[Asterisk-Dev] SIP channels not cleared

Olle E. Johansson oej at edvina.net
Sun Aug 21 02:33:10 MST 2005


John Todd wrote:
> At 7:06 AM +0200 on 8/19/05, Olle E. Johansson wrote:
> 
>> Chee Foong wrote:
>>
>>>  OK, I under stand.
>>>  So, can this be considered a bug in asterisk?
>>>  Since it knows how to response to a BYE, it should also know it's
>>> time to
>>>  clear the channel.
>>
>>
>> The real fault here is that the other end issues a BYE when we have no
>> session set up by
>> INVITE/200 OK/ACK - to cancel a pending INVITE you use CANCEL, not BYE.
>> That is a bug, please ask your vendor to look up CANCEL in the SIP rfc.
>>
>> And yes, we should be able to handle faulty devices better, but will
>> concentrate our energy on being able to improve the way we handle
>> devices that actually support basic SIP according to the standard. ;-)
>>
>> /Olle
> 
> 
> 
> This problem could perhaps could be resolved by implementation of
> session-timers on the Asterisk side, assuming that the UAC also
> supported (or at least did not crash on) such timers.
> 
> http://www.faqs.org/rfcs/rfc4028.html
> 
> If Asterisk sent re-INVITEs after the Session-Expires: duration, then it
> (Asterisk) could close channels which did not respond.  I would think
> that this would be something that could be set on a per-peer basis or
> globally.
> 
> I believe my previous tests with Asterisk showed that Asterisk supported
> Session-Expires: in a non-harmful way (i.e.: did not crash) but Asterisk
> did not seem to have any "hooks" for generating a Session-Expires:
> header or creation of timers.  Does anyone have any alternate
> information?  It's been a year or so since I experimented with equipment
> using session-timers.
> 
...and you haven't seen my bug report with a patch for SIP timers
either? Not session timers, but as a starting point an implementation of
the standard T1 timer for retransmits. When that is done, SIP timers
would not be a bad thing to add.

/O



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