[Freeswitch-users] RFC 4028 - SIP Session Timers

Iñaki Baz Castillo ibc at aliax.net
Tue Nov 18 15:26:55 PST 2008

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 


Iñaki Baz Castillo

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