[OpenSIPS-Users] NAT handling for internally generated messages in local_route

[Rodrigo Pimenta Carvalho]

I was right now reading about nat traversal.

I think it will be important for you too: http://www.opensips.org/html/docs/modules/2.2.x/nat_traversal.html

NAT Traversal Module - OpenSIPS<http://www.opensips.org/html/docs/modules/2.2.x/nat_traversal.html>
www.opensips.org
The nat_traversal module implements a very sophisticated keepalive mechanism, that is able to handle the most complex environments and use cases ...

If can help you keep the NAT mappings valid during calls, until SIP BYE is sent.


Best regards.


RODRIGO PIMENTA CARVALHO
Inatel Competence Center
Software
Ph: +55 35 3471 9200 RAMAL 979


________________________________
De: users-bounces at lists.opensips.org <users-bounces at lists.opensips.org> em nome de Husnain Taseer <husnain.taseer at gmail.com>
Enviado: sexta-feira, 5 de fevereiro de 2016 15:06
Para: users at lists.opensips.org
Assunto: [OpenSIPS-Users] NAT handling for internally generated messages in local_route

Dear Users,
I am facing the same issue as discussed in the below thread few years ago.

http://opensips.org/pipermail/users/2009-March/003648.html

I am setting $DLG_timeout for every call so after this amount of seconds BYE is generated by TM module and sent to both caller and callee. But if callee or caller are behind NAT then the contact IP address (in their respective contact fields in dialog table) is private because of which opensips is sending BYE to their private addresses which is not reaching them and call don't disconnects. Is there any solution available to handle this issue.

Regards,
Husnain Taseer
VoIP Developer
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