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

[Brett Nemeroff]

On Thu, Jul 29, 2010 at 9:25 AM, Alejandro Recarey <alexrecarey at gmail.com>wrote:

> When using a pure SIP solution like OpenSIPS, and session timers are
> not enough, how do you bill your customers?
>
> I have done a number of configurations where calls are billed solely on
signalling. Before everyone jumps on my back here, I'll give you the same
disclaimer I give everyone. Billing records based on proxy records are not
authoritative and it's very important you understand the limitations and
potential security risks before even considering using billing records from
a proxy. For what it's worth, in a practical setting, I get very good
results usually.


> 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?
>
>
Using mediaproxy + opensips can absolutely provide an authoritative call
record as it *knows* how long the RTP lasted, which is what's really
important. There will be some correlation required on your behalf to make
this work. If I remember correctly, mediaproxy makes a JSON record of each
call as it terminates in media_sessions. This record has really fantastic
information about the call.

Mediaproxy scales much better than Asterisk. Keep in mind that Asterisk's
scalability really has more to do with the way it was put together and not
so much just on the architecture. In other words, it wasn't really built to
RTP proxy 100 calls at once. I'm sure there are plenty of you out there that
will disagree with me on that; I always end up hearing someone say that they
ran 10,000 calls and never noticed a problem :)  Mediaproxy on the other
hand I believe had been tested pretty successfully up to
 2000 simultaneous calls on a single box. Of course, just like Asterisk,
these numbers are *meaningless* without factoring in your application and
the specific hardware you are using. As always, you'll need to do your own
benchmark, but I feel pretty confident in saying you'll see a *huge*
performance improvement using mediaproxy + opensips over asterisk. Of
course, you give up a lot of functionality, but you might not be using much
of that anyway.

Billing SIP calls is like religion; and this is what I believe.

-Brett
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