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ibc at aliax.net
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 Posted: Tue Nov 18, 2008 10:10 am Post subject: [Freeswitch-users] RFC 4028 - SIP Session Timers |
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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 vÃa 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>
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ibc at aliax.net
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| Posted: Tue Nov 18, 2008 6:36 pm Post subject: [Freeswitch-users] RFC 4028 - SIP Session Timers |
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El Martes, 18 de Noviembre de 2008, Iñaki Baz Castillo escribió:
| Quote: | 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 vÃa 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
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Freeswitch-users@lists.freeswitch.org
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brian at freeswitch.org
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ibc at aliax.net
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| Posted: Tue Nov 18, 2008 6:50 pm Post subject: [Freeswitch-users] RFC 4028 - SIP Session Timers |
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El Miércoles, 19 de Noviembre de 2008, Brian West escribió:
| Quote: | yes but RTP timers are in there too.
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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
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