Video Calls Keep Dropping: Find the Cause on Your Home Internet

By Paul Snyman · Published September 12, 2026 · 7 min read

Speed Is Not the Measurement

A video call needs a few megabits in each direction. Almost every home connection has that. What a call cannot tolerate is inconsistency: packets that arrive in a clump after a pause, and packets that never arrive at all.

• Jitter is the variation in arrival time. Audio and video are played back on a clock, so a receiver holds a small buffer; when jitter exceeds the buffer, you get the robot voice and the frozen frame.
• Packet loss is packets that disappear. Above a few percent, calls degrade visibly. Above ten, they collapse.
• Upload is the direction people forget. You are sending video too, and asymmetric connections often have a tenth of the download figure going up.

None of those appear on a speed-test result headline. Here is how to measure each one.

Step 1: Measure During an Actual Call

This matters: measure while a call is running, from the same device if you can or from a phone on the same WiFi if not. A network that is clean when idle tells you nothing about a network under a call's load.

Open PingKit, go to the Monitor tab and start Live Connection. It pings once a second and graphs the last sixty results with the current, average, minimum and maximum latency, the jitter figure and any timeouts.

Live Connection runs while the view is open and the screen is on. That is an iOS constraint, not a design choice, and it is why the last step of this page exists. For the background on the measurement itself, Network Jitter: What It Is and How to Fix It.

Step 2: Put a Number on the Loss

Open Ping Test, target 1.1.1.1, set the count to 100 and run it during a call. One packet a second, with loss reported as a percentage. Under 1% is fine. Two to five percent will be audible. Above ten, nothing will help until it is fixed.

Then run the Speed Test and read the upload figure. If it is under about 3 Mbps, you are close to the floor for a single good video call, and two people on calls in one house will fight each other.

Step 3: Find Out Where the Loss Starts

Open MTR, target 1.1.1.1, and let it run. MTR pings every hop on the path repeatedly, so it reports loss per hop rather than just end to end.

• Loss at hop 1 (your router). Your WiFi or the local link. This is by far the most common and the most fixable.
• Loss starting at your ISP's first hop and continuing to the end. The problem is your line or your provider. Now you have something to report.
• Loss at one middle hop that does not continue. That router is deprioritising its own ICMP replies. It is normal and it is not your fault.

The "continuing to the end" test is the important one, and it is the thing traceroute cannot tell you: What Is MTR and Why It's Better Than Traceroute.

Step 4: The Fixes, in Order of How Often They Work

1. Use Ethernet for the machine you take calls on. It removes jitter caused by the radio entirely. If that is not possible, move to 5 GHz and get closer to the access point.
2. Stop competing uploads. Cloud photo backups, a synchronising drive, a game update and a second person's call all take the same upstream. On an asymmetric line, upload is the scarce resource.
3. Look at the 2.4 GHz band. If the calling device is on 2.4 GHz in a busy building, jitter is nearly guaranteed in the evening. A Mac can show you its own band, channel, signal strength and noise floor: Which WiFi Channel and Band Your Mac Is Using. An iPhone cannot, because iOS does not expose it.
4. Turn off the VPN for the call, as a test. A VPN adds a hop, latency and often jitter. If calls are fine without it, you have your answer.
5. Reduce what the call sends. Turning off HD video, or your own camera, cuts the upstream requirement substantially and is a legitimate fix on a thin line.

More on reducing the variation itself: How to Reduce Jitter on WiFi and Internet.

Step 5: Catch It When You Are in the Meeting, Not Watching a Graph

The difficulty with calls is that you cannot stare at a latency chart while presenting. An iPhone cannot watch for you, because iOS does not let an app keep measuring while it is closed.

The free PingKit Agent on a Mac pings the internet every thirty seconds and raises a High Latency alert when the round trip crosses a threshold you set. It is on by default at 100 ms, with a ten-minute cooldown so a bad hour does not produce a hundred alerts. On a fibre line, setting that to 60 ms will catch problems long before a call does. The Agent also raises Internet Down after three consecutive failed checks, which is what tells you the call did not drop because of Zoom.

Those alerts, and the latency history behind them, appear on the iPhone with PingKit Guardian. The alert reaches the iPhone through your own private iCloud, which is one hop and best effort rather than instant. The useful part is afterwards: you can look at the timeline and see whether the 3pm call dropped during a latency spike, or whether the network was clean and the problem was somewhere else entirely.

Honest Limits

• PingKit does not measure the call itself. It measures the network the call is riding on, which is the part you can change.
• Live Connection runs only while its view is open. It cannot record a call you take tomorrow.
• Latency alerts are based on ping round-trip time to public addresses. They will not catch a fault that only affects one conferencing provider's servers.
• There is no bufferbloat or latency-under-load test in PingKit on either platform, so the specific "fast line that collapses under load" measurement has to be inferred from Live Connection during a transfer.
• The Agent needs a Mac that stays awake, on the same network.

Frequently Asked Questions

Why do my video calls keep dropping when my internet is fast?

Because calls fail on jitter and packet loss, not on throughput. A fast line can deliver packets in irregular clumps or lose a few percent of them, and both wreck a call while leaving a speed test looking healthy.

What jitter is acceptable for Zoom or Teams?

Single-digit milliseconds is comfortable. Tens of milliseconds is where audio artefacts and frozen video start. The figure matters far more during a call than when the network is idle, so measure while one is running.

How much upload speed does a video call need?

Roughly 3 Mbps for one good HD call, and more if several people in the house are calling at once. On asymmetric connections upload is the scarce direction, which is why a 500 Mbps download figure can coexist with unusable calls.

Is it my WiFi or my ISP?

Run MTR and look at where loss begins. Loss at the first hop is your local network. Loss that starts at your provider's first hop and continues to the destination is theirs. Loss at one middle hop that does not continue is that router deprioritising replies and is not a fault.

Can I be alerted when my connection gets bad?

A Mac running the free PingKit Agent raises a High Latency alert when the round trip crosses a threshold, on by default at 100 ms with a ten-minute cooldown, and an Internet Down alert after three consecutive failed checks. With Guardian those alerts, and the latency history behind them, reach your iPhone.

Related Articles

• Live Connection: latency, jitter and loss as it happens
• Network Jitter: What It Is and How to Fix It
• How to Reduce Jitter on WiFi and Internet
• MTR: per-hop loss and latency along the whole path