Glossary
Jitter
Jitter is the variation in the time intervals at which packets arrive at a receiver. In real-time audio and video communication, high jitter causes audible glitches, choppy playback, and degraded call quality even when overall bandwidth is sufficient.
General definition
Jitter arises because IP networks route packets independently, and packets in the same stream can take different paths or experience different queuing delays at routers. While the total bandwidth may be adequate, uneven packet arrival timing disrupts the smooth playback required for real-time media.
- Low jitter (under about 30 ms for voice): generally imperceptible
- Moderate jitter (30-100 ms): noticeable gaps and slight choppiness in audio
- High jitter (above 100 ms): significant distortion or lost syllables in voice calls
- Jitter buffers smooth arrival variance by holding packets briefly before playback
WebRTC engines include adaptive jitter buffers that dynamically adjust their hold time based on observed network conditions. Longer buffers reduce jitter impact at the cost of added latency, so implementations tune the trade-off continuously. Jitter is measured alongside packet loss and round-trip time as one of the three primary network quality signals for real-time media.
In the Ethora ecosystem
Ethora’s voice and video calling is built on WebRTC, which includes built-in adaptive jitter buffering in all major browser implementations. This means jitter compensation is handled automatically, and call quality metrics including jitter levels are available via the WebRTC Stats API for monitoring and diagnostics.
Teams running Ethora in a self-hosted configuration on high-performance infrastructure can minimise jitter at the network level by optimising server placement and using dedicated network paths, which matters particularly for clinical or financial use cases where call quality directly affects user trust.
