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.