<div dir="auto"><div> Hi yuriy, </div><div dir="auto"><br></div><div dir="auto">Of course per default Kamailio don't handle rtp. I forgot that I use rtpengine on the "sbc" with Kamailio. </div><div dir="auto"><br></div><div dir="auto">And as you all know, if it's sounds bad, then we sip guys get the tickets... </div><div dir="auto"><br></div><div dir="auto">So that's an question for your experience in this area. </div><div dir="auto"><br></div><div dir="auto">Cheers </div><div dir="auto">Karsten <br><br><div class="gmail_quote" dir="auto"><div dir="ltr" class="gmail_attr">Yuriy Gorlichenko <<a href="mailto:ovoshlook@gmail.com">ovoshlook@gmail.com</a>> schrieb am Do., 6. Feb. 2020, 10:27:<br></div><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="auto">Kamailio does not handle RTP.<div dir="auto"><br><div dir="auto">But you can use heplify ( or homer) </div><div dir="auto">That will alliw you to collect rtcp in your infrastructure and map it with the SIP calls you have. </div></div></div><br><div class="gmail_quote"><div dir="ltr" class="gmail_attr">On Thu, 6 Feb 2020, 09:23 Karsten Horsmann, <<a href="mailto:khorsmann@gmail.com" target="_blank" rel="noreferrer">khorsmann@gmail.com</a>> wrote:<br></div><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="auto">Hi List,<div dir="auto"><br></div><div dir="auto">I have a new setup with two Kamailios installations. One serves siptrunks from the internet and one is for internal routing.</div><div dir="auto"><br></div><div dir="auto">So far so good. At the end I have an 3rd party b2bua that receives and sends calls via the both Kamailios. </div><div dir="auto"><br></div><div dir="auto">For testing purposes I setup an freeswitch with beep and then echo application to the caller. </div><div dir="auto"><br></div><div dir="auto">My call flow are like this </div><div dir="auto"><br></div><div dir="auto">Call 1 to kam1 sbc then internal kam2 and b2bua. </div><div dir="auto">B2bua make then new call no 2 vice versa to PSTN freeswitch with echo.</div><div dir="auto">After this b2bua bridges the calls together. </div><div dir="auto"><br></div><div dir="auto">This generates for the caller an beep and echo.</div><div dir="auto"><br></div><div dir="auto">The interesting thing is now, that Call 1 gets an hearable delay of 1 second. </div><div dir="auto"><br></div><div dir="auto">But only in the rtp steam from me to the caller. </div><div dir="auto"><br></div><div dir="auto">The second calls seems equal of timing. </div><div dir="auto"><br></div><div dir="auto">Since is a really new complete setup of hardware and stuff, the question of its working before is answered with an no. </div><div dir="auto"><br></div><div dir="auto"><br></div><div dir="auto">Now my question to you guys. </div><div dir="auto"><br></div><div dir="auto">How can I get an measurable method to finding the delaying parts (could be network, servers, applications etc).</div><div dir="auto"><br></div><div dir="auto">Only capturing on one place don't did the trick for me. </div><div dir="auto"><br></div><div dir="auto">Thanks for your hints </div><div dir="auto"><br></div><div dir="auto">Cheers </div><div dir="auto">Karsten </div></div>
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