[asterisk-dev] SIP Timers and "100 Trying" on a Re-INVITE. Resource leak.
Kevin P. Fleming
kpfleming at digium.com
Wed Jun 13 10:32:13 CDT 2012
On 06/13/2012 10:27 AM, Steve Davies wrote:
> Should I raise this as a JIRA ticket (with the cleaned-up grammar)?
Yes, I think you should, and the summary can be as simple as "SIP
re-INVITEs have no transaction timeout" or similar. I've already
notified the rest of the Asterisk development team here at Digium about
it, so they'll be watching for the issue to be entered.
As best I can tell, this is the *only* SIP request method where this
problem can happen, because it is the only one that allows provisional
responses.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber: kfleming at digium.com | SIP: kpfleming at digium.com | Skype: kpfleming
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at www.digium.com & www.asterisk.org
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