Dear OpenSIPS-users,
First of all, happy new year.
I'm still reading about NATs, SIP proxies and SIP, to understand how to solve a question in my project. I'm not expert in TCP, IP, routers, NATs and networks. That is why I would like to get some help here.  I guess I'm almost getting the point to solve it.
In my current network topology case I have:
                                                                                                 Mobile Phone 1 on Internet
                                                                                                               / \                 |
                                                                                                                |                  |
                                                                                                                |      SIP     |
                                                                                                                |                  \/                 Real IP  (Internet)
                                                                                 NAT  ------------------------------------------------
                                                                                                                /\                 |                  'Nated' IP (Wan)
                                                                                                                |                  |
                                                                                                                |     SIP       |
                                                                                                                |                  |
                                                                                                                |                  |
                                                                                                                |                  \/
                                                                       ROUTER   ------------------------------------------------
                                                                                                                /\                 |                  (Lan)
                                                                                                                |       SIP     |
                                                                                                                |                  |
                                                                                                      SIP Proxy OPENSIPS
                                                                                                                |                  |
                                                                                                                |       SIP    |
                                                                                                                |                  |
                                                                                                Mobile Phone 2 on local network
When Phone 2 calls Phone 1, everything is ok. Phone 1 is registered on OpenSIPS with 'Nated' IP, by someway. So, Phone 2 sends INVITE to such IP. When Phone 1 answers with SIP OK, the Contact header filed has the same 'Nated' IP. Then, Phone 2 can send the ACK to the correct path.
However, when Phone 1 calls Phone 2 and Phone 2 answer with SIP OK, Phone 1 can't send the ACK, because  the Contact header field from SIP OK has the local IP for Phone 2.
It is true because when Phone 2 registers itself on OpenSIPS, the local IP is recorded in the local database table.
So I have the following questions:
1 - Can this problem be solved without using the Nat Traversal Module?
2 - Can NATHELPER module fix the Contact header field (as a I saw this module has functions related with it), when Phone 2 sends SIP OK to Phone 1?  If yes, how can such module determine a 'good' IP to put in this header field?
3 - If Phone 2 uses a stun server, will such phone be registered on OpenSIP with a real IP, won't be? Could it be a solution too?
Any hint will be very helpful!
Thanks a lot!
RODRIGO PIMENTA CARVALHO
Inatel Competence Center
Software
Ph: +55 35 3471 9200 RAMAL 979
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.opensips.org/pipermail/users/attachments/20160106/e6d1e5ff/attachment-0001.htm>