Jitter Test

Jitter is the variation in the time it takes voice packets to arrive. If one packet takes 20 ms and the next takes 70 ms, that 50 ms swing is jitter. Phones smooth it with a jitter buffer; too much jitter causes late packets to be discarded, which sounds choppy or robotic.

Under 30 ms is the common target for business VoIP, and under 20 ms is good. Short spikes are absorbed by the phone's jitter buffer; sustained jitter above 30 ms usually causes audible problems.

Wi-Fi interference and retries, a congested internet connection without QoS, bufferbloat in the modem or router, bad cables or duplex mismatches, VPN tunnels, and congestion on the internet provider's network.

Move phones to wired Ethernet, enable QoS and prioritize voice (DSCP EF 46), enable Smart Queue Management on the router, put phones on a voice VLAN, replace damaged cables, and contact the internet provider if jitter is high on a quiet wired connection.

A jitter buffer is a small holding area in the receiving phone that collects incoming voice packets and plays them out at a steady rate. It trades a little extra delay for smoother audio. Adaptive jitter buffers resize themselves as network conditions change.

For call quality, often yes. Callers tolerate a steady delay far better than an unsteady one. A steady 80 ms connection usually sounds better than a 20 ms connection that swings to 150 ms every few seconds.

Yes. Video and voice meetings use the same real-time techniques as VoIP phones, so high jitter causes frozen video, robotic voices and dropped audio there as well.