[OpenSIPS-Users] RTPPRoxy behind NAT

[SamyGo]

Hi,

Do you've any SIP traces captured on your OpenSIPS server for the calls
with No-Audio ? attach a sip trace and I'll see what I can find.  There is
only one option that I can assume at the moment (Your firewall/router is
perfectly working I assume) There maybe some NAT condition which breaks
this setup and your audio from both ends gets lost.

i.e


*User------------------YourFirewall-------------OpenSIPS*
192.168.1.5<------>  WAN/LAN <---------->10.1.2.103
SIP<--------------------><NAT><-------------->SIP (Perfectly here)
RTP
 RTP(trying to send to 192.168.1.5 in LAN)
(trying to send to 10.1.2.103 in LAN)

But thats just again a rough sketch. Do share the SIP tarces.


TC
Sammy
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