[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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