VoIP excessive jittter
Jitter describes the variations in latency of a VoIP transmission. In data networks, jitter refers to packet jitter, not bit jitter. Excessive packet jitter causes voice to sound garbled. Network components compensate for jitter with buffers. Jitter buffers store incoming packets and send them in a more constant stream; the buffers smooth the delivery of packets to produce a more even flow of voice data.
The size of a jitter buffer affects both jitter and latency. If audio transmissions have enough jitter to annoy users, then increasing the capacity of a jitter buffer can reduce jitter to acceptable levels. Too large a buffer, however, may cause latency to increase to a point where it's annoying to users. A typical jitter buffer delay is 20 ms, but often reaches 80 ms. There is no optimal size of jitter buffer because the buffer size will vary from network to network. For high-quality voice, the average inter-arrival time at the receiver should be nearly equal to the inter-packet gaps at the transmitter and the standard deviation should be low. Jitter buffers (packet buffers that hold incoming packets for a specified amount of time) are used to counteract the effects of network fluctuations and create a smooth packet flow at the receiving end.
Note that Observer can display jitter in RTP Time units or milliseconds (as described in the Expert Explanation for Jitter). When you change the display unit, the Expert Threshold value is unchanged; you should probably adjust it manually to reflect the different scale of measurement.
See also: