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Hi Tito,<br>
<br>
I am curious about your situation because I am in the early stages
of doing call recording as well. I have some questions if you do
not mind.<br>
<br>
Are you doing DTLS-SRTP and or SDES-SRTP, or are you simply using
ICE and rtpengine is a turn relay? If you are not doing DTLS or
ICE, perhaps you should look into rtpproxy which does have a
recording feature. If you are doing DTLS, is your recording server
DTLS capable - and if so would you mind telling me what you are
using for a recording server?<br>
<br>
I am also curious how you are detecting a recording request - is it
via DTMF, or some external signalling mechanism like clicking record
on a web interface? If you are doing DTMF detection, I'd like to
know how you are doing it.<br>
<br>
-Eric<br>
<br>
What are you using as the recording server - also are you doing DTLS
with rtpengine or just ICE, I would also like to know how you plan
to trigger the detection of a recording request - are you <br>
<br>
<div class="moz-cite-prefix">On 09/08/2015 12:27 PM, Tito Cumpen
wrote:<br>
</div>
<blockquote
cite="mid:CANZPVB4m4-ZwBs-u+pnKmkjkcSDk-uELtR1Twi4RKN1V3zJzPg@mail.gmail.com"
type="cite">
<div dir="ltr">Bogdan,
<div><br>
</div>
<div>Thanks for your reply and questions. Currently call flows
are using ICE and rtpengine as a turn relay and so there's
nothing in between . In the case I get a request to begin
recording I'd like to move the active call to a media server
that bridges the call making it appear seamless for the caller
and callee. If I trigger a RE-INVITE to both A and B with the
media server address this should work but I am not sure how I
can use opensips to send a blank invite on behalf of both A
and B utilizing the same call id to media server then
utilizing the reply as the RE-INVITE to A and B. In essence
putting the media server in between without forcing a hang
up. </div>
<div><br>
</div>
<div>Thanks,</div>
<div>Tito</div>
</div>
<div class="gmail_extra"><br>
<div class="gmail_quote">On Mon, Sep 7, 2015 at 6:20 AM,
Bogdan-Andrei Iancu <span dir="ltr"><<a
moz-do-not-send="true" href="mailto:bogdan@opensips.org"
target="_blank"><a class="moz-txt-link-abbreviated" href="mailto:bogdan@opensips.org">bogdan@opensips.org</a></a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0
.8ex;border-left:1px #ccc solid;padding-left:1ex">
<div text="#000000" bgcolor="#FFFFFF"> <tt>Hi Tito,<br>
<br>
Do you want to move on the call legs to the call
recording server (like to a VM or so) or while A talks
to B, you want to have something in the middle to record
the call between those two parties ?<br>
<br>
Best regards,<br>
</tt>
<pre cols="72">Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
<a moz-do-not-send="true" href="http://www.opensips-solutions.com" target="_blank">http://www.opensips-solutions.com</a></pre>
<div>
<div class="h5">
<div>On 03.09.2015 01:13, Tito Cumpen wrote:<br>
</div>
</div>
</div>
<blockquote type="cite">
<div>
<div class="h5">
<div dir="ltr">Group,
<div><br>
</div>
<div>Has anyone had experience reinviting an
ongoing session between two sip clients to a sip
capable media server for call recording purposes
without dropping the ongoing call? Is the best
practice to use XML_RPCNG/fifo command and have
opensips interact as 3rd party call control. Or
would the 3rd party entity need to hijack the
ongoing session as pose as the remote party. I
have a requirement to record video and audio
legs. The media server is capable for recording
these streams just need to find a way to do this
without dropping the call.</div>
<div><br>
</div>
<div><br>
</div>
<div>Thanks,<br>
Tito</div>
</div>
<br>
<fieldset></fieldset>
<br>
</div>
</div>
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