Re: [SR-Users] NAT Traversal Problem
Hi, To enable nat traversal execute RTPProxy. http://www.rtpproxy.org It supports many thousand calls -- Laurent PIERRE On 18 April 2013 16:04, Ishan Sawhney wrote: > Hi, > > We have a solution which requires only outbound NAT on SIP calls. > > Do you have a product which would solve the NAT Traversal problem on the > outbound SIP calls? > If yes, how much would be the cost of this product? > Will it support 2000 simultaneous calls? > > BR// > Ishan > > ___ > 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
Re: [SR-Users] NAT Traversal Problem
will it be work on NAT Thanks Jagan On 19 April 2013 08:40, PIERRE Laurent wrote: > Hi, > > To enable nat traversal execute RTPProxy. > > http://www.rtpproxy.org > > It supports many thousand calls > > > -- > Laurent PIERRE > > > On 18 April 2013 16:04, Ishan Sawhney wrote: > > Hi, > > > > We have a solution which requires only outbound NAT on SIP calls. > > > > Do you have a product which would solve the NAT Traversal problem on the > > outbound SIP calls? > > If yes, how much would be the cost of this product? > > Will it support 2000 simultaneous calls? > > > > BR// > > Ishan > > > > ___ > > 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 > > ___ 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
Re: [SR-Users] NAT traversal problem with Kamailio when trying to communicate between two different LANs
Hi All, I installed Kamailio server on my ubuntu machine. And installed rtpproxy also in the same machine. I didm't make many changes to default configuration files, kamctlrc and kamailio.cfg. Now I am able to communicate between two SIP clients, if both sip clients are in the same LAN. But I am not able to communicate between one sip client in one LAN and the other SIP client in other LAN. Sip Call handling and ringing etc. is ok but there is no voice for this case. I don't have much experience in routing for NAT traversal. I tried to experiment with rtpproxy_manage function in kamailio.cfg file. But I was not successful. Please help me by providing any hints or pointers to proceed further. I am using only Kamailio and Rtpproxy. No other software like Asterisk or FreeSwitch. * These are defines I used in kamailio.cfg file: #!define WITH_MYSQL #!define WITH_AUTH #!define WITH_ALIASDB #!define WITH_USRLOCDB #!define WITH_ANTIFLOOD #!define WITH_NAT #!define WITH_PRESENCE listen ip address changed *** Kamailio is compiled with following modules: include_modules= db_mysql dialplan presence presence_xml Regards, Sateesh ___ 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
Re: [SR-Users] NAT traversal problem with Kamailio when trying to communicate between two different LANs
Did you try this; http://www.palner.com/blog/303/kamailio-behind-nat/#more-303 Regards, Arun From: Neo Quartz To: sr-users@lists.sip-router.org Sent: Wednesday, February 19, 2014 8:22 AM Subject: Re: [SR-Users] NAT traversal problem with Kamailio when trying to communicate between two different LANs Hi All, I installed Kamailio server on my ubuntu machine. And installed rtpproxy also in the same machine. I didm't make many changes to default configuration files, kamctlrc and kamailio.cfg. Now I am able to communicate between two SIP clients, if both sip clients are in the same LAN. But I am not able to communicate between one sip client in one LAN and the other SIP client in other LAN. Sip Call handling and ringing etc. is ok but there is no voice for this case. I don't have much experience in routing for NAT traversal. I tried to experiment with rtpproxy_manage function in kamailio.cfg file. But I was not successful. Please help me by providing any hints or pointers to proceed further. I am using only Kamailio and Rtpproxy. No other software like Asterisk or FreeSwitch. * These are defines I used in kamailio.cfg file: #!define WITH_MYSQL #!define WITH_AUTH #!define WITH_ALIASDB #!define WITH_USRLOCDB #!define WITH_ANTIFLOOD #!define WITH_NAT #!define WITH_PRESENCE listen ip address changed *** Kamailio is compiled with following modules: include_modules= db_mysql dialplan presence presence_xml Regards, Sateesh ___ 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