[OpenSIPS-Users] Restrict Simultaneous-Use

[Bogdan-Andrei Iancu]

It does help, but it is still not accurate. you can reduce the interval 
for re-INVITEs, but you overload the proxy and network -> smallest error 
in detecting errs; if you use big intervals for re-INVITEs, you get less 
load, but the error gets higher.

For accounting the BYEs generated by dialog module, use local_route.

Regards,
Bogdan

Brett Nemeroff wrote:
> Question, do SST help the situation? Is it widely accepted enough in 
> the protocol to provide some backup mechanism to maintaining dialog 
> state in the event of a lost BYE.
>
> BTW, how is a BYE accounted for in ACC when generated locally because 
> of an expired dialog?
>
>
> On Thu, Mar 5, 2009 at 10:00 AM, Bogdan-Andrei Iancu 
> <bogdan at voice-system.ro <mailto:bogdan at voice-system.ro>> wrote:
>
>     Iñaki Baz Castillo wrote:
>     > 2009/3/5 Bogdan-Andrei Iancu <bogdan at voice-system.ro
>     <mailto:bogdan at voice-system.ro>>:
>     >
>     >> Hi Inaki,
>     >>
>     >> This is an old, hot topic.
>     >>
>     >> There are many services that are more appropriate via a B2BUA
>      (like acc,
>     >> dialog stuff, security, etc) - last time the discussion started
>     from the
>     >> question if a proxy is the best place to do accounting.
>     >>
>     >> In all the case is about compromising - how much you are
>     willing to lose.
>     >> You may loose time/resources to build and implement a platform
>     were there is
>     >> no way for bad thinks to happen (you deal with all corner
>     cases) - and you
>     >> end up with a huge platform, very complex, difficult to
>     maintain, expensive
>     >> to run, etc .
>     >>
>     >> Or you can loose some corner cases and build a simpler and more
>     efficient
>     >> platform.
>     >>
>     >
>     > Yes Bogdan, also the approach described in the wiki is really
>     > appropiate and valid when the caller or callee is a PSTN gateway
>     which
>     > sends a BYE if RTP is lost.
>     >
>     > But as always, pure "Internet" calls (between users) are really
>     > difficult to monitorize and control by a proxy.
>     >
>
>     I agree on this. But this is SIP - from signalling level only you
>     do not
>     have information about the call status (during the call).
>
>     Regards,
>     Bogdan
>
>     _______________________________________________
>     Users mailing list
>     Users at lists.opensips.org <mailto:Users at lists.opensips.org>
>     http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>