<div dir="ltr"><div>Eric,</div><div><br></div><div><br></div><div>I am going to be using the dialogic XMS. I believe it handles SAVPF(DTLS). The request to record will be triggered by an api. Below is a diagram of what I intend to do.</div><div><br></div><div><br></div><div>Bogdan,</div><div><br></div><div>I'd like to know if I can trigger a reinvite via the MI_http interface which will enact a b2bua scenario with the intention to move both legs to the media server. Below is a diagram. One detail I'd like to point out is the blank reinvite needs to source from opensips as it carries all the headers. Please advise if this is possible or if I can do anything aside from using a b2bua scenario.</div><div><br></div><div><br></div><div><br></div><div><br></div><div><br></div><div><br></div><img src="cid:ii_14fb40241dd75f5b" alt="Inline image 1" width="405" height="524"><br><div><br></div><div><br></div><div><br></div></div><div class="gmail_extra"><br><div class="gmail_quote">On Wed, Sep 9, 2015 at 6:09 AM, Bogdan-Andrei Iancu <span dir="ltr"><<a href="mailto:bogdan@opensips.org" target="_blank">bogdan@opensips.org</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>
A SIP call can have only 2 end-points. What is not clear for me
is: after inserting the media server, what is the final
configuration in terms of who's talking to who? Still A talks to
B, but media server is recording ? or A talks to media server
(like VM) and B drops out ?<br>
<br>
Best regards,<br>
</tt><span class="">
<pre cols="72">Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
<a href="http://www.opensips-solutions.com" target="_blank">http://www.opensips-solutions.com</a></pre>
</span><div><div class="h5"><div>On 08.09.2015 21:27, Tito Cumpen wrote:<br>
</div>
<blockquote 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 href="mailto:bogdan@opensips.org" target="_blank">bogdan@opensips.org</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 href="http://www.opensips-solutions.com" target="_blank">http://www.opensips-solutions.com</a></pre>
<div>
<div>
<div>On 03.09.2015 01:13, Tito Cumpen wrote:<br>
</div>
</div>
</div>
<blockquote type="cite">
<div>
<div>
<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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