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Problem: Jitter

Excessive jitter can result from congestion on LANs, Access Links, low bandwidth WAN links/trunks or the use of load sharing.

In depth discussion of jitter sources


High levels of jitter cause large numbers of packets to be discarded by the jitter buffer in the receiving IP phone or gateway. This may result in severe degradation in call quality or large increases in delay.

Example audio file - 5% packet discard rate:

Example audio file - 10% packet discard rate:

Example audio file - 25% packet discard rate:


Jitter measurements can be difficult to interpret. RFC3550 (RTP/RTCP) provides a simple running average of the packet to packet delay variation - this provides useful information about the 500 milliseconds before the RTCP report was sent and hence only reports on about 5-10% of the call. The Jitter Envelope measurement provided by VQmon is more accurate and measures jitter in a way that correlates better with the impact of jitter on VoIP systems.

For jitter levels under 100 milliseconds then it may be acceptable to increase the jitter buffer size in end-systems or to enable adaptive jitter buffer operation.

For jitter levels over 100 milliseconds then increasing the jitter buffer size to avoid packet discards will introduce significant delay and cause conversational problems; in this case it is preferable to take steps to isolate the source of jitter and eliminate the problem at source.


Network Analyzer, VQmon

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