient. Unlike traditional calls that reserve a dedicated circuit, VoIP packets share network infrastructure — making VoIP dramatically cheaper.
Step 5 — Reassembly and playback: At the receiving end, the VoIP software collects arriving packets, reorders them using sequence numbers, buffers them briefly to smooth out jitter, and feeds the audio stream to a DAC (Digital-to-Analogue Converter) for playback.
“The entire process — from your voice leaving your mouth to the recipient hearing it — takes less than 150 milliseconds when done properly. That’s faster than human perception of delay.”
Traditional Phone (PSTN) vs VoIP
The Public Switched Telephone Network (PSTN) — built since the 1870s — works completely differently from VoIP. Understanding the difference explains why VoIP is not just cheaper, but fundamentally more capable.
- Dedicated circuit reserved for entire call
- Expensive per-minute billing, especially international
- Physical infrastructure — copper wires, exchanges
- Adding lines requires physical installation
- Voice only — no video, chat, or SMS
- Reliable on good copper infrastructure
- Shares bandwidth — no dedicated circuit needed
- Up to 70% cheaper — especially international
- 100% software — runs on any internet connection
- Add users instantly — no engineer required
- Voice, video, chat, SMS, fax — one platform
- Works anywhere with internet — remote-ready
PSTN uses circuit switching — a dedicated continuous connection for the full call duration, whether you’re speaking or silent. VoIP uses packet switching — voice is broken into packets that share the network with millions of other packets, using bandwidth only when there is audio to send. This efficiency is why VoIP costs a fraction of traditional telephony.
VoIP Protocols and Codecs Explained
Behind every VoIP call are two sets of standards working together: signalling protocols that manage the call, and codecs that handle the audio itself.
Key VoIP Protocols
Common VoIP Codecs
| Codec | Bandwidth | Quality | Best For |
|---|---|---|---|
| G.711 | 64 kbps | Excellent — HD-clear | Offices with fast broadband |
| G.729 | 8 kbps | Very good | Low-bandwidth connections |
| Opus | 6–510 kbps | Excellent — adaptive | WebRTC, variable networks |
| G.722 | 64 kbps | Wideband HD | HD voice calls |
| G.726 | 16–40 kbps | Good | Legacy systems |
VoIP Office automatically selects the best codec for your connection — defaulting to G.711 or G.722 for HD quality on strong connections, and switching to G.729 or Opus when bandwidth is limited. No manual configuration needed.
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What Affects VoIP Call Quality?
VoIP call quality is determined by four network factors. Understanding them tells you what to look for when choosing a provider or troubleshooting issues.
MOS Score — How Call Quality Is Measured
The industry standard for VoIP audio quality is the Mean Opinion Score (MOS) — a scale from 1 (bad) to 5 (excellent):
| MOS Score | Quality | User Experience | Typical Scenario |
|---|---|---|---|
| 4.3 – 5.0 | Excellent | Indistinguishable from face-to-face | G.711/G.722 on fast broadband |
| 4.0 – 4.3 | Good | Noticeable but comfortable | G.729 on standard broadband |
| 3.6 – 4.0 | Fair | Acceptable for business calls | Moderate packet loss or jitter |
| 3.1 – 3.6 | Poor | Difficult — repetition required | High jitter or packet loss >3% |
| < 3.1 | Bad | Unusable for business | Severe network problems |
VoIP Office targets a MOS score of 4.3+ on all calls using adaptive codec selection, global server infrastructure to minimise routing hops, and real-time quality monitoring via RTCP.
Types of VoIP Systems
Not all VoIP deployments are the same. There are four main types, each suited to different business sizes and requirements:
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Hosted VoIP / Cloud PBX — The provider hosts and manages all infrastructure. Your business uses the service via apps and a web dashboard. No hardware, no IT team needed. This is what most small and medium businesses use — and what VoIP Office provides.
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On-Premises IP PBX — The business owns and operates VoIP hardware on-site. Offers maximum control but requires significant upfront investment and dedicated IT expertise to maintain.
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SIP Trunking — Connects an existing on-premises PBX to the internet via SIP trunks, replacing physical ISDN lines. Businesses keep their existing phone system but get VoIP cost benefits.
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Hybrid VoIP — A combination of on-premises hardware and cloud services. Common in businesses mid-migration from traditional PBX to full cloud, or those with specific on-site data requirements.
- 1–200 employees: Hosted VoIP / Cloud PBX — lowest cost, fastest setup, no IT overhead.
- 200–1000 with existing PBX: SIP Trunking to extend your current system while cutting costs.
- 1000+ with dedicated IT team: On-premises or hybrid, depending on compliance requirements.
Ready to move your business to VoIP?
Our team will recommend the right deployment type for your size, industry, and existing setup.
Why VoIP Matters for Business
Understanding how VoIP works is useful — but what matters to a business owner is what it delivers in practice:
| VoIP Capability | Business Outcome |
|---|---|
| Packet-switched calls | 50–70% lower call bills vs PSTN — typically pays back within months |
| Software-based system | Add or remove users in minutes — no engineer, no hardware, no waiting |
| Any device, any location | Full remote and hybrid working — teams across offices or countries on one system |
| CRM integration via API | Customer records auto-pop on inbound calls — agents save 30–60 sec per call |
| Call recording & transcription | Compliance, dispute resolution, and training — no additional hardware |
| Real-time analytics | Call volumes, response times, missed calls — data-driven team management |
| Omnichannel messaging | Voice + video + chat + SMS + WhatsApp — one platform, no app-switching |
Frequently Asked Questions
No. A landline uses a dedicated copper wire circuit switched through telephone exchanges. VoIP sends your voice as digital data packets over any internet connection. To a caller you sound identical, but VoIP is far cheaper, more flexible, and far more feature-rich.
Yes — VoIP works on any smartphone via a dedicated app (iOS or Android). On WiFi it behaves identically to a desk phone. On 4G/5G it still works well — codecs like Opus are specifically designed for variable mobile network conditions. VoIP Office’s mobile app gives your personal phone your business number, without exposing your personal number.
Enterprise-grade VoIP providers use SRTP (Secure Real-Time Transport Protocol) and TLS encryption to protect calls in transit. VoIP Office encrypts all voice and signalling traffic by default, making interception practically impossible without encryption keys.
If your internet fails, VoIP calls over that connection will drop. The solution is automatic failover: VoIP Office lets you configure call forwarding rules that trigger if your connection is unavailable, redirecting incoming calls to mobile numbers instantly. Many businesses also use a 4G backup router for critical call traffic.
A standard G.711 call uses approximately 87 kbps including headers. G.729 uses around 31 kbps. A 100 Mbps office connection can theoretically support over 1,000 simultaneous calls — a team of 50 with everyone on a call simultaneously needs roughly 4–5 Mbps, a tiny fraction of typical office bandwidth.
Yes — this is a core feature of business VoIP. Your provider connects to the PSTN via internet gateways, so you can call any mobile or landline number worldwide. The call quality is identical to a landline for the recipient. You can also receive calls from landlines and mobiles on your VoIP number.
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