[OpenSER-Users] FreeRADIUS-CDRTool Prepaid Connector 1.1 Released

Dan-Cristian Bogos danb.lists at googlemail.com
Wed Feb 13 19:26:16 CET 2008


Andy,

I would say both methods are having disadvantages and advantages.

1. The mediaproxy timeout is a plus if this turns to be stable . I had some
not so good experiences in the past and not really responsive support for my
issues, so I have dropped the idea. I will need to recheck, perhaps the
issues were solved.
2. Yate has no rtp detection, therefore will not detect your dead sessions.
I preferred to use it due to prepaid stuff and automatic header masking
features I told you about.
Accounting issues were discussed and rediscussed over and over on this list,
so I will not pop up the subject again.
I think the best accounting technique would be still the last device which
is in touch with your carrier which charges you, so if you send it to PSTN,
then I would say use accounting provided by your PSTN gateway.

Cheers,
DanB


On Feb 13, 2008 6:06 PM, Andy Smith <a.smith at ukgrid.net> wrote:

>  Hi Dan,
>
>   one other query on the below, regarding Yate providing more accuarate
> accounting, if OpenSER is used with mediaproxy will this not provide the
> same level of accuracy (as Mediaproxy actually sits in the RTP stream)?
>
>         thanks Andy.
>
> ----- Original Message -----
>  *From:* Dan-Cristian Bogos <danb.lists at googlemail.com>
> *To:* A.smith <a.smith at ukgrid.net>
> *Cc:* users at lists.openser.org
>  *Sent:* Wednesday, February 13, 2008 1:07 PM
> *Subject:* Re: [OpenSER-Users] FreeRADIUS-CDRTool Prepaid Connector 1.1Released
>
> Hi Andy,
>
> The original config was built with Yate in mind due to openser incapacity
> (before release 1.3) of disconnecting the calls. Since 1.3.0 the dialog
> module should be able to timeout the calls, in theory you should no longer
> need extra software like Yate.
>
> I would still recommend using Yate combined with OpenSER in the case you
> are doing some sort of "Carrier business", for  the following reasons:
> 1. It creates two different legs for your call (in and out) same as Cisco
> does, and hides one side from the other (eg. removes the via headers and any
> revealing ip information inside SDP - so your partners should not know where
> the traffic comes from).
> 2. You have more protocols available in.
> 3. Accounting it is bit more accurate (you have the session total duration
> inside the accounting packets), so radius will be no longer responsible of
> calculating the session durations from timestaps.
> 4. Yate can work in rtp_forward mode, therefore no extra overhead given by
> rtp processing.
>
> So basically what the connector does (as specified in the documentation),
> for each call which is authorized by radius, the connector will ask
> permission from cdrtool. If permission is granted, it will return in a avp
> to openser the maximum duration allowed for the call (timeout value) plus
> credit available, for the case of special uas able to display that.
> By receiving accounting stop packet, the connector will inform cdrtool
> about call disconnection therefore clearing the lock and debiting the
> balance inside cdrtool. The rtp stream has nothing to do with this scenario,
> so you don't need to touch your NAT support configuration, it's all in the
> signaling.
>
> Let me know if you need further info.
>
> Cheers,
> DanB
>
>
>
>
>
> On Feb 13, 2008 12:53 PM, A.smith <a.smith at ukgrid.net> wrote:
>
> > Hi Dan/List,
> >
> >  I was reading the post below and trying to understand how your config
> > works. If
> > you are implementing this with something like a Cisco PSTN then you need
> > all
> > of
> > these: PSTN, OpenSER, Mediaproxy and Yate involved in the SIP route?
> > Does
> > the RTP
> > stream have to route via Yate and mediaproxy? :S
> >
> > thanks for any help! cheers Andy.
> >
> > >Hey Marc,
> > >
> > >I use Yate for doing that. It is simple and works out of the box (with
> > adding few
> > >lines in configs of course).
> > >
> > >I take Session timeout returned from connector and pass it to yate in a
> > sip
> > header
> > >Process that header in regex routing and define the value as timeout
> > for
> > session.
> > >Yate knows by default that when a session has a parameter "timeout"
> > returned
> > >from routing to disconnect the call when timeout is hit.
> > >
> > >Let me know if you need further info, so I can send you some config
> > files
> > if you
> > >want to. You can contact me on IRC for live support (DanB).
> > >
> > >
> > >All the best,
> > >DanB
> >
> > ________________________________________________
> > Message sent using UK Grid Webmail 2.7.9
> >
> >
> >
> > _______________________________________________
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> > Users at lists.openser.org
> > http://lists.openser.org/cgi-bin/mailman/listinfo/users
> >
>
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