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

Iñaki Baz Castillo ibc at aliax.net
Wed May 6 22:22:56 CEST 2009


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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