VoIP Active Jitter
Voice quality depends on packets arriving in even stream; early or late arrival due to network impairments is called "jitter." Observer quantifies jitter as the mean variation packet delay, expressed in timestamp units. Timestamp units are dependent upon the codec and sampling rate in use.
Jitter can be caused by a number of factors, including:
LAN congestion.
"Route flapping," where network configuration and current conditions are causing routers to change routes upon every table update.
Firewall routers that duplicate the IP stream (rather than passing it through) can cause jitter as they become overwhelmed with traffic.
Firewall routers that duplicate the IP stream (rather than passing it through) can cause jitter as they become overwhelmed with traffic.
Access link impairments, such as congestion or the effects of load sharing schemes. Any bottlenecks along the voice path can result in jitter.
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.