[OpenSIPS-Users] rtp and fake hangup

[mancyborg at gmail.com]

Very interesting, thank you very much Bogdan and Dani :)



On Thu, 06 Jan 2011 19:24:54 +0200
Bogdan-Andrei Iancu <bogdan at voice-system.ro> wrote:

> Hi Mike,
> 
> it is not the only one. You can detect such ghost calls also at 
> signalling level (via SST).
> 
> But detection at media level:
>     - is the most accurate as time error
>     - it adds the biggest load on the system (as you need to put all RTP 
> streams on your servers).
> 
> So, which one to use (signalling versus media), is about making a 
> compromise between accuracy and load (considering that such ghost calls 
> are corner cases).
> 
> Regards,
> Bogdan
> 
> mancyborg at gmail.com wrote:
> > Hi All,
> >
> > in a scenario where opensips routes calls from sip user agents to voip carriers:
> > can you please confirm that the only way to be sure to prevent fraud false hangups
> > is to force the voice (rtp) to pass through opensips ?
> >
> >
> > Thanks and regards,
> > Mike
> > _______________________________________________
> > Users mailing list
> > Users at lists.opensips.org
> > http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> >
> >   
> 
> 
> -- 
> Bogdan-Andrei Iancu
> OpenSIPS Event - expo, conf, social, bootcamp
> 2 - 4 February 2011, ITExpo, Miami,  USA
> www.voice-system.ro
> 
> 
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users