Quality tests SIPP

Good day, I am doing quality tests on an Asterisk server with SIPP, as you can see in the picture, apparently the calls are made, however when reviewing the bandwidth consumption in Wireshark there is no change, any suggestions?

Looks like the sipp script only makes the call and asterisk answers and disconnects the call, as the average call length is 190/1000 of a second.

To get a better idea of load and quality, have asterisk answer the call, start ie MOH for a random amount of time then disconnect the connection.

Then you test call setup, load generated by MOH on the system and bandwidth consumption, although its only rtp from asterisk not going to asterisk.

Can you show the ladder diagram section? Are you sending an RTP load?

Increase the time per call, however there is no evidence of any increase in bandwidth.

You should add an RTP load and leave the call up for longer if you’re trying to test anything other than the call connecting and then immediately disconnecting. Also, 20 calls isn’t very many.