A speed test can show 500 Mbps and your calls can still sound like a bad radio. That's because call quality depends far less on speed than on how steadily your connection delivers small packets of sound. Three numbers decide it: latency, jitter and packet loss.
This guide explains each one in plain words, gives the targets published by the ITU (the international telecom standards body), Microsoft and Twilio, shows how to test your own connection with free tools, and lists the fixes in the order worth trying. For a symptom-by-symptom list, see common VoIP problems.
The three numbers that decide call quality
- Latency: How long a packet takes to get from one end to the other. Too much and people talk over each other. Tests usually report round-trip time (there and back), which is roughly twice the one-way delay.
- Jitter: How uneven the arrival times are. Packets are sent every 20 ms or so; if they arrive at 5 ms, then 60 ms, then 15 ms, the app has to wait for stragglers. That waiting area is called a jitter buffer. A small buffer drops late packets; a big one adds delay.
- Packet loss: The share of packets that never arrive. A little is hidden by the codec; more causes clipped words and robotic audio. Loss that comes in bursts is worse than the same amount spread out.
Bandwidth matters only once it runs out. A voice call is small, but if something else fills your upload, every packet waits in line behind it, and latency, jitter and loss all rise together. That effect has a name, bufferbloat, and it's one of the most common causes of bad calls on otherwise fast connections.
What good looks like: published targets
Different organizations measure different parts of the path, so their numbers aren't identical. Here's what each one actually says.
| Source | What it measures | Target |
|---|---|---|
| ITU-T Recommendation G.114, One-way transmission time | One-way, mouth-to-ear delay | Below 150 ms: most applications feel essentially instant. Above 400 ms: not acceptable for general network planning. |
| Microsoft's Teams network requirements | From your device to Microsoft's network, including Wi-Fi | One-way latency under 50 ms (round trip under 100 ms); packet loss under 1% in any 15 seconds; jitter under 30 ms in any 15 seconds. |
| Microsoft's Teams network requirements | From your office's internet edge to Microsoft's network | One-way latency under 30 ms; packet loss under 0.1%; jitter under 15 ms. |
| Twilio's Voice SDK network connectivity requirements | From your device to Twilio, for browser and app calls | Round-trip latency under 200 ms; jitter under 30 ms; packet loss under 3%. |
A practical reading for a small business: aim for jitter under 30 ms and packet loss under 1% from your device, and keep one-way delay well under 150 ms. If you're over those numbers, callers will notice. If you're comfortably under and calls are still bad, look at the device, headset or the other end of the call.
How much bandwidth a call needs
Less than most people think, which is why buying more speed rarely fixes jitter.
| Codec | Bandwidth per call, each way | Source |
|---|---|---|
| Opus (common in browser and app calls) | 40 kbps | Twilio's Voice SDK network connectivity requirements |
| G.711 / PCMU (classic phone-quality audio) | 100 kbps | Twilio's Voice SDK network connectivity requirements |
| G.711 at 20 ms packets, on Ethernet, with all headers | 87.2 kbps | Cisco's per-call bandwidth guide |
| G.729 (compressed) at 20 ms packets, on Ethernet | 31.2 kbps | Cisco's per-call bandwidth guide |
Worked example: an office with five people on G.711 calls at the same time needs about 5 × 100 kbps = 500 kbps up and 500 kbps down, using Twilio's figure. That's half a megabit. The calls fail not because the line is too slow but because a cloud backup or a video upload fills the upload at the same moment. Upload is usually the side that runs out, since home and small-business plans often have far less upload than download.
How to test your connection (free tools)
Run tests the way you make calls: from the same device, on the same network, at the time of day problems happen. Run each test three times, and do it once on Wi-Fi and once on a cable. All tools below were checked October 19, 2026.
- Cloudflare Speed Test: Runs in the browser and reports download, upload, latency, jitter and packet loss, with a rating for video calls. A good first test.
- Twilio Network Test: Twilio points to it from its own connectivity requirements as the way to test your connection for browser and app calls. It runs in the browser.
- Waveform Bufferbloat Test: Measures how much your latency rises while the connection is busy downloading and uploading, and gives a letter grade. A big jump under load is bufferbloat.
- M-Lab Speed Test: A speed test from Measurement Lab, an open internet measurement project. Useful as a second opinion on speed and latency.
- ping and traceroute: Built into every computer. In Terminal on a Mac, ping -c 100 1.1.1.1 sends 100 packets and prints the loss and the min, average and max times; on Windows use ping -n 100 1.1.1.1. traceroute (Mac) or tracert and pathping (Windows) show where along the route delay or loss starts.
