[OpenSER-Users] MediaProxy 1.9.0 - Radius

[Jeremy McNamara]

CSB wrote:
> Can you elaborate a little on "proper call accounting" as I am battling with
> this currently. An example:
> SER sends INVITE to Asterisk. Depending on the circumstances we might want
> to record a voicemail message, hit an IVR or queue, or pass the call on to
> the PSTN. 
>
>   

Since OpenSER only manages the SIP sessions, OpenSER does not 
specifically know if any particular call is active or not.  If you were 
only relying OpenSER's accounting a crafty voip user could simply never 
send you a BYE message, thus effectively never hanging up any calls.

This is why mediaproxy has a call accounting process.    I believe RTP 
Proxy has also been updated or is scheduled to be updated with an 
accounting process.

In my opinion, the only truly viable way to do call accounting is along 
with the media stream (RTP), then you will know if the call is still in 
session and have the ability to specifically terminate the call for any 
given reason (by stopping the RTP from flowing.)




> Since Asterisk can't deal with OpenSER's authentication limitations we can
> only have one effective SIP peer (based on the IP of OpenSER) and therefore
> one context for accounting purposes. This makes even routing the call a
> challenge (how do you make sure that only certain users can get out to the
> PSTN whereas others stay internal to Asterisk). How do you get usable
> accounting records? If there are 5 calls from different users being passed
> to Asterisk they will all be accounted in the same way and it is not
> possible to bill them separately. If anyone has any advice on what I'm
> missing or how to get useful accounting records in Asterisk I would
> appreciate it.
>
>   



One option might be to send custom SIP header(s) to Asterisk.    I wrote 
my own CDR module for Asterisk, to deal with my own particular environment.





Jeremy McNamara