Jitter test for live streaming: what jitter is and how we check a venue.
A jitter test measures how much the delay on a connection varies from moment to moment. A stream can live with a steady delay. It struggles with a delay that keeps changing, because every buffer has to be sized for the worst moment. This is what jitter does to live video, the thresholds StreamCertified uses, and how we check a venue for it.
- Reading time
- Updated
- < 15 ms
- 30 ms
- <= 80 ms
- 40
What does a jitter test measure?
A jitter test measures jitter: how much the delay on a network connection varies from one moment to the next. If every packet takes 40 ms, the delay is steady and jitter is low. If packets take 20 ms, then 90 ms, then 35 ms, jitter is high.
A live stream can cope with a steady delay. It struggles when the delay keeps changing, because the encoder, the transport and the player all have to buffer for the worst moment. Too little buffer and the picture stutters. Too much and the stream falls further behind the room.
We check jitter on every venue with the StreamCertified jitter test.
Jitter and latency are not the same thing
| Latency | Jitter | |
|---|---|---|
| What it is | How long a packet takes to arrive | How much that time varies |
| What the viewer sees | The stream runs behind the room | Stutter, buffering, audio drift |
| Can a buffer absorb it? | It does not need to; the delay is fixed | Only by adding delay |
| Ready (StreamCertified) | 80 ms or less to the test server | Under 15 ms |
A satellite link can have high latency and low jitter, and carry a clean stream. A busy venue Wi-Fi can have low latency and high jitter, and break one.
Why jitter matters for live video
Live video is a steady flow of frames. The network delivers them unevenly. Buffers smooth that out, and every buffer adds delay.
- SRT holds packets for a set latency window so it can re-order them and re-send lost ones. If jitter is larger than that window, late packets count as lost. See SRT.
- Players buffer a few seconds of video. High jitter drains the buffer and the viewer sees a spinning wheel.
- Interactive paths, such as remote speakers and interpreters, need a small buffer to keep conversation natural. They feel jitter first.
So jitter decides how much delay you must accept to keep the picture clean. For low latency streaming, jitter is often the real limit.
How much jitter is too much
These are the thresholds StreamCertified uses. Jitter is graded strictly under each figure.
| Jitter | Verdict | What we do |
|---|---|---|
| Under 15 ms | Ready | Standard settings. |
| 15 ms to under 30 ms | Check | Larger transport buffer, wired port, re-test under load. |
| 30 ms or more | Not ready | Not the show path on its own. Fix the network or add a second path. |
Latency to the test server is graded alongside it: 80 ms or less is ready, up to 150 ms needs checking. That round trip is to the server named on the result, not to your internet provider.
How the jitter test measures it
The tool times 40 round trips to its test server, with a 3-second timeout on each. Jitter is the standard deviation of those round trips. Latency is the fastest one, which is the path with queueing noise stripped out. The result also shows the average, the slowest trip, how many requests did not answer, and a second jitter figure based on the difference between consecutive trips.
It measures round-trip jitter over HTTP, not the jitter of a media stream itself. That makes it a fast, fair read of the network before any kit is set up. On show day, SRT's own statistics give us the media-path picture.
How we check a venue for jitter
- Wired and Wi-Fi, side by side. Wi-Fi jitter climbs with the crowd. The wired port is usually far steadier. The comparison makes the case to venue IT.
- Look at the slowest trip. The average hides spikes. One 400 ms round trip in forty is the moment a stream stutters.
- Test when the room is full. Registration, coffee breaks and the moment everyone opens a phone are when jitter peaks.
- Set the buffer from the reading. We size the SRT latency window from the measured round trip and its variation, not from a default.
Jitter is one line of our full venue readiness check, alongside upload and packet loss.
What we change when jitter is high
We move the encoder off Wi-Fi. We ask for a dedicated line or VLAN, with traffic priority if the venue supports it. We raise the transport buffer and accept a little more delay for a clean picture. And we add a second path, bonded 5G, so if the venue network turns unstable the stream carries on and a failure never reaches your audience.
This is standard on our live streaming in Dubai shows.
Run the free jitter test on StreamCertified
If the result worries you, send us the venue and we will test it first.
Questions we get asked.
What is a good jitter for live streaming?
Under 15 ms is ready by the thresholds StreamCertified uses. From 15 ms to under 30 ms needs checking. At 30 ms or more the connection should not carry a live stream on its own.
Is jitter worse than latency for a live stream?
Usually. A steady delay can be planned around. A delay that keeps changing forces bigger buffers, and past a point it causes stutter and lost packets.
Why is jitter so high on venue Wi-Fi?
Wi-Fi is shared airtime. Every guest phone competes for it, so the delay swings with the crowd. A wired port on a dedicated line is far steadier.
Can SRT handle jitter?
Yes, within its latency window. SRT buffers and re-orders packets for the time you set. If jitter is larger than that window, late packets are treated as lost.
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