Packet loss, delay and jitter are factors that affect voice quality
on a VoIP network. The VoIP Data screen is designed to provide you the
ability to place and receive calls, and measure packet loss, delay and
jitter.
Once the meter has successfully ranged and registered with the CMTS, the
View Data menu item is enabled on the View menu.
NOTE:
- VoIP test can only be initiated after the meter has successfully registered
with the CMTS, and the MTAs has successfully initialized.
- VoIP mode cannot be initiated if Ranging Only is configured.
- A call cannot be placed or received if registration with the CMTS has
failed.
Results for the current, maximum, and average packet loss, delay, and jitter
are displayed on the screen once a call is placed or received. You can
save test results for archival purposes and use the existing VoIP file
to compare previously collected data. Details of each parameter is provided
as follows:
Packet loss
Significant packet loss degrades voice quality. Packet loss occurs when
packets are lost on a network, when packets are delayed too long, when
packets arrive out of order. Packet loss can cause reconstructed speech
to sound choppy and distorted.
Delay
Total end-to-end packet delays severely degrade voice quality by causing
long delays between callers and echo problems. Any packet delays less
than 150 ms provide acceptable speech quality. Delays between 150 and
400 ms begin to interfere with conversations and cause noticeable degradation.
Any delays greater than 400 ms is unacceptable.
Jitter
Jitter occurs when packets are sent at equal intervals, but received at
uneven time intervals, this can cause audible pops, clicks and a greater
delay of audio communications. Gateways compensate for this by accumulating
the received packets into an internal buffer and "playing them out"
at the proper time intervals and in packet order. The more jitter buffer
available, the more the network can reduce the effects of jitter.