Re: [Freeswitch-users] RFC 4028 - SIP Session Timers
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: While SIP Session Timers parameters are those: Am I right? Thanks a lot. -- Iñaki Baz Castillo ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] RFC 4028 - SIP Session Timers
yes but RTP timers are in there too. /b On Nov 18, 2008, at 5:26 PM, Iñaki Baz Castillo wrote: > > They seem to be related to SIP Session Timers (nothing related to > RTP), am I > right? ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] RFC 4028 - SIP Session Timers
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: They seem to be related to SIP Session Timers (nothing related to RTP), am I right? Thanks. -- Iñaki Baz Castillo ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] RFC 4028 - SIP Session Timers
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 <[EMAIL PROTECTED]> ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
