[OpenSIPS-Users] OpenSIPS generating BYE after notification "rtp session ended" from MediaProxy

Adrian Georgescu ag at ag-projects.com
Thu May 7 16:20:30 CEST 2009


What you describe already works as you suggested today, BYE messages  
are sent automatically by call_control and mediaproxy modules in  
various cases. Did you test  and it did not work for you?

Regards,
Adrian

On May 6, 2009, at 10:22 PM, Iñaki Baz Castillo wrote:

> Hi, MediaProxy can account the end of a "broken" session (after X  
> secons from
> the loss of RTP). This is done via Radius or MySQL directly from  
> MediaProxy,
> but the still alive endpoint doesn't receive a BYE.
>
> In real life, when talking from my mobile to other (GSM network), if  
> the
> remote loses the GSM connection (i.e. in a tunnel) I leave hearing  
> audio and
> my cell phone receives a "BYE" after X seconds (5-10). This is very  
> useful
> since I can understand that the session has ended due to loss of  
> connetion in
> the remote party.
>
> But with OpenSIPS + MediaProxy is not possible to do it. MediaProxy  
> can
> account the call for the real duration of the RTP session, but there  
> is no BYE
> when one of the endpoints loses the connection.
>
> So I wonder how feasible would be the following:
>
> - MediaProxy detects loss of RTP.
> - MediaProxy accounts the call (via Radius/MySQL) as now.
> - MediaProxy notifies OpenSIPS (MediaProxy -> dispacher ->  
> mediaproxy module)
> about the end of the call.
> - OpenSIPS generates BYE (local_route) in both directions of the  
> dialog (even
> if one of them probably wouldn't arrive to the endpoint).
>
>
> Other option would be:
>
> - MediaProxy detects loss of RTP.
> - MediaProxy does *NOT* account the call.
> - MediaProxy notifies OpenSIPS (MediaProxy -> dispacher ->  
> mediaproxy module)
> about the end of the call.
> - OpenSIPS generates BYE in both directions of the dialog.
> - OpenSIPS (in local_route) does the accounting (it must be know some
> MediaProxy parameters, as the time after MediaProxy ends the session  
> if RTP is
> lost). This BYE would also account the call end reason ("media  
> timeout") by
> setting some AVP used in the Radius acc module...
>
>
> Opinions?
>
>
> For now, media-relaying solutions (as MediaProxy) are useful, and  
> mostly used,
> for calls between SIP clients and SIP gws, so if the user (or gw)  
> crashes, the
> accounting is "fixed". But I expect that there is more world than  
> just calls
> between SIP users and PSTN gateways (imagine for example OpenSIPS +  
> MediaProxy
> being used in an IMS environment where calls between users *are*  
> really
> accounted for billing).
>
>
> Thanks for any comment. Best regards.
>
>
> -- 
> Iñaki Baz Castillo <ibc at aliax.net>
>
> _______________________________________________
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> Users at lists.opensips.org
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