Hi Ryan, try it again. Kamailio itself didnt care about rtp (audio). So you must setup an multihomed kamailio and rtpproxy in bridging mode between the networks.
Then kamailio calls rtpproxy_manage to rewrite the sdp and the rtp traffic goes thru rtpproxy. 2012/4/4 Ryan Gholam <ryangho...@gmail.com>: > Hello , > > i am facing an issue concerning kamailio where i am trying to connect > the kamailio to the asterisk using a private ip and the kamailio is > connected using a public ip for the clients .I am trying to create a > call from the client the phone rings but there is no audio > conversation happening , what is the best method to use ? > > > N.B : i have tried RTPproxy but it didnt work i think because their > is no proper NATING . > > > Thank you > > _______________________________________________ > 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 -- Mit freundlichen Grüßen *Karsten Horsmann* _______________________________________________ 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
