[OpenSIPS-Users] How to bill SIP session time correctly?

[Bogdan-Andrei Iancu]

Hi Alejandro,

When comes to billing, for Voip, you have two directions:
 
1) relay on call data from RTP level - this part was alrready covered, 
so I will not detail

2) relay on call data from SIP level - here you have several mechanism 
to help you. But first of all, you need to understand that doing ACC 
based on SIP signalling may not provide accuracy in case of errors - 
because there is no continues check at signalling level to detect the 
ghost calls.
Assuming you use the SIP Session Timer mechanism with a 5 minutes 
interval, it means you will have a 0-to-5 mins error in detecting a 
broken call.

Right now I'm considering enhancing the dialog module to be able to do 
SIP pinging (OPTIONS, re-INVITE) from opensips side, in both caller + 
callee direction, to check the call health - this will eliminate any 
dependency to the caller/callee device (like you have on the SST mechanism).


What is important to understand : there are various mechanism available 
to check the ACC accuracy, but at the end it is a compromise between the 
effort/cost to run a perfect ACC system and the losses due an inaccurate 
system - like if bandwidth required to do RTP relay or SIP probing (to 
detect the ghost calls) is more expensive than the losses because you 
had 10 ghost calls per month ....

Regards,
Bogdan

Alejandro Recarey wrote:
> Hi all, and thanks for taking the time to read my mail.
>
> I am currently studying OpenSIPS to replace Asterisk in a network that
> I administer. I am doing this because Asterisk call quality quickly
> starts degrading once you hit 100 simultaneous calls.
>
> Although NAT issues are not terribly important (most of my calls are
> routed directly to VoIP carriers) I am routing RTP through my asterisk
> boxes, this allows me to bill call length without making errors.
>
> When using a pure SIP solution like OpenSIPS, and session timers are
> not enough, how do you bill your customers?
>
> I have seen that one solution is to use MediaProxy or RTP proxy to
> proxy the RTP stream and inform OpenSIPS when the RTP stream
> terminates. Won't this have the same scalability problems as Asterisk?
> Is it a robust solution?
>
> Do any of you have experience using OpenSIPS paired with MediaProxy or
> RTPproxy to correct call times? Does this have a large impact on the
> scalability of the solution?
>
> Thanks for the replies!
>
> Alex
>
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>   


-- 
Bogdan-Andrei Iancu
OpenSIPS Bootcamp
20 - 24 September 2010, Frankfurt, Germany
www.voice-system.ro