[OpenSIPS-Users] Relaying of RTP packets between two opensips/mediaproxy

[José María Jiménez]

Hi all,

I'm testing a VoIP architecture in order to make a call between two IP
phones in a LOCAL network environment (I'm not using any public IP) where
the packets through 2 proxies (composed by a OpenSIPS, a MediaProxy module
and a MediaProxy) and an Asterisk. After several attempts of configuring
OpenSIPS and MediaProxy, I can't achieve RTP relaying between P1, P2 and
Asterisk.

VoIP Architecture:

           192.x.x.x                172.x.x.x                172.y.x.x
+------------------------------+---------------------------+----------------------------+

S1-+
      |
      +<---SIP/RTP---> P1 <---SIP/RTP---> P2 <---SIP/RTP---> A
      |
S2 -+


   S1 = Softphone 1
   S2 = Softphone 2

   P1 = Proxy 1 ( OpenSIPS + MediaProxy )
   P2 = Proxy 2 ( OpenSIPS + MediaProxy )

   A = Asterisk

SIP traffic works correctly:
- The phones are registred (REGISTER) in Asterisk. OpenSIPS 1 and 2 only
relay the SIP packets.
- The caller sends the initial "INVITE" to P1, P1 to P2 and P2 to Asterisk.

When I make a call, signaling works correctly but audio (RTP) doesn't. The
phones send their RTP packets to Proxy1, But P1 is unable to forward them to
P2. I know it can be due to NAT problems, but I still have some doubts:

1) About the NAT problem, Does it affect to local networks? All elements of
this architecture are in differents local networks (phones in 192.x.x.x,
proxies in 172.x.x.x, asterisk in 172.y.x.x) and every element of the
solution knows how to routing IP packets from one network to another, so...
is NAT affecting to this VoIP architecture?

2) Another question... It's possible relaying RTP traffic from one
MediaProxy directly to another, right?

Thanks in advance for your help :) I have read a lot about NAT but I still
don't understand if this affects to my VoIP architecture if I work just with
private IPs.

José M.
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