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

Yuriy Gorlichenko ovoshlook at gmail.com
Thu Feb 6 10:25:52 CET 2020


Kamailio does not handle RTP.

But you can use heplify ( or homer)
That will alliw you to collect rtcp in your infrastructure and map it with
the SIP calls you have.

On Thu, 6 Feb 2020, 09:23 Karsten Horsmann, <khorsmann at gmail.com> wrote:

> 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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