Hi all,<br><br>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.<br>
<br>VoIP Architecture: <br><br> 192.x.x.x
172.x.x.x 172.y.x.x<br>+------------------------------+---------------------------+----------------------------+<br><div id=":2sv" class="ii gt"><br>S1-+<br>
|<br>
+<---SIP/RTP---> P1 <---SIP/RTP---> P2 <---SIP/RTP--->
A<br> |<br>S2 -+<br><br><br> S1 = Softphone 1<br> S2 =
Softphone 2<br><br> P1 = Proxy 1 ( OpenSIPS + MediaProxy )<br>
P2 = Proxy 2 ( OpenSIPS + MediaProxy )<br><br> A = Asterisk<br><br>SIP
traffic works correctly:<br>- The phones are registred (REGISTER) in
Asterisk. OpenSIPS 1 and 2 only relay the SIP packets.<br>- The caller
sends the initial "INVITE" to P1, P1 to P2 and P2 to Asterisk.<br>
<br>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:<br>
<br>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?<br>
<br>2) Another question... It's possible relaying RTP traffic from one
MediaProxy directly to another, right? <br><br>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.<br>
<br>José M.<br><br></div><div style="margin: 0pt;" name="sig_93ca30fa1b"></div>