[SR-Users] Best practice to figure out one way rtp delay

[Karsten Horsmann]

Hi List,

I have a new setup with two Kamailios installations. One serves siptrunks
from the internet and one is for internal routing.

So far so good. At the end I have an 3rd party b2bua that receives and
sends calls via the both Kamailios.

For testing purposes I setup an freeswitch with beep and then echo
application to the caller.

My call flow are like this

Call 1 to kam1 sbc then internal kam2 and b2bua.
B2bua make then new call no 2 vice versa to PSTN freeswitch with echo.
After this b2bua bridges the calls together.

This generates for the caller an beep and echo.

The interesting thing is now, that Call 1 gets an hearable delay of 1
second.

But only in the rtp steam from me to the caller.

The second calls seems equal of timing.

Since is a really new complete setup of hardware and stuff, the question of
its working before is answered with an no.


Now my question to you guys.

How can I get an measurable method to finding the delaying parts (could be
network, servers, applications etc).

Only capturing on one place don't did the trick for me.

Thanks for your hints

Cheers
Karsten
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