Guillaume,
Just some thoughts on your questions.
Guillaume asked: "Is there any way (from example with SDP) to be
inform of amount of data
transmitted during a session ?"
- yes, some vendors do this but I don't know of a standard that
covers it, I believe it is proprietry to the manufacturers. Which is
problematic because a mix of vendors for handsets will mean the feature may
be supported anywhere from 0-100% and if supported by different vendors the
implementation will probably differ making extracting this information in a
uniform way an issue.
- also, what about getting these statistics from network devices
rather than the end points. You are more likely to have only a few core
routers etc that are used to interconnect your network (where data is
aggregated). Again this is most likely to be proprietry implementation by
the vendor but you will have much fewer devices and *may* be less likely to
have a mix of router device vendors. There are sextra systems and
performance issues that will need addressing with this kind of
implentation, but another advantage is it will be applicable to media other
than audio when these are added (messaging, video etc...).
Regards - Wayne.
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 ;)
_______________________________________________
Sip-implementors mailing list
Sip-implementors@cs.columbia.edu
http://lists.cs.columbia.edu/mailman/listinfo/sip-implementors
**********************************************************************
Any personal or sensitive information contained in this email and
attachments must be handled in accordance with the Victorian Information
Privacy Act 2000, the Health Records Act 2001 or the Privacy Act 1988
(Commonwealth), as applicable.
This email, including all attachments, is confidential. If you are not the
intended recipient, you must not disclose, distribute, copy or use the
information contained in this email or attachments. Any confidentiality or
privilege is not waived or lost because this email has been sent to you in
error. If you have received it in error, please let us know by reply
email, delete it from your system and destroy any copies.
**********************************************************************
_______________________________________________
Sip-implementors mailing list
Sip-implementors@cs.columbia.edu
http://lists.cs.columbia.edu/mailman/listinfo/sip-implementors
