All SPA9xx phones have option to Send Stats in Bye . The key role of this function is to send statistics of made calls(time, used codecs, no of packets send/received, no of lost packets, delay and jitter). Statistics are being sent, but not everything is OK, the value of delay is always 0 and jitter is about 0-2, these values seem to be not real.
This function would be nice add on to use on ITSP platfroms and servers. Does anyone know how these values in SPA9xx are calculated?
Is there anything man can do to obtain real values of delay and jitter?
BYE only shows the snapshot at the end of the call.
Consider using packet lost information if you're only relying on the last BYE/200 information.
If you want more information, consider monitoring the entire call for RTCP exchanges.
While on a call, you can use the phone's web-ui to view the Info tab and view the current values. Use the browser's refresh to update the stats.
delay = 0 means that the delay cannot be measured for the call.
To measure delay, both peers must have RTCP enabled.
You can use 2 SPA942 phones, enable RTCP on both, and try again. The delay should be small, but should be non-zero.
On a LAN, the delay will be tens of microseconds. Delay may still show 0 in the statistics which are expressed in ms).
Delay is not very accurate because it only measures the delay due to the network traffic and does not include any buffering delay experienced in the receivers.
Jitter is the running average.
You can also refer to the RTCP RFC to see how delay and jitter are computed.
Thanks for the questions.