Every test on this site
Each test has its own page with a deeper measurement and a full FAQ. Start with the one that matches your symptom.
What a good result looks like
| Measurement | GOOD | FAIR | POOR | What you hear when it's bad |
|---|---|---|---|---|
| Latency (round trip to nearby server) | ≤ 100 ms | 101–200 ms | > 200 ms | Delay, people talking over each other |
| Jitter | ≤ 20 ms | 21–30 ms | > 30 ms | Choppy, robotic, "underwater" audio |
| Packet loss | 0% | ≤ 1% | > 1% | Clipped words, gaps, dropped calls |
| MOS (estimated) | ≥ 4.0 | 3.6–3.99 | < 3.6 | Overall "this call sounds bad" |
| Bufferbloat (latency increase under load) | < 30 ms | 30–99 ms | ≥ 100 ms | Calls fall apart when someone uploads or streams |
These are the pass/fail lines this site uses. They are built from ITU-T G.114 (keep one-way mouth-to-ear delay under 150 ms for most calls), the common carrier guidance of jitter under 30 ms and loss under 1%, and the E-model MOS scale. Your carrier may publish its own numbers — use theirs if they differ.
Match your symptom to the test
| What callers say | Most likely cause | Run this |
|---|---|---|
| "You're breaking up" / robotic voice | Jitter or packet loss | Jitter, Packet loss |
| "There's a delay" / talking over each other | High latency or bufferbloat | Latency, Bufferbloat |
| Only one person can hear | NAT, firewall, SIP ALG | IP, NAT & STUN |
| Fine until someone uploads or joins Zoom | Bufferbloat / no QoS | Bufferbloat |
| "You sound far away" / echo | Headset, speakerphone, mic level | Mic & headset |
| Phone shows "No service" / can't register | DNS, firewall, provisioning | DNS & SRV, Status lights |
| Cell calls drop inside the building | Weak RSRP / poor SINR | Cellular test |
Why a phone tech built this
Most speed tests answer "how fast?" Voice doesn't care much about fast — a business call uses less than a tenth of a megabit. Voice cares about consistent: every 20 milliseconds a phone sends a small packet, and if packets arrive late, out of order, or not at all, the caller hears it. These tests are tuned for that, and every result comes with the explanation I'd give a customer standing in the phone closet.
When the numbers say the problem is bigger than a router setting, send me your results or call 804-222-8009 and an Infotel Systems tech will look at it with you.
Frequently asked questions
How do I test the call quality of my internet connection?
Press Start on the call quality test at the top of this page. In about 25 seconds it measures latency, jitter and packet loss with repeated round trips to the nearest Cloudflare server, measures download and upload speed, and estimates a MOS score. Run it from the same network, and ideally the same computer or phone location, where call problems happen.
What internet speed do I need for VoIP phones?
Each G.711 call uses about 87 kbps in each direction on Ethernet, so 10 simultaneous calls need under 1 Mbps up and down. Plan on 100 kbps per call plus headroom. Upload is usually the limit on cable internet. Speed alone does not guarantee good calls — latency, jitter and loss under load matter more.
What is a good ping for VoIP?
Under 100 ms round trip to a nearby server is good. ITU-T G.114 recommends keeping one-way mouth-to-ear delay under 150 ms, which is roughly a 300 ms round trip end to end, including the carrier network and the phones' jitter buffers. A high ping to a nearby test server leaves little budget for the rest of the call path.
What is an acceptable jitter and packet loss for VoIP?
Jitter under 30 ms and packet loss under 1% are the common industry targets. Phones use a jitter buffer to smooth out delay variation, and codecs can conceal occasional lost packets, but beyond those levels callers hear choppy, robotic or clipped audio.
What is a MOS score?
MOS (Mean Opinion Score) rates call quality from 1 (bad) to 5 (excellent), as defined for listening tests in ITU-T P.800. Network tools estimate it from latency, jitter and loss using the ITU-T G.107 E-model. 4.0 or higher is good business quality; ordinary G.711 calls top out around 4.4.
Why does the test look fine but my calls are still bad?
Common reasons: the problem is intermittent (peak hours, someone uploading), it happens on Wi-Fi but you tested wired, it is on the carrier's side beyond the nearest test server, or it is a configuration problem the network test can't see, such as SIP ALG, NAT timeouts or a bad headset. Run the test at the time calls go bad, then try the Bufferbloat and IP/NAT tests.
Is this call quality test free and private?
Yes. The tests run in your browser. They send small test requests to public servers (Cloudflare for speed and latency, ipify for your public IP, Google and Cloudflare for STUN and DNS). Your latest results are kept only in your own browser so they can pre-fill the contact form if you choose to send them.
Can a better router fix VoIP call quality?
Often, yes. A router with Smart Queue Management or properly configured QoS can stop latency spikes when the line is busy, and turning off SIP ALG fixes many one-way audio and registration problems. If loss or jitter starts outside your building, the fix is with the internet provider or a second connection for backup.
Sources & references
- ITU-T G.114 — One-way transmission time
- ITU-T G.107 — The E-model (R-factor)
- ITU-T P.800 — Methods for subjective determination of transmission quality (MOS)
- IETF RFC 3550 — RTP: A Transport Protocol for Real-Time Applications
- Cloudflare speed test (endpoint used by these tools)
Written by Eric · Reviewed November 2026