RTPProxy is able to fork the RTPStream in Realtime to a new destination.<br><br>I guess it would be possible to implement a module (or enhance some one existent, i think could be a B2B special scenario) in Opensips to create a new branch to a listen destination and interact with RTPPtoxy to fork the RTP Stream of an stablished call to this new destination.<br>
<br>What do you think? This could be an interesting new feature to Opensis supports the Lawful interception requirements.<br><br>Best regards,<br><br clear="all">Antonio Anderson Souza<br>Voice Technology<br><a href="http://www.antonioams.com">http://www.antonioams.com</a><br>
<br><br><div class="gmail_quote">On Mon, May 24, 2010 at 9:15 AM, Bogdan-Andrei Iancu <span dir="ltr"><<a href="mailto:bogdan@voice-system.ro">bogdan@voice-system.ro</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin: 0pt 0pt 0pt 0.8ex; border-left: 1px solid rgb(204, 204, 204); padding-left: 1ex;">
Hi Nehru,<br>
<br>
An idea will be to have rtpproxy in the calls (the media will be relayed<br>
by rtpproxy). And rtpproxy provide the "record" option, that will save<br>
the RTP stream into a local file - after the call is completed, you can<br>
take the RTP files and listen them.<br>
<br>
Regards,<br>
<font color="#888888">Bogdan<br>
</font><div class="im"><br>
<br>
Indiver wrote:<br>
> Hi every one,<br>
><br>
> We got an requirement regarding call spying.I want to know whether call<br>
> spying is possible in opensips. I heard that in asterisk they have given<br>
> option for silent listening of conversations between two parties with out<br>
> knowing the two parties.<br>
><br>
> Regards,<br>
> Nehru.<br>
><br>
<br>
<br>
--<br>
</div><div class="im">Bogdan-Andrei Iancu<br>
<a href="http://www.voice-system.ro" target="_blank">www.voice-system.ro</a><br>
<br>
<br>
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