[SR-Users] Kamalio + RTP Engine + SIP Client behind NAT

[Sergio Charrua]

Olá Nuno,

I've similar setup, and everything works fine. I can share the setup, if
you wish.
Contact me directly.

Cheers,


*Sérgio Charrua*

*www.voip.pt <http://www.voip.pt/>*
Mobile: +351 91  <callto:+351+91+104+12+66>631 11 44

Email : *sergio.charrua at voip.pt <sergio.charrua at voip.pt>*
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On Wed, Jan 15, 2020 at 6:49 PM Nuno Miguel Reis <nmreis at student.dei.uc.pt>
wrote:

> Hi guys.
>
> I'm replacing a environment where I was using kamailio + freeswitch by
> another where I'm adding rtpengine to the mix.
> One of the issues I'm having now is when I have a SIP Client behind NAT:
>
> When I send the INVITE from the SIP Client, the SDP is using the private
> LAN IP address + the advertised media port, let's admit it's
> 10.10.10.10:5000 when the call establishes with a public server running
> kamailio + rtpengine, the actual RTP arrives from the home router public IP
> on a natted port, let's admit it's 100.100.100.100:65100, event though,
> RTPENGINE assumes that RTP stream is coming from 100.100.100.100:5000.
> I'm using the '--sip-source' with RTPENGINE to make it use the received ip
> address instead of the private IP coming in the SDP but I'm not finding
> anything to make RTPENGINE adapt an start sending the RTP FLOW to the port
> where he starts receiving the RTP flow from, discarding the SDP media port
> information.
>
> In my previous setup with freeswitch alone, freeswitch did this
> transparently. How do you guys solve this issue in RTPENGINE?
>
> Regards,
> --
> Nuno Miguel Reis
> Departamento de Engenharia Informática
> Faculdade de Ciências e Tecnologia
> Universidade de Coimbra
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