Re: [SR-Users] NAT traversal problem with Kamailio when trying to communicate between two different LANs

2014-02-20 Thread Neo Quartz
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
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Re: [SR-Users] NAT traversal problem with Kamailio when trying to communicate between two different LANs

2014-02-20 Thread arun Jayaprakash
Did you try this;

http://www.palner.com/blog/303/kamailio-behind-nat/#more-303


Regards,
Arun



 From: Neo Quartz neoqua...@gmail.com
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