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(a)voip.pt <sergio.charrua(a)voip.pt>*
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On Wed, Jan 15, 2020 at 6:49 PM Nuno Miguel Reis <nmreis(a)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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