[OpenSER-Devel] [Serdev] Possible bug in the tm module in the presence of packet loss/branches

[Jiri Kuthan]

At 15:37 12/03/2008, Maxim Sobolev wrote:
>Dan Pascu wrote:
>> Anyway, examples are many and I'm sure people can even find more practical 
>> ones. The main reason for all of them being possible, is the fact that a 
>> CANCEL is no longer really canceling the call under these circumstances, 
>> and that the user is stripped from it's ability to control if a call 
>> should end or not before being setup, by random conditions (network 
>> packet loss) or purposeful abuses of the protocol behavior.
>
>All this has nothing to do with the SIP, really. It just illustrates the 
>point that SIP proxy is bad choice for real-time VoIP accounting.

That's a very true statement indeed. In fact a PROXy can server as an
aPROXimation of what's going on and that's it.

> If you 
>use B2BUA, all those concerns go away, as you will get two separate SIP 
>calls, so that your ingress call is isolated from any issues with 
>egress, such as packet loss and so on.

My most favorite choice is producing reliable accounting data from PSTN
gateways. In the end that's where the paid-for service is provided from,
and it can include most complete data (including Q.931 error causes if
you wish, and media QoS statistics)

-jiri


>Regards,
>-- 
>Maksym Sobolyev
>Sippy Software, Inc.
>Internet Telephony (VoIP) Experts
>T/F: +1-646-651-1110
>Web: http://www.sippysoft.com
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--
Jiri Kuthan            http://iptel.org/~jiri/