Hi all,<br><br>I'm testing the integration of OpenSIPS + RTPproxy and I have some doubts:<br><br>1) Is it possible to test a VoIP infrastructure (with OpenSIPS, RTPProxy and IP-phones) in a local network? I'm working on the local network of an university and it's difficult to me to work with an public IP.<br>
<br>2) Is it possible to chain directly 2 OpenSIPS+RTPProxy solutions (in a local network)?<br><br>IP-Phones <----- sip/rtp -----> OpenSIPS/RTPProxy <----- sip/rtp -----> OpenSIPS/RTPProxy <----- sip/rtp -----> Asterisk<br>
<br>I'm getting some problems in the SDP body of the second OpenSIPS/RTPProxy. I don't know if this wrong behavior is due to the fact I'm working in a local network and functions liks fix_nated_sdp doesn't work in this context. Can I modify the SDP body directly with functions like subst_body() (from nathelper module)?<br>
<br>Thanks in advance!<br><br>José M.<br><br><div style="margin: 2em 0pt;" name="sig_93ca30fa1b"><table width="336" border="0" cellpadding="0" cellspacing="0">
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<td height="80" valign="top" width="233"><font color="#4e4a49" face="Arial" size="2"><strong>José María Jiménez</strong><br>
R&D Engineer <br>
<br>
</font> <font color="#4e4a49" face="Arial" size="1">Grupo Gesfor R&D.<br>
<a href="http://innovacion.grupogesfor.com">innovacion.grupogesfor.com</a></font></td>
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