27 Jan
2011
27 Jan
'11
2:51 p.m.
o rabs(a)dimension-virtual.com on 01/27/2011 01:13 PM:
Danny Dias <ing.diasdanny(a)gmail.com> escribió:
no, I
think what he wants is in the proxy to intercept the call which
is from UAC A to UAC B, and place two additional calls to his
SIP-RECORDER, which should transmit both signaling and RTP to the
SIP-RECORDER.
To the OP: this is IMO not possible to do properly in a SIP proxy,
even if its as flexible as s-r; while you can replicate the initial
INVITE to the SIP-RECORDER, and even change the RTP media address in
the SDP in order to make RTP go through rtpproxy, which might be
modified to send a copy of the traffic to your SIP recorder, I really
doubt that you can change the 200 to establish another INVITE to the
SIP-RECORDER, let alone that you need to process the responses to the
INVITEs to SIp-RECORDER, generate ACKs, handle in-dialog requests
(e.g. session timer, hold etc etc).
For this stuff you need a SIP app server/B2BUA/media server, which has
two dialogs on each side, and another two dialogs to the SIP-RECORDER.
You might possibly be interested in SEMS' sbc module (shameless ad ;),
latest development features RTP relay, you might be able to change it
to create two more dialogs to SIP-RECORDER, using the original
signaling, and relay the RTP also there. If that doesn't suit you,
possibly you could have a look at implementing this using pjsip,
freeswitch, asterisk (maybe in that order, depending on how
transparent the thing in the middle should be). An alternative is to
use s-r proxy and rtpproxy, and add some custom app server (again
built possibly with sems/pjsip/freeswitch etc), which somehow gets
signaling and RTP traffic from s-r/rtpproxy and sends them into two
separate calls which are established to SIP-RECORDER. This type of
stuff is usually implemented for LI (which hopefully we all don't
like) and has further requirements regarding being transparent etc.
hth
Stefan
--
Stefan Sayer
VoIP Services Consulting and Development
Warschauer Str. 24
10243 Berlin
tel:+491621366449
sip:sayer@iptel.org
email/xmpp:stefan.sayer@gmail.com
Hi Iñaki,
2011/1/27 Iñaki Baz Castillo <ibc(a)aliax.net>
So ... if each UAC send an INVITE to the SIP-RECODER ... where is the
problem? ... the RTP will be from UAC A to SIP-RECORDER .. that's all. 2011/1/26 Danny Dias
<ing.diasdanny(a)gmail.com>om>:
B ---INVITE---> SIP_PROXY --INVITE--> SIP_RECORDER
Each peer must send an INVITE to the sip_recorder server, to stablish a
session with it... 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