[Freeswitch-users] RFC 4028 - SIP Session Timers

Iñaki Baz Castillo ibc at aliax.net
Tue Nov 18 07:07:09 PST 2008

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 at aliax.net>

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