VoIP Bandwidth Test: How Much Internet Speed Do You Actually Need?

Switching to VoIP is one of the smartest moves a UK business can make — but if your internet connection cannot keep up, call quality will suffer. Choppy audio, delays, dropped calls, and echo are all symptoms of insufficient or poorly managed bandwidth.

This guide cuts through the vague advice and gives you real numbers. You will know exactly how much bandwidth your business needs for VoIP and how to test whether your current connection is up to the job.

VoIP Bandwidth Requirements: The Real Numbers

Every VoIP call uses a specific amount of bandwidth depending on the codec (compression method) used:

Bandwidth Per Call by Codec

• G.711 (uncompressed) — 87 Kbps per call (both directions). Best audio quality, most widely supported. This is the standard for business VoIP.
• G.729 (compressed) — 32 Kbps per call. Good quality with much lower bandwidth. Useful when bandwidth is limited.
• G.722 (wideband) — 87 Kbps per call. HD voice quality — noticeably clearer than G.711.
• Opus (adaptive) — 6-510 Kbps, adjusts dynamically. Increasingly used in modern platforms like Microsoft Teams and WebRTC-based systems.

These figures include packet overhead (IP, UDP, and RTP headers), which adds roughly 50% on top of the raw audio bitrate.

How Many Calls Can Your Connection Handle?

Using G.711 (the most common business codec) at approximately 100 Kbps per call with some safety margin:

• 10 Mbps connection — approximately 80-100 simultaneous calls (theoretical maximum, but never use more than 80% of your bandwidth for voice)
• 50 Mbps connection — approximately 400+ simultaneous calls
• 100 Mbps connection — bandwidth is not a concern for voice; focus on quality metrics instead

But raw bandwidth is only part of the story. A 100 Mbps connection with high packet loss will deliver worse VoIP quality than a 10 Mbps connection with clean, stable performance.

The Metrics That Actually Matter

Bandwidth gets all the attention, but these three metrics determine whether your VoIP calls will actually sound good:

Packet Loss

The percentage of data packets that do not arrive at their destination.

• Below 0.5% — excellent; no impact on call quality
• 0.5% to 1% — acceptable; minor quality reduction under heavy load
• 1% to 2.5% — noticeable; choppy audio, especially on longer calls
• Above 2.5% — calls will sound terrible or drop entirely

Jitter

The variation in packet arrival times. Voice packets need to arrive at consistent intervals. High jitter means some packets arrive late and get discarded.

• Below 15ms — excellent
• 15ms to 30ms — acceptable with a good jitter buffer
• Above 30ms — audio quality degrades noticeably

Latency

The time it takes for a packet to travel from your phone to the VoIP server and back.

• Below 80ms — excellent; conversations feel natural
• 80ms to 150ms — acceptable; slight delay noticeable in fast-paced conversation
• Above 150ms — people talk over each other; conversation becomes difficult

How to Run a VoIP Bandwidth Test

A standard speed test from Ookla or Fast.com tells you your raw bandwidth but nothing about packet loss, jitter, or latency. You need a VoIP-specific test.

Recommended Testing Tools

• Your VoIP provider test tool — many providers offer a test page tailored to their infrastructure. This is the most relevant test because it measures the actual path your calls will take.
• Ping and traceroute — run a continuous ping to your VoIP provider server IP for 5-10 minutes. Watch for packet loss and latency spikes. Traceroute shows you every hop between your network and the server.
• iPerf — a free tool for testing network throughput and quality between two points on your network. Useful for testing internal network segments.

How to Test Properly

1. Test during peak hours — run tests when your office is busiest (10am-12pm and 2pm-4pm are typical peaks). Testing at 6am on a Sunday tells you nothing about real-world performance.
2. Test under load — have your team working normally while you test. You need to see how VoIP performs when competing with everyday office traffic.
3. Test upload and download separately — VoIP is symmetrical; it needs the same quality in both directions. Many broadband connections have much lower upload than download, which creates a bottleneck.
4. Test over several days — performance varies. A single test is a snapshot; you need a pattern.
5. Test wired, not WiFi — test from a device connected directly to your network via Ethernet. WiFi introduces its own latency and packet loss that has nothing to do with your internet connection.

Calculating Your Business Requirements

Here is a practical formula for working out what you need:

Step 1: Count Your Concurrent Calls

Not every phone is in use at the same time. For most businesses:

• Small office (5-10 staff) — expect 3-5 concurrent calls at peak
• Medium office (20-50 staff) — expect 10-20 concurrent calls
• Call centre or sales team — potentially 1:1 ratio (every agent on a call simultaneously)

Step 2: Calculate Voice Bandwidth

Multiply concurrent calls by 100 Kbps (for G.711 with overhead):

• 5 concurrent calls = 500 Kbps (0.5 Mbps)
• 20 concurrent calls = 2 Mbps
• 50 concurrent calls = 5 Mbps

Step 3: Add Your Non-Voice Traffic

Factor in everything else your business does online:

• Email and web browsing — 1-2 Mbps per user
• Cloud applications (Microsoft 365, Google Workspace) — 2-5 Mbps per active user
• Video conferencing — 2-4 Mbps per active call (much more than voice alone)
• File transfers and backups — variable, can saturate a connection during large uploads

Step 4: Apply the 80% Rule

Never plan to use more than 80% of your total bandwidth. The remaining 20% provides headroom for traffic spikes and ensures QoS has room to work.

For businesses choosing between broadband types, our business broadband comparison guide helps you find the right speed and connection type for your needs.

What to Do If Your Connection Is Not Good Enough

Quick Wins

• Enable QoS on your router to prioritise voice traffic above everything else
• Schedule heavy traffic — run backups, updates, and large uploads outside business hours
• Use wired connections for VoIP phones — never rely on WiFi for desk phones
• Use a compressed codec — switching from G.711 to G.729 cuts bandwidth per call by 60%

Infrastructure Upgrades

• Upgrade to full fibre (FTTP) — dramatically better upload speeds and lower latency compared to FTTC
• Consider a leased line — dedicated, symmetric bandwidth with guaranteed performance and SLAs. Ideal for businesses with 20+ concurrent calls or where call quality is mission-critical.
• Add a dedicated voice line — some businesses run a separate internet connection exclusively for VoIP traffic, completely isolating it from data.
• Implement SD-WAN — software-defined networking that intelligently routes voice traffic over the best available connection in real time.

For a full rundown of VoIP solutions that include connection assessment and optimisation, see our hosted VoIP solutions guide.

Monitoring Bandwidth and Call Quality Over Time

A one-off test tells you how things are right now. For ongoing confidence in your VoIP quality:

• Use your VoIP provider call quality dashboard — most providers show MOS (Mean Opinion Score), jitter, and packet loss per call
• Set up network monitoring — tools like PRTG, Nagios, or even simple ping monitors can alert you to degradation before it affects calls
• Review monthly — bandwidth needs grow as you add staff, adopt new cloud tools, and increase video conferencing