<div>Many thanks Jaremya,</div><div><br></div><div>The main problem is that both terminals, SHALL (required and must not be changed, because of standards of EUROCAE ED-137 Part3) initiate a session with the recorder server (a commercial one, can't use Asterisk for my disgrace) sending INVITE and receiving the subsequent responses from sip recording server to stablish the session with it...after this, when the media starts to go directly peer to peer (the normal call), the terminals (specials ones) must summarize the IN+OUT audio to the recording server and through rtsp the media should be recorded...it's weird, but thats the requirement :S</div>
<div><br></div><div>i mean....</div><div><br></div><div>signaling: A---->PROXY---->B (the normal procedure)</div><div><br></div><div>At the same time, this must be done: (I'm not sure how to do this...the proxy could be out of this or not, not sure :()</div>
<div><br></div><div>A ---INVITE---> SIP_PROXY ---INVITE---> SIP_RECORDER</div><div>B ---INVITE---> SIP_RECORDER --INVITE--> SIP_RECORDER</div><div><br></div><div>Then, The audio will go directly from A to B (because of the normal procedures), and also, A and B, will summarize IN+OUT on each site and send this result through RTSP to the recording server (this is not important to the proxy righ not)...My real doubt is how to stablish the session between the peers A and B to the recording server through the Proxy and also (at the same time) continue with the normal flow of the call (invite from a to b, 200 ok viceversa etc etc...)</div>
<div><br></div><div>Should i use some function like t_replicate to send 2 invites like this:</div><div><br></div><div>A --INVITE--> PROXY --INVITE--> B</div><div> .</div><div> . INVITE</div>
<div> .</div><div> RECORDER SERVER</div><div><br></div><div><br></div><div>But the problem here is that the session between A and PROXY would be OK, but i can't see the way how B should send INVITE to the recorder server..</div>
<div><br></div><div>I hope to be clear on my problem :( and i know it looks very weird, but it's the requirement of the document mentioned above</div><div><br></div><div>Thanks in advance!!!</div><div><br></div><div><br>
</div><div><div><div><div><div><br><div class="gmail_quote">2011/1/26 <span dir="ltr"><<a href="mailto:rabs@dimension-virtual.com">rabs@dimension-virtual.com</a>></span><br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;">
<div class="im">Danny Dias <<a href="mailto:ing.diasdanny@gmail.com" target="_blank">ing.diasdanny@gmail.com</a>> escribió:<br>
<br>
</div><div class="im"><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
Thanks Jeremya, but it's a requeriment from the client to record the calls<br>
through an external server and not with rtpproxys, my question is how the<br>
media should be handled in order to record the conversations if the server<br>
is external?<br>
<br>
Signaling: Phone_A <---> Proxy <---> Phone_B<br>
<br>
Media: Phone_A <---> SIP RECORDER <---> Phone_B (Changing the SDP headers to<br>
send RTP to the IP of the SIP RECORDER). The main problem is that the<br>
recording must be made in ACTIVE way, it means, we should record (IN+OUT) in<br>
Phone A, and the same in B, 2 recording for each call...the customer says<br>
that it's working now in his arquitecture (its analog), and we made the same<br>
with the IP technology...resuming: with a sip recorder in the middle of the<br>
media should work right?<br>
</blockquote>
<br>
<br></div>
2 ways of doing that:<br>
<br>
a)<br>
Signaling: A <-> Proxy <-> B2BUA (recorder) <-> B<br>
Media: A <-> B2BUA <-> B<br>
<br>
b) Prefered way<br>
Signaling: A <-> Proxy <-> B<br>
Media: A<-> RTPPROXY <-> B<br>
<br>
At the end, both solutions are THE SAME, what you do is to tell A to send media to the B2BUA or the RTPPRoxy.<br>
<br>
As a matters of scale ... b) solution is the best one.<br>
<br>
Also, another things to take into account are:<br>
<br>
1- Transcoding issues (RTPPRoxy does not do transconding, not easly)<br>
2- Secured RTP (ZRTP, SRTP, etc.)<br>
3- LAG in audio.<div class="im"><br>
<br>
<br>
----------------------------------------------------------------<br>
This message was sent using IMP, the Internet Messaging Program.<br>
<br>
<br>
_______________________________________________<br></div><div><div></div><div class="h5">
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list<br>
<a href="mailto:sr-users@lists.sip-router.org" target="_blank">sr-users@lists.sip-router.org</a><br>
<a href="http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users" target="_blank">http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users</a><br>
</div></div></blockquote></div><br><br clear="all"><br>
</div></div></div></div></div>