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 <[EMAIL PROTECTED]> 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 <[EMAIL PROTECTED]> > *To:* A.smith <[EMAIL PROTECTED]> > *Cc:* users@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 <[EMAIL PROTECTED]> 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 > > > > > > > > _______________________________________________ > > Users mailing list > > Users@lists.openser.org > > http://lists.openser.org/cgi-bin/mailman/listinfo/users > > > > ------------------------------ > > No virus found in this incoming message. > Checked by AVG Free Edition. > Version: 7.5.516 / Virus Database: 269.20.2/1271 - Release Date: > 11/02/2008 08:16 > >
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