What Is Jitter? Why It Ruins Video Calls & How to Fix It
You are on an important Zoom call. Your internet speed test says you have 200 Mbps. But voices sound robotic, the video freezes every few seconds, and people keep asking you to repeat yourself. The culprit is almost certainly jitter -- one of the most misunderstood metrics in internet performance, and one that traditional speed tests rarely measure.
Jitter is the variation in packet arrival times on your network. When data packets arrive at irregular intervals instead of in a smooth, consistent stream, real-time applications like video calls, VoIP, and online gaming break down. Even if your raw download speed is blazing fast, high jitter can make your internet connection feel completely unreliable.
What Is Jitter in Networking?
At its simplest, jitter (also called packet delay variation) measures how consistent your connection's timing is. When you send or receive data across the internet, that data is split into small packets. Each packet travels from point A to point B, ideally arriving at evenly spaced intervals.
Imagine a metronome ticking perfectly at 20 milliseconds per beat. That is a stable, low-jitter connection -- packets arrive like clockwork at 20ms, 20ms, 20ms. Now imagine that metronome going haywire: 20ms, then 45ms, then 12ms, then 60ms, then 25ms. The average delay might still be reasonable, but the variation is enormous. That variation is jitter, and it wreaks havoc on anything that requires real-time data delivery.
Network jitter is typically measured in milliseconds (ms) and calculated as the standard deviation or average difference between consecutive packet arrival times. When you run a jitter test on pong.com, we send multiple ping samples in rapid succession and calculate how much the timing varies between them. This gives you a precise picture of your connection's consistency -- something a basic speed test completely ignores.
In the illustration above, both connections deliver packets -- but the jittery connection's wildly inconsistent timing makes it far worse for real-time applications, even though individual packets may sometimes arrive faster than those on the stable connection.
How Jitter Destroys Video Calls
Real-time applications like Zoom, Microsoft Teams, and Google Meet are uniquely vulnerable to jitter. Unlike loading a web page or downloading a file -- where your device can wait for delayed packets and reassemble data at its leisure -- video conferencing requires packets to arrive in a continuous, predictable stream.
When jitter is high during a video call, here is what you experience:
- Choppy or robotic audio: Voice packets arriving out of order or with gaps create that unmistakable "robot voice" effect where words become garbled or fragmented.
- Frozen video frames: When video packets arrive late, your device displays the last good frame while waiting, creating freeze-frames that last anywhere from a fraction of a second to several seconds.
- Lip sync issues: Audio and video packets travel separately. When jitter affects them differently, you see someone's mouth moving out of sync with their voice -- like watching a badly dubbed movie.
- Audio dropouts: Packets that arrive too late are simply discarded by real-time applications. This creates gaps in audio where entire words or phrases disappear.
- Increased CPU usage: Your device's jitter buffer works overtime trying to smooth out irregular packet delivery, consuming more processing power and sometimes making things worse.
Video conferencing apps use a "jitter buffer" to smooth out minor variations in packet timing. But this buffer can only compensate for so much -- typically 20-50ms of variation. Beyond that, quality degrades rapidly. The larger you set the buffer, the more delay you add to the call.
Jitter vs Latency vs Packet Loss
These three metrics are related but measure very different things. Understanding the distinction helps you diagnose and fix the right problem.
| Metric | What It Measures | Analogy | Impact |
|---|---|---|---|
| Latency (Ping) | Total round-trip time for a packet | How long the commute takes | Overall delay / responsiveness |
| Jitter | Variation in packet arrival times | How unpredictable the commute is | Audio/video quality and consistency |
| Packet Loss | Percentage of packets that never arrive | Packages lost in the mail | Missing data, glitches, disconnects |
Latency is the total time for a data packet to travel to a server and back. If your ping is 30ms, every packet takes about 30ms to make the round trip. High latency means everything feels delayed, but at least it is consistently delayed.
Jitter is the inconsistency in that timing. You might have an average latency of 30ms, but if individual packets vary between 10ms and 80ms, that is severe jitter. Your applications cannot predict when data will arrive, making it impossible to deliver smooth audio and video.
Packet loss means some packets never arrive at all. While latency and jitter affect timing, packet loss removes data entirely. Even 1-2% packet loss can devastate call quality. When you test your connection at pong.com, we measure all three metrics so you can identify exactly where your connection falls short.
What Is an Acceptable Jitter for Internet?
Acceptable jitter depends on what you are doing online. General web browsing is almost unaffected by jitter, while VoIP phone calls demand extremely low jitter to maintain voice clarity. Here is how different jitter ranges translate to real-world experience:
For most people, jitter under 15ms is the target. At this level, video calls remain smooth, VoIP audio sounds natural, and online games feel responsive. Once jitter climbs above 30ms, you will start noticing quality problems in almost any real-time application. Above 50ms, calls become difficult and gaming becomes frustrating.
The acceptable jitter threshold also varies by application. Cisco and the ITU-T recommend keeping jitter below 30ms for VoIP, but professional-grade calls and conferencing perform best under 15ms. Most competitive gaming platforms recommend under 30ms jitter, though serious players prefer under 10ms.
