[OpenSIPS-Users] opensips-cp CDR correlation

[Dan Pascu]

On Wednesday 29 April 2009, Iñaki Baz Castillo wrote:
> No, SessionTimers just requieres one endpoint supporting them. In the
> case I described, the B2BUA (acting as UAS for phone 1 and UAC for
> phone 2) does support SessionTimers so can monitorize the dialog
> status in both legs.

You realize that a B2BUA completely separates the 2 call legs, which 
includes audio. If you only attempt to split the signaling path, but leave 
the media path to flow directly, you may have trouble with NAT. So if you 
replace a media relay with a B2BUA which does media transcoding, I do not 
believe you are any better.

> Imagine a company using a hosted virtual PBX solution (the
> proxy/SA/B2BUA has public IP while the phones are behind NAT).
> Imagine the boss wishing to have an accurated log (cdr) of how long
> his employers are speaking between them. These calls take place,
> probably, into the same LAN so a media-proxy is not required to allow
> bidirectional audio. 

It may not be required, but it can be used to get some additional 
benefits, even more in this particular example, considering that LAN 
traffic is cheap.

> Also, forcing a media-proxy could consume high bandwitch and create
> audio delay.

This is actually incorrect. A media relay should be positioned next to the 
PSTN gateway, so that the network will not see more traffic than it would 
have seen if the media streams would have reached the gateway directly.

> It shouldn't be required just to get accurate accounting.

I do not recall anyone making such a statement. Using a media relay is 
just one solution to avoid issues with signaling or problems caused by 
malicious user activity. You're free to consider it or not.

All your examples where the system could be circumvented are based on the 
fact that someone was attempting to fool the SIP proxy to consider the 
session has ended by transmitting doctored SIP messages, while at the same 
time trying to preserve the media stream and keep talking. If any attempt 
to terminate the SIP signaling will also terminate the media, the users 
will have no incentive to try to circumvent the SIP signaling, because 
they will be left without the desired fruit of their work (the free media 
stream). Like it or not, what defines a voice call is the presence of an 
active media stream, not some sporadic circumventable SIP signaling. A 
user can try to fool the later, but cannot have a call without the former.

A smart entrepreneur, will attempt to take away the incentive to 
circumvent the system, like for example using a flat fee billing scheme. 
That is a much desirable and cheaper thing to do, than to attempt to put 
layer after layer of protection that will be ultimately circumvented 
sooner or later anyway. If one keeps promoting a model based on scarcity 
and high prices, the incentive to hack the system will always be there.

-- 
Dan
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.opensips.org/pipermail/users/attachments/20090429/ebbe49d1/attachment.htm