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

[Alejandro Recarey]

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