Jitter Impact by Application Type
Different applications have vastly different tolerances for jitter. This is because of how each application handles incoming data. Buffered content like Netflix or YouTube can absorb jitter by pre-loading seconds of video. Real-time applications cannot buffer ahead because the data has not been created yet.
Notice that VoIP and phone calls are the most sensitive. This is because audio compression codecs like Opus and G.711 generate tiny packets at very frequent intervals (typically every 20ms). Any disruption in that steady cadence is immediately audible. Video conferencing is slightly more tolerant because video codecs can handle a few dropped frames without catastrophic quality loss. Gaming tolerates somewhat higher jitter because game engines use prediction algorithms to smooth out small timing inconsistencies.
How Pong.com Measures Jitter
When you run a jitter test at pong.com, the measurement goes far beyond a simple ping. Here is what happens behind the scenes:
- We send a rapid sequence of ping samples to Cloudflare edge servers distributed across over 200 cities worldwide.
- Each ping sample records a precise round-trip time in milliseconds.
- We calculate the difference between consecutive ping measurements.
- We compute the standard deviation of those differences, which gives you a single jitter value in milliseconds.
- We also measure jitter under load -- while your connection is simultaneously handling download and upload traffic -- to simulate real-world conditions where you are on a video call while others in your household are streaming.
This approach is critical because jitter often increases dramatically under load. Your connection might show 2ms jitter when idle but spike to 40ms when someone starts a download. Testing jitter under load reveals problems that idle-only tests miss entirely. Pong.com reports both idle and loaded jitter so you get the full picture of your connection's consistency.
Unlike traditional speed tests that measure jitter to a server inside your ISP's network, pong.com measures through the real public internet. This means your jitter results reflect the actual path your video call data travels -- not an artificially optimized route.
What Causes Jitter on Your Network?
Understanding what causes jitter is the first step to fixing it. Network jitter typically originates from one or more of these sources:
Wi-Fi Interference and Congestion
Wi-Fi is the single most common cause of high jitter in home networks. The 2.4 GHz band is crowded with signals from neighboring networks, Bluetooth devices, microwaves, and baby monitors. Even on the cleaner 5 GHz band, walls, floors, and distance from the router introduce signal variability. Each packet has to wait for a clear moment to transmit, and those waits are unpredictable -- creating jitter. Switching to a wired Ethernet connection can drop your jitter from 15-30ms down to 1-3ms instantly.
Network Congestion and Bufferbloat
When multiple devices compete for bandwidth on your home network, packets get queued in your router's buffers. This queuing adds variable delay to every packet. The problem gets worse with bufferbloat -- oversized router buffers that hold packets for hundreds of milliseconds. A saturated connection with bufferbloat can send jitter soaring from single digits to 100ms or more. You can test for bufferbloat alongside jitter at pong.com.
Outdated or Overloaded Hardware
Old routers with slow processors struggle to manage modern traffic loads. When a router's CPU maxes out, packet forwarding becomes inconsistent, adding jitter. ISP-provided modem/router combos are frequent offenders -- they are built to a budget and often lack the processing power for smooth traffic management under load.
ISP Routing and Peering Issues
Sometimes jitter originates beyond your home network. Congested links between ISPs (peering points), overloaded routing equipment at your ISP, or suboptimal routing paths can all introduce packet timing variability. This type of jitter is harder to fix because it is outside your control -- but identifying it (by ruling out home network causes) gives you leverage when contacting your ISP.
How to Fix High Jitter
The good news is that jitter is often fixable with straightforward changes. Start with the easiest fixes and work your way down:
- Use a wired Ethernet connection -- This is the single most effective fix. Run a Cat 5e or Cat 6 cable from your router to your computer. Wi-Fi jitter disappears immediately.
- Enable SQM (Smart Queue Management) -- If your router supports SQM with fq_codel or CAKE, enabling it can eliminate jitter caused by bufferbloat. This is the second-most impactful fix.
- Reduce network congestion -- Limit simultaneous heavy downloads during video calls. Use QoS (Quality of Service) settings to prioritize real-time traffic.
- Upgrade your router -- Replace ISP-provided hardware with a quality router that supports SQM. Routers running OpenWrt or those with built-in SQM support are excellent choices.
- Switch to 5 GHz or Wi-Fi 6 -- If you must use Wi-Fi, use the 5 GHz band and a Wi-Fi 6 (802.11ax) router. Less interference means more consistent packet timing.
- Close background applications -- Cloud backups, software updates, and streaming on other devices all add congestion that increases jitter.
- Restart your networking equipment -- A simple power cycle of your modem and router clears memory buffers and can resolve intermittent jitter issues.
- Contact your ISP -- If jitter persists after optimizing your home network, the issue may be upstream. Share your pong.com test results showing high jitter to support your case.
Run a jitter test at pong.com before and after each fix to measure the improvement. This helps you identify which change had the biggest impact and gives you concrete data if you need to escalate to your ISP.
Frequently Asked Questions About Jitter
What is an acceptable jitter for internet?
Does jitter affect gaming?
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Is 10ms jitter good?
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