Hello Klaus, I had already two sockets bound each to two independent physical interfaces. I have added the force_send_socket at each rtpproxy
It is necessary to use the cwie / cwei flags in the rtpproxy_manage call? Currently audio does not flow back to the softphones, it gets lost at Kamailio. Thank you for your help > ----- Original Message ----- > From: Klaus Darilion > Sent: 01/23/14 12:26 AM > To: Kamailio (SER) - Users Mailing List > Subject: Re: [SR-Users] Kamailio behind NAT > > Am 21.01.2014 17:33, schrieb John Smith: > > The next test has been to comment out the rtpproxy_manage at NATMANAGE > > function and to put it both at route[RELAY] and onreply(route) following > > your post in this list from January > > 2013:http://lists.sip-router.org/pipermail/sr-users/2013-January/076254.html. > > > > Now the media flows from Phone1 to Kamailio, from Kamailio to Asterisk and > > back, but it gets stuck at Kamailio. I cannot see it flow towards the > > public IP of the Phone2. > > > > The force_send_socket you used could be of any use here? > That's what I recommend: > > - use 2 sockets, one for communication with internal nodes, one for > external clients > - in your Kamailio config check the direction of every message: i->e or > e->i (for requests and responses). Depending on the direction set the > proper IP when calling manage_rtpproxy and force the send socket: > > regards > Klaus > > > _______________________________________________ > 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 _______________________________________________ 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
