VoIP jitter,
commonly known as network jitter, is a type of jitter that occurs in VoIP phone system. It happens when the network delay is greater than the codec delay, and
it will result in unstable call quality.
This can be
easily fixed by adding more bandwidth to VoIP phones and servers. Another
solution to solving this problem is to use a QoS (Quality of Service) solution
to manage the traffic flow. It will ensure QoS is delivered in a VoIP network
so that voice quality can be controlled.
VoIP jitter
usually referred to as network jitter, is the time delay experienced by VoIP
phone systems. This will result in unstable calling quality and can cause
problems like choppy or garbled voice.
Why is network jitter important?
The scenario
described above is precisely why combating VoIP network jitter is so important.
These problems are an inconvenience for users, but they can also cause
expensive call delays that cut into your business's profits.
Once again,
the solution to minimizing jitter in VoIP phone systems is to upgrade bandwidth
and add QoS (Quality of Service) on network switches. Voice quality could be
easily controlled using QoS.
When VoIP jitter is in the normal range?
VoIP jitter
can be deemed in normal ranges when the delay in network frame is in the 30ms
to 120ms range. It is also important to note that "jitter buffer"
should be properly sized for better performance of voice quality. The rule of
thumb would be to size it about 75% of the normal range.
VoIP jitter is an unavoidable thing?
Yes, VoIP
jitter is not something that can be completely eliminated. However, there are
things you can do to minimize it and keep your phone system running smoothly.
One option would be to simply upgrade bandwidth on your network switches. This
will help to minimize the delay in the network frame and improve the quality of
your VoIP phone system.
What is a VoIP jitter buffer?
A VoIP
jitter buffer is a mechanism that controls the timing of voice packets, so they
can be transmitted at inconsistent time intervals. It also helps to combat
packet loss which causes choppiness in calls.
In order to
maintain a high-quality VoIP phone system, proper sizing of the jitter buffer
is required. The rule of thumb would be to size it about 75% of the normal
range. Excessive or insufficient sizes will result in an unstable voice call
which represents poor performance and satisfaction issues for users.
What is VoIP latency what does it have to
do with network latency?
VoIP jitter
is the variability over time of network latency. It happens when the network delay is greater
than the codec delay and can result in unstable call quality.
The packet
loss will cause choppiness and garbled voice. When this happens, it usually
means that there is a problem with your bandwidth or router, or perhaps you are
working too far from the data center where your VoIP phone system is hosted.
Latency is
an inevitable and often unavoidable problem, and it's important to find a way
to minimize the effect of high latency on voice quality. One option would be to
upgrade bandwidth or add QoS (Quality of Service) on network switches. This
will help control traffic flow in your network so that voice quality can be
controlled.
How can I reduce latency in my VoIP phone
system?
Latency
occurs when the time to transmit packets is greater than the codec delay. This
will result in delayed voice transmission, which can cause garbled or choppy
phone calls.
To manage
the issue of latency, you should usually upgrade bandwidth for both your VoIP
servers and the network switches. QoS (Quality of Service) can be an added
advantage for controlling traffic flow in your VoIP phone system so that voice
quality can be controlled.
