Hi, I've read that FS supports/implements Session Timers to monitorice both legs of a call. How to enable it? I mean: alice ------- FS -------- bob - alice calls bob vía FS - FS calls bob. - bob answers (sends 200 OK). - "bypass_media" mode, no RTP through FS. - FS establishes a SIP dialog with alice and other one with bob. - From this moment FS starts sending periodically in-dialog INVITE/UPDATE to both legs in order to check if each SIP dialog is still alive in both endpoints. - In case alice crashes (looses dialog info), alice will reply "481 Call/Transaction doesn't exist" when the in-dialog INVITE/UPDATE arrives from FS, so FS will understand that alice has ended the dialog (or has crashed) and sends a BYE to bob. Is it possible with FS? how to enable it? -- Iñaki Baz Castillo <ibc@aliax.net>
El Martes, 18 de Noviembre de 2008, Iñaki Baz Castillo escribió:
Hi, I've read that FS supports/implements Session Timers to monitorice both legs of a call. How to enable it? I mean:
alice ------- FS -------- bob
- alice calls bob vía FS - FS calls bob. - bob answers (sends 200 OK). - "bypass_media" mode, no RTP through FS. - FS establishes a SIP dialog with alice and other one with bob. - From this moment FS starts sending periodically in-dialog INVITE/UPDATE to both legs in order to check if each SIP dialog is still alive in both endpoints. - In case alice crashes (looses dialog info), alice will reply "481 Call/Transaction doesn't exist" when the in-dialog INVITE/UPDATE arrives from FS, so FS will understand that alice has ended the dialog (or has crashed) and sends a BYE to bob.
Is it possible with FS? how to enable it?
I've found those options in Sofia profiles: <param name="enable-timer" value="false"/> <param name="minimum-session-expires" value="120"/> They seem to be related to SIP Session Timers (nothing related to RTP), am I right? Thanks. -- Iñaki Baz Castillo
El Miércoles, 19 de Noviembre de 2008, Brian West escribió:
yes but RTP timers are in there too.
Well, but I expect that RTP timers parameters are the following: <param name="use-rtp-timer" value="true"/> <param name="rtp-timer-name" value="soft"/> <param name="rtp-timeout-sec" value="300"/> <param name="rtp-hold-timeout-sec" value="1800"/> While SIP Session Timers parameters are those: <param name="enable-timer" value="false"/> <param name="minimum-session-expires" value="120"/> Am I right? Thanks a lot. -- Iñaki Baz Castillo
participants (2)
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Brian West
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Iñaki Baz Castillo