- Your provider's call stats: Most VoIP services show per-call quality (jitter, packet loss, sometimes a quality score) in the admin. Look at the bad calls, not the average.
Microsoft also offers a Teams Network Assessment Tool for testing an office's internet edge against its targets, which IT teams may already use.
Reading the results
| What you see | What it usually means | Where to look |
|---|---|---|
| Fine on a cable, bad on Wi-Fi | The Wi-Fi is the problem | Band, distance, interference, access point placement |
| Fine when idle, bad when loaded | Bufferbloat: a full upload or download queue | Uploads, backups, QoS or smart queue management on the router |
| High latency even when idle | A long route, a VPN or satellite internet | VPN settings, your internet provider |
| Packet loss starting at the first hop past your router | Trouble on the internet provider's line | Your internet provider, with traceroute output |
| Everything clean, calls still bad | The device, headset or the other end | Swap the headset; call other numbers; ask your provider |
Mean opinion score (MOS)
Many phone systems summarize quality as a mean opinion score. The scale comes from ITU-T Recommendation P.800 (checked October 19, 2026), which describes how to have listeners rate calls from 1 (bad) to 5 (excellent). Today most systems estimate the score from latency, jitter, loss and codec instead of asking people. Twilio's Voice Insights call summary docs (checked October 19, 2026) say a score of 4.2 or higher is generally considered a good call, and flag calls whose score falls below 3.5. Use MOS to spot bad calls, then look at the three underlying numbers to find out why.
Fixes in order
- 1. Use a cable for desk phones and the computer you call from. It removes Wi-Fi from the list of suspects in one step.
- 2. Stop the competition. Schedule cloud backups and big uploads outside business hours; pause them when calls matter.
- 3. Turn on smart queue management or QoS on the router, with voice at the highest priority. This is the most effective fix for bufferbloat.
- 4. Fix the Wi-Fi. Use 5 GHz, put access points closer, use channels that don't overlap, and turn on WMM, per Microsoft's guide to preparing a network for Teams.
- 5. Keep calls out of the VPN (split tunneling).
- 6. Update or replace the router. Old consumer routers handle queues badly.
- 7. Escalate to your internet provider with traceroute and test results if loss starts on their side.
- 8. Add upload capacity or a second line only if tests show the upload is full during business hours.
What Chargezen does instead
Chargezen isn't a VoIP provider, and it doesn't tune networks. What it changes is who answers. Concierge answers calls on your business number with AI, so a customer still gets an answer when the person who usually picks up is on a weak connection, on another call or off for the day.
- Concierge answers on your number: Concierge answers calls and texts on your US business number with AI, day and night, from the instructions and details you give it. See the AI receptionist and the phone channel.
- Teammates join from Finn or the browser: Listen in with your microphone off, tap Join the call, and Concierge tells the caller you're there. From the Finn app on iPhone or Mac you can transfer the call to a teammate with a note. See phone calls and the join and transfer guide.
- One thread per customer: Everything said on the call, by Concierge and by your team, is saved in the customer's conversation with their texts, so a callback starts where the last call left off.
When a teammate joins from the browser, their audio travels over their own internet connection, so the targets and fixes above apply to them. Joining on a cable with a wired headset gives the best result.
To be plain about it: Chargezen doesn't sell VoIP lines, desk phones or a phone system (PBX). It has no phone menu, ring groups, call forwarding or simultaneous ring, and its numbers are US (+1) only. If you need extensions on desk phones, a VoIP provider is still the right tool.
Concierge Starter is $20 a month with 20,000 credits. Pro is $199 a month with 199,000 credits, or $1,990 a year. Phone and text charges are billed as used, with no per-user fees. See pricing.
Related terms: VoIP, AI receptionist and voice AI agent. Related guides: common VoIP problems, why your phone keeps dropping calls and IVR vs AI phone agent.
Frequently asked questions
What is a good jitter for VoIP?
Under 30 ms from your device is the target both Microsoft (for Teams) and Twilio (for its Voice SDKs) publish. Lower is better; a well-run wired network often shows single digits.
Is a 50 ms ping good for calls?
Yes. A 50 ms round trip is about 25 ms each way, far below the 150 ms one-way level in ITU-T G.114. Watch what happens to it while the connection is busy, though; that's where calls usually suffer.
Can jitter be on the other person's side?
Yes. If they're on poor Wi-Fi or a weak cell signal, you'll hear choppy audio even though your network is fine. Call another number to compare.
Is call quality the same thing as jitter?
Jitter is one part of it. Call quality is the combined effect of latency, jitter, packet loss, the codec and the devices at each end, which is what a mean opinion score tries to summarize.
