Openser doesn&#39;t do a thing with RTPs. and with rewritehostport you just forward the signalling to another server.<br>On the other hand, If you DON&#39;T use rtpproxy and If you just forward all signalling to *, and * supports SRTP you should have no problem as * with setup the rtp flow as it will be * who will handle it.<br>
<br>Am I correct here? Not sure, try it!<br><br>This is mostly trial-and-error!<br><br><br><div class="gmail_quote">On Thu, Jun 19, 2008 at 9:33 AM, Iñaki Baz Castillo &lt;<a href="mailto:ibc@in.ilimit.es">ibc@in.ilimit.es</a>&gt; wrote:<br>
<blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">El Wednesday 18 June 2008 15:07:51 <a href="mailto:Martin-Gruber.External@infineon.com">Martin-Gruber.External@infineon.com</a><br>

escribió:<br>
<div class="Ih2E3d">&gt; hi guys,<br>
&gt;<br>
&gt; I have the following scenario. I want to use Asterisk as voicemailserver,<br>
&gt; but need an rtp-Proxy because the my phone speaks srtp. I get SIP-Message<br>
&gt; from an PBX to openser and RTP Proxy. My question now is, how I can handle<br>
&gt; this case or how I can forward from rtpproxy to asterisk? just with<br>
&gt; rewritehostport?<br>
<br>
</div>Hi, please, don&#39;t write in blue to a maillist, use plain text (no HTML).<br>
<br>
rewritehostport handles SIp signalling, no RTP. RtpProxy doesn&#39;t speak SIP.<br>
<div><div></div><div class="Wj3C7c"><br>
--<br>
Iñaki Baz Castillo<br>
<a href="mailto:ibc@in.ilimit.es">ibc@in.ilimit.es</a><br>
<br>
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