What internet speed does a VoIP contact centre need?
A practical guide to sizing bandwidth for a VoIP contact centre: per-call maths, the overhead everyone forgets, and whether fibre or wireless should carry your calls.
What internet speed do you need for a VoIP contact centre?
Budget about 100 kbps up and down per concurrent call on the G.711 codec, multiply by your peak simultaneous agents, then add roughly 30% headroom for signalling and other traffic. A 20-seat centre needs around 2.5 Mbps dedicated to voice alone, on top of everything else your network carries.
The four numbers that decide call quality
Raw download speed is the least important figure. Voice quality is decided by four metrics, and only one of them is “speed”.
Bandwidth is how much data the link can carry. Latency is the delay a packet takes to arrive, and you want one-way latency under 150 ms. Jitter is the variation in that delay; high jitter makes voice choppy. Packet loss is data that never arrives, and above about 1% you hear it as robotic or dropped audio. A fast line with high jitter still delivers bad calls.
How much bandwidth one call actually uses
It depends on the codec, the compression method your platform uses to turn voice into data.
| Codec | Quality | Bandwidth per call |
|---|---|---|
| G.711 (uncompressed) | High, PSTN quality | ~100 kbps |
| G.729 (compressed) | Good | ~30–40 kbps |
| Opus (dynamic) | Excellent, adaptive | ~40–100 kbps |
Plan on the safe side with G.711 at 100 kbps per call, both directions. Then scale by concurrency: it is your peak simultaneous calls that sets the requirement, not your total headcount.
Sizing the link for a contact centre
The traffic everyone forgets to count
Voice is never alone on the wire. The most common sizing mistake is budgeting for calls and ignoring everything else.
Your agents’ calls share the link with CRM lookups, cloud dashboards, file uploads, video meetings and often CCTV streams. If you size the connection for voice only, those workloads eat into call quality the moment the centre gets busy. Count total usage across departments, then give voice its own protected lane.
Fibre or wireless for a contact centre?
For a centre of any size, fibre should be the primary link and wireless the backup, not the other way around.
| Requirement | Fibre | Fixed wireless |
|---|---|---|
| Consistency under load | Most stable at scale | Good, tower-dependent |
| 10+ agents | Comfortable headroom | Workable, watch contention |
| Remote / no-fibre sites | Only where built | Reaches where cable can’t |
| Role in the design | Primary link | Ideal failover path |
Wireless is a strong backup and the right primary choice in remote areas, but for a growing centre the reliability of fibre carries the core load, with wireless on standby for failover.
Optimising the network for voice
The right speed on paper still delivers bad calls without the right configuration underneath it.
A contact centre doesn’t fail on download speed. It fails on jitter and packet loss at the busiest hour – the exact moment no speed test is watching.
WhichVoIP editorial view
Our verdict
Size a VoIP contact centre from peak concurrent calls, not headcount: 100 kbps per call on G.711, times your busiest simultaneous total, plus about 30% overhead, and then add the rest of your business traffic on top. Make fibre the primary link, keep wireless as failover, prioritise voice with QoS on a dedicated VLAN, and test at peak. Do that and the connection stops being the thing that limits your customer service.
Sizing the connection for your contact centre?
Tell us your agent count and location and we’ll help you compare fibre and wireless options built for voice.
Frequently asked questions
How much internet speed does a VoIP contact centre need?
How much bandwidth does one VoIP call use?
How do I calculate bandwidth for 20 agents?
Should a contact centre use fibre or wireless?
What latency and packet loss are acceptable for VoIP?
Why do my calls drop even though my speed test looks fine?
Keep reading
Sources: ITU-T G.711 and G.729 codec references; Opus codec (IETF RFC 6716); ITU-T G.114 one-way latency guidance (under 150 ms); standard VoIP network engineering practice (QoS, VLAN isolation, peak-concurrency sizing). Verified 1 July 2026.