OOps, made a mistake on tipping.....take a look down please... 2011/1/26 Danny Dias <ing.diasda...@gmail.com>
> Many thanks Jaremya, > > 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 > > i mean.... > > signaling: A---->PROXY---->B (the normal procedure) > > 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 :() > > A ---INVITE---> SIP_PROXY ---INVITE---> SIP_RECORDER > B ---INVITE---> SIP_RECORDER --INVITE--> SIP_RECORDER > B ---INVITE---> SIP_PROXY ---INVITE---> SIP_RECORDER > 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...) > > Should i use some function like t_replicate to send 2 invites like this: > > A --INVITE--> PROXY --INVITE--> B > . > . INVITE > . > RECORDER SERVER > > > 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.. > > 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 > > Thanks in advance!!! > > > > 2011/1/26 <r...@dimension-virtual.com> > > Danny Dias <ing.diasda...@gmail.com> escribió: >> >> Thanks Jeremya, but it's a requeriment from the client to record the >>> calls >>> through an external server and not with rtpproxys, my question is how the >>> media should be handled in order to record the conversations if the >>> server >>> is external? >>> >>> Signaling: Phone_A <---> Proxy <---> Phone_B >>> >>> Media: Phone_A <---> SIP RECORDER <---> Phone_B (Changing the SDP headers >>> to >>> send RTP to the IP of the SIP RECORDER). The main problem is that the >>> recording must be made in ACTIVE way, it means, we should record (IN+OUT) >>> in >>> Phone A, and the same in B, 2 recording for each call...the customer says >>> that it's working now in his arquitecture (its analog), and we made the >>> same >>> with the IP technology...resuming: with a sip recorder in the middle of >>> the >>> media should work right? >>> >> >> >> 2 ways of doing that: >> >> a) >> Signaling: A <-> Proxy <-> B2BUA (recorder) <-> B >> Media: A <-> B2BUA <-> B >> >> b) Prefered way >> Signaling: A <-> Proxy <-> B >> Media: A<-> RTPPROXY <-> B >> >> At the end, both solutions are THE SAME, what you do is to tell A to send >> media to the B2BUA or the RTPPRoxy. >> >> As a matters of scale ... b) solution is the best one. >> >> Also, another things to take into account are: >> >> 1- Transcoding issues (RTPPRoxy does not do transconding, not easly) >> 2- Secured RTP (ZRTP, SRTP, etc.) >> 3- LAG in audio. >> >> >> >> ---------------------------------------------------------------- >> This message was sent using IMP, the Internet Messaging Program. >> >> >> _______________________________________________ >> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list >> sr-users@lists.sip-router.org >> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >> > > > > -- Ing. Danny Dias www.DannTEL.net
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