[SR-Users] SIP Recorder

[Danny Dias]

OOps, made a mistake on tipping.....take a look down please...

2011/1/26 Danny Dias <ing.diasdanny at 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 <rabs at dimension-virtual.com>
>
> Danny Dias <ing.diasdanny at 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.
>>
>>
>>
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>>
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>
>
>
>


-- 
Ing. Danny Dias
www.DannTEL.net
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