Jitter: definition and a practical example
Short answer: Jitter is the variation in the arrival interval of a voice call's packets, which forces the receiver to hold packets in a buffer before playing the audio.
In Portuguese: Jitter
What is jitter?
Jitter is the variation in the arrival time of voice packets. On a VoIP call, the phone sends an RTP packet at fixed intervals, usually every 20 ms. If the network delivered every packet with the same delay, they would also arrive every 20 ms. In practice, router queues, Wi-Fi and congested links make some arrive early and others late. That irregularity is jitter.
To play the voice without stutters, the receiver holds packets in a jitter buffer and releases them at the right pace. The buffer handles small variations, but it has a price: every millisecond of buffer is one more millisecond of delay in the conversation. If a packet arrives after the moment it should have played, it is discarded and the listener hears a gap, just like packet loss.
RTCP, the control protocol that accompanies RTP, reports an estimate of interarrival jitter calculated with the method in RFC 3550.
Latency
Latency is the delay of a packet from one end to the other, regardless of variation. ITU-T G.114 recommends not exceeding 400 ms of one-way delay for general network planning and notes that below 150 ms most applications have essentially transparent interactivity. Above that, people start talking over each other and waiting for the other side to respond. Total delay adds up the network, the codec and the jitter buffer.
In practice
A phone sends four packets, one every 20 ms. Here is when they arrive:
| Packet | Sent (ms) | Arrived (ms) | Delay (ms) |
|---|---|---|---|
| 1 | 0 | 40 | 40 |
| 2 | 20 | 75 | 55 |
| 3 | 40 | 72 | 32 |
| 4 | 60 | 110 | 50 |
The delay ranges from 32 to 55 ms, a swing of 23 ms. With a 30 ms buffer, all of them play on time, and the perceived delay stays around 70 ms, well under G.114’s 150 ms. If a fifth packet arrived with 100 ms of delay, it would overflow the buffer and become a gap in the audio.
How Monitor uses this
The data circuit agent that 2N5 installs at each site measures latency, packet loss and jitter on that location’s internet link, and shows charts for the last 100 hours. That way you can see whether the link was unstable at the time voice quality dropped. See data circuits.
Frequently asked questions
What is acceptable jitter for VoIP?
No standard sets a single limit. What matters is whether the jitter fits in the phone's or the PBX's buffer. A 60 ms buffer absorbs variations smaller than that, but adds 60 ms to the call's delay. When jitter exceeds the buffer size, packets arrive too late, get discarded, and the effect is the same as packet loss.
Are jitter and latency the same thing?
No. Latency is how long a packet takes to travel from one end to the other. Jitter is how much that time varies from one packet to the next. A network can have high, stable latency with little jitter, or low latency with a lot of jitter. The first gets in the way of conversation; the second chops the audio.