Thanks to Diego B, Rajesh V R and Wayne Davies for their answers.
I need to specify some questions. My role is to add a feature in a SIP network
to be able to do billing on a session.
Of course it must be generic and reusable. I understant why a simple SIP proxy
won't be a good idea. I had a look to 3CPP server, like the sun one using jain,
and I see what is missing in my simple proxy architecture.
If I understand well, you suggest this kind of architecture :
_______________ _______________
| | | |
| SIP Proxy | <--------> | B2BUA |
|_______________| |_______________|
^ ^
| \
v \
_______________ \ _______________
| | v| |
| UA A | | UA B |
|_______________| |_______________|
My "new" idea would be a SIP proxy with a B2BUA
UA A send an INVITE request, Proxy check for authentication, then INVITE is
forward to B2BUA, which send back an invite for UA B to SIP proxy if
authorization (and accouting) is ok.
When no more money is left, B2BUA sends 1 or more BYE to each UA connected
through the proxy.
That is what I understand when I look at some B2BUA architectures (Radvision
ENSEMBLE or PortaSIP, I found some powerpoint files on google).
I can add a RTP Proxy too.
But my purpose is only to bill a sip session in a realtime prepaid context.
Prepaid means prepaid by money, which can give x minutes of call, or y kb of
download.
My role is to study if SIP could be a good way to have concurrent use on a same
balance with voice and web at the same time ... And how realize it. I
understand when you explain me that proxy is not a safe way to do it, but how
is it possible to "check a few media packets to calculate amount of data (
media ) streamed per second, type of codec used, video or voice, etc ( this
will not work with silence supression )." without having a RTP proxy ?
I just need for audio or video a time duration, so a begin time, the final end
time, and renew the permission each x seconds or minuts to reserv money on the
balance
For data (web), I need to know what amount of data had passed through the
stream from begin time of the session till now.
Thank you again for your answers. You're helpful ;)
Guillaume
-----Message d'origine-----
De : Diego B [mailto:[EMAIL PROTECTED]
Envoy� : vendredi 15 avril 2005 16:55
� : Des Pommare Guillaume
Cc : [email protected]
Objet : Re: [Sip-implementors] SIP Billing proxy with billing on amount of data
and not on time
Hi;
First of all I thing a SIP Proxy server is not the right place to do SIP
billing, the sip proxy is defined as a transactinal engine, so there is no way
it can correlate INVITE and BYE transactions ( I know, there are lot of ways to
do that, but I'm refering to the standard.).
.
One end-point may send ACK or BYE directlly to the other end-point skiping the
proxy, so you will end up with lot of calls.
You may be using Record-Route, but htis not ensure you the end points will
support this.
the best solution is to use a 3PCC server implmeneting a B2BUA.
There you can control the 2 legs of the call. You can also check a few media
packets to calculate amount of data ( media ) streamed per second, type of
codec used, video or voice, etc ( this will not work with silence supression ).
Regards
Diego B..
Des Pommare Guillaume wrote:
>Hi !
>
>I'm in charge of creating a SIP billing proxy. For the moment I have no
>real problem to do it with billing on time duration. But what about
>billing on amount of data, for example 5$/Mb ?
>
>I set a timer on a 200OK from callee, and stop it on a BYE (end of
>balance or BYE from the callee or caller). This give me a time duration.
>
>
>If I want to make billing on amount of data, should I set up a RTP
>proxy server? this is "expensive" to do with huge traffic, and operator
>should be in charge of that kind of thing, not our billing proxy.
>
>Is there any way (from example with SDP) to be inform of amount of data
>transmitted during a session ?
>
>I am not really famialiar with SIP, I just study it since 2 weeks, and
>I found this problem in the SIP FAQ of Jonathan Rosenberg, this could
>be possible if the operator's gateway record "call" informations. If
>there is no gateway, what can be done ? (in a SIP only environment)
>
>
>"How do I charge/bill for Internet telephony using SIP?
>
> This depends on whether you plan to charge for SIP services like
>directory look-ups, call processing or mobility, for gateway services
>to the PSTN, or for carrying media data:
>
> SIP services
> The Authorization header can be used to indicate a customer
>identity that associates a SIP request with a billable entity.
>
> Examples of possibly chargeable SIP services include:
>
> * Directory services such as SIP proxy/redirect lookups;
> * Customer profile management;
>
> SIP server operations can be charged based on server logs or,
>for real-time billing, via AAA.
> Media services
> Media services include retrieving and storing voice mail, as
>well as transcoding of media streams. They are not initiated by SIP,
>but, for example, via RTSP.
>
> Gateway services
> Similar to SIP services. Care has to be taken to stop billing
>when (say) RTP voice data is no longer flowing through the gateway. The
>gateway will generate call detail records (CDRs) either directly or
>through RADIUS."
>
>Thanks
>
>Guillaume
>
>PS : sorry if my questions seems quite simple, for me there are
>difficult to assimilate ;)
>
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>
>
>
>
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