Summary:ASTERISK-00021: [request] SIP Session Timer Support
Reporter:timecop (timecop)Labels:
Date Opened:2003-07-27 18:33:58Date Closed:2011-06-07 14:05:25
Versions:Frequency of
Description:According to

This is a rather new SIP feature designed to timeout idle SIP sesions.
Unfortunately, the SIP provider I use decided to implement this feature. Asterisk does not support it.

The real reason for this SIP session timer (at my provider) seems to be because they bill per-minute for all SIP calls, therefore they don't want to bill for hours of calls that were over a while back.


I (temporarily) hacked in required headers into transmit_invite, however that cancels the call after Session-Expires timeout because INVITE/UPDATE isnt sent back during that period. Someone with more "advanced" knowledge of asterisk internals and who knows how to implement something like that, please take a look.
Comments:By: timecop (timecop) 2003-11-12 08:42:34.000-0600

SIP-Session-Timer support is now at version 1.2:

Can someone hack in support for this?

By: John Todd (jtodd) 2003-11-14 01:59:54.000-0600

See if you can drum up some support for it on the -dev list.

By: zoa (zoa) 2004-01-09 19:33:36.000-0600

this is related to http://bugs.digium.com/bug_view_page.php?bug_id=0000207

By: zoa (zoa) 2004-01-11 21:09:44.000-0600

incorrectly thought this was the same as bug 207... timecop says its not ...

By: timecop (timecop) 2004-01-11 21:18:52.000-0600

while this bug might be somehow related to the hanging sip channels, all this is really an extension to the SIP protocol to provide a way to timeout a session if "nothing happened" on it for a given period of time.

At the beginning of a session, new header is included, "Session-Expires"
as well as "Supported". Refer to the rfc for more info.

Before the timeout specified in Session-Expires runs out, uac must send either INVITE or UPDATE with another Session-Expires header, extending the timeout.

This is to prevent my SIP provider which manages to charge per minute for all SIP calls, from overbilling me incase my session gets stuck.

They don't support refresher:uas, so this would have to be done by asterisk.

By: jrollyson (jrollyson) 2004-01-12 00:16:26.000-0600

Discussion moved to bug 207

By: jrollyson (jrollyson) 2004-01-12 00:16:49.000-0600

bleh... meant closed

By: timecop (timecop) 2004-01-23 09:37:43.000-0600

bug ASTERISK-204 is going in the wrong direction.
I don't care for silence detection, or other crap.
I just want session timers implemented so that my sip provider doesnt disconnect
my calls after 5 minutes.

By: Brian West (bkw918) 2004-01-24 12:26:38.000-0600

207 is timer support and timer support only.  Thats the discussion we had on the last conf call about it.  We didn't even talk about silence detection/supression.