I beg to differ, but this is just my humble opinion based on my
experience with my particular customers.
The most economic and future-proof way to perform accounting for SIP
sessions is the SIP Proxy server alone.
My personal experience is that gateways come and go in a provider
configuration and they are in many cases under the control of a third-
party that provides the PSTN termination service. When you do LCR
across many different gateways, which are not even yours the only
aggregation point for traffic is the SIP proxy that authenticates and
authorizes the requests. Over time, the gateways change hands, get
upgraded or removed much more often then the proxy itself, which
maintains its central role over time. Secondly, once you do more the
voice like video and other services that require billing and are not
PSTN related, the SIP Proxy is the only network element that has
access to the signalling and is able to generate accounting tickets.
Solving the accounting related problems at the SIP Proxy level is a
worthwhile investment while other options are just temporary fixes
that work in a particular case for a limited amount of time and that
is a waste of money.
Adrian
On Jan 7, 2009, at 2:25 AM, Jiri Kuthan wrote:
authentication does not provide actually value here. dialog would not
either, since
the same trick can be achieved for example by low max-forwards. IMO
the
proper
choice is accounting from the gateway, which provides the actual
service.
A proxy can only provide an approximation which is inherentely to some
extent
more error-prone than the box doing the actual job.
-jiri
Bogdan-Andrei Iancu wrote:
Hi Iñaki,
Have you consider requesting auth for the BYE ? from SIP point of
view
is perfectly valid....
Regards,
Bogdan
Iñaki Baz Castillo wrote:
Hi, I'm thinking in the following flow in which the caller/attacker
would get an unlimited call (but a limited CDR duration):
--------------------------------------------------------------------------
attacker OpenSIPS (Acc)
gateway
INVITE (CSeq 12) ------>
<-------- 407 Proxy Auth
INVITE (CSeq 13) ------>
INVITE (CSeq 13)
------>
<-------------------
200 Ok
<------------------- 200 Ok
<< Acc START >>
ACK (CSeq 13) ----------->
ACK (CSeq 13)
----------->
<******************* RTP ************************>
# Fraudulent BYE !!!
BYE (CSeq 10) ----------->
<< Acc STOP >>
BYE (CSeq 10)
----------->
<-- 500 Req Out of
Order
<-- 500 Req Out of Order
--------------------------------------------------------------------------
The call hasn't finished, but OpenSIPS has ended the accounting for
this call since it received a BYE. And this BYE will generate a
correct ACC Stop action (since it matches From_tag, To_tag and
Call-ID).
I think this is *VERY* dangerous and I hope I'm wrong.
Would help the dialog module here? does the dialog module check the
CSeq of the BYE in some way and could it prevent OpenSIPS from
generating the ACC STOP action? (I don't think so).
Any idea?
_______________________________________________
Users mailing list
[email protected]
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
_______________________________________________
Users mailing list
[email protected]
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
_______________________________________________
Users mailing list
[email protected]
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
