How Much Bandwidth Do I Need for VoIP?

If you are moving to internet calling, our business VoIP service and the connectivity behind it have to be planned together — one without the other is where call quality goes wrong.

How much bandwidth does one VoIP call use?

One VoIP call typically consumes 80–100 kbps per direction when using standard business codecs. That figure is small, but it must be sustained and consistent — a call is real-time, so a momentary dip that would be invisible during a file download becomes audible as choppy or dropped audio.

The exact figure depends on the audio codec your phone system uses:

Most UK business VoIP platforms default to G.711 or G.722. Microsoft Teams, by comparison, runs audio at roughly 30–130 kbps depending on conditions, per Microsoft's published network requirements. The 100 kbps-per-call planning figure is a safe upper bound for capacity sizing.

How do I calculate my business VoIP bandwidth?

Multiply your number of simultaneous calls by 100 kbps. That gives the voice-only requirement — then add generous headroom for everything else your connection carries: email, Microsoft 365, cloud apps, file transfers and video.

A simple sizing table:

As a rule of thumb, allocate at least double your voice requirement as headroom for other business traffic — preferably more. A 20-person office rarely has all 20 lines active at once, so size to realistic peak concurrency, not headcount.

Why do latency and jitter matter more than speed?

For VoIP, consistency beats raw bandwidth. Two parameters decide whether a call sounds natural: latency (the delay before the other person hears you) and jitter (variation in how packets arrive). Get these wrong and even a gigabit line will deliver poor calls.

Latency

Latency is the one-way delay between speaking and being heard. Below 150 ms it is generally imperceptible; above 300 ms conversation becomes awkward, with people talking over each other. Most UK business broadband connections reach a VoIP provider's servers in 5–20 ms — comfortably inside the acceptable range.

Jitter

Jitter is variation in packet arrival times. Voice packets should arrive at steady intervals; irregular timing produces robotic or stuttering audio. A jitter buffer absorbs minor variation, but jitter above roughly 30 ms degrades calls noticeably. It is most common on congested or poorly provisioned connections — which is why the connection type matters.

Which internet connection is best for VoIP?

For most UK SMEs, full-fibre (FTTP) is the practical sweet spot for VoIP, while a leased line is the right call where voice is genuinely business-critical or spread across multiple sites. FTTC copper connections still work for light use but are being retired as full fibre expands.

FTTP (full fibre)

FTTP runs fibre all the way to your premises, with download speeds typically 100 Mbps–1 Gbps and low, consistent latency. For most SMEs running up to 20 concurrent calls, FTTP is more than adequate provided other data traffic is managed. Business FTTP typically starts from £35 per month. See our business broadband options for availability.

Leased line

A leased line gives you a dedicated, uncontended connection with guaranteed symmetrical bandwidth and SLA-backed latency and uptime. Where call quality is business-critical, or VoIP spans several sites, it is the connection we recommend — there is no contention to fight for your packets. Leased lines start at around £69/month for a 100 Mbps connection. Our dedicated internet access page explains how the SLA works.

FTTC (fibre to the cabinet)

FTTC uses fibre to the street cabinet and copper for the final stretch, so performance falls with distance. Upload speeds of 10–20 Mbps suit modest VoIP use but get tight beyond 15 concurrent calls. Openreach is retiring copper-based services as full fibre rolls out — see Openreach on the network upgrade. The UK's wider full-fibre progress is tracked by Ofcom.

Do I need QoS for VoIP?

Yes, on any shared connection. Quality of Service (QoS) is a router or switch setting that prioritises voice packets over less time-sensitive traffic, so a large upload or video call cannot starve your calls of capacity. It is the single most effective configuration for protecting call quality on a mixed-use line.

Most business-grade routers and managed switches support QoS. With AMVIA, it is configured as part of provisioning — one provider handling both the connection and the phone system means voice priority is set correctly from day one, not bolted on later.

How do I test my connection before migrating to VoIP?

Test with a VoIP-specific tool that measures latency, jitter and packet loss — not just a speed test. A connection well-suited to VoIP shows latency below 150 ms, jitter below 30 ms and packet loss under 1%. Packet loss above 1% will audibly degrade calls; fix the underlying connectivity before you deploy.

Many VoIP providers offer their own test utility. If a test reveals problems, the answer is usually QoS configuration, a more stable connection type, or both. Security matters too — voice runs over the same network as your data, so review our VoIP security guidance before go-live.

What AMVIA recommends

Size voice at 100 kbps per concurrent call, double it for headroom, insist on QoS, and choose the connection that matches how critical calls are to your business. For most SMEs that means business-grade FTTP; for multi-site or call-heavy operations it means a leased line. Pair the right line with a properly configured hosted phone system and call quality stops being a worry.

The advantage of one accountable provider is simple: when AMVIA supplies both the connection and the phones, there is no finger-pointing between a broadband company and a phone vendor when something sounds wrong. One provider. Security-first. Microsoft-certified.