What is jitter, how serious is it and how can I get rid of it?

Jitter is the term used to explain the measure of the variable over time of the latency that appears across a network. In order to have good quality voice and video you must get rid of as much of the jitter as possible.

Jitter is caused by transit delay, which is caused by cueing, contention, and serialization effects. These issues take place on the pathway of the network that transfers the data from the beginning point to the destination point.

Jitter is more likely to come about when there is a slow or heavily congested link. This congestion is the cause of bottlenecking in the highway for the data packet's movement.

With VoIP when the timing variations happen the impact of the delay will come across in the sound quality. Therefore, the goal is to have as little jitter in the set up or your service and network as possible.

If you think about the jitter as a road block in the process of the data from getting from the staring location to the final location you can see the importance of removing as many of these jitter blocks as possible.

Jitter much like that of latency is a problem that originally caused some hesitation in the use of VoIP. However, with the technologies available today, the quality of sound through the process of minimizing jitter is great.

In order to remove or minimize jitter, allocation of the right amount of bandwidth and network usage is important. You can optimize the use of jitter buffering and packet size by implementing the use of the G711 codec in addition to avoiding asynchronous transcoding.

With the technology of the of "QoS" control mechanisms such as class based queuing, bandwidth reservation and of higher speed links such as 100 Mbit Ethernet, E3/T3 and SDH will reduce the incidence of jitter related problems to come at some point in the future, however jitter will remain a problem for some time.

Therefore prioritizing the VoIP traffic over the designated network layers 2 and 3 will make it so that there is a significant improvement in both jitter and latency.

Access paths that are congested will cause problems with jitter in your VoIP communications. There are IP service providers that will route routine traffic over a multiple internal route in their network; this is in order to provide a higher level of resilience. However, this can introduce a jitter result.

There have been many different approaches used to measure jitter. How ever at this time there does not appear to be a good representation of the jitter process.

There are jitter buffers that are designed to remover the residual effects of jitter in a network. The way this works is that the buffer will buff each arriving packet, this is for a very short time before creating the sound we hear.

A fixed jitter buffer maintains a constant size whereas an adaptive jitter buffer has the capability of adjusting its size dynamically in order to optimize the delay/discard tradeoff

Whether a fixed or an adaptive buffer is used, they are both capable of automatically adjusting the changes in delay.

Through most of the VoIP providers with today's technology the IP Telephony, devices use end-points or gateways that will take out the jitter caused by IP networks.

This de-jitter function is configurable to fit your networking needs. Someone listening to the voice quality of calls now made with VoIP would never know that at one point jitter was a real problem. With the right amount of bandwidth and a good buffer, jitter is almost a thing of the past.