It will be helpful if you can provide a pcap with the network capture of sip packets for such call taken on kamailio server. Then we can see if rtp relaying was engaged or not.
Cheers, Daniel On 04/08/16 04:55, Kotb, Amir wrote: > > [03:50] <meamo> I have a small query, can anybody assist please? > [03:51] <meamo> I am running kamailio + rtpengine on a google compute > cloud instance > [03:51] <meamo> have setup them to work with devices behind nat, and > everything works > [03:52] <meamo> no i have added freeswitch. but I don't know how to > configure rtpengine to work with both > [03:52] <meamo> when I use #!define with_nat > and #!define with_freeswitch, I get no voice in the calls. When I > remove, #!define with_freeswitch, everything works normally > > Best regards, > Amir > > -- > *Amir KOTB*, Msc., Bsc. > > Postgraduate Researcher > Department of Electrical Engineering & Electronics, > *The University of Liverpool*, > Brownlow Hill, > Liverpool L69 3GJ, > UK > > Mobile: _+44-(0) 7428844234_ > Email: _a.k...@liverpool.ac.uk_ > Skype: _A.kotb1_ > Web: _Http://uk.linkedin.com/in/AOKotb_ > > Please consider the environment before printing this email > > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users@lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla http://www.asipto.com - http://www.kamailio.org http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
_______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
