Summary:	ASTERISK-25794: Chan_sip allocates RTP ports even for rejected calls
Reporter:	Stefan Engström (StefanEng86)	Labels:
Date Opened:	2016-02-15 07:02:48.000-0600	Date Closed:	2016-02-15 10:47:19.000-0600
Priority:	Major	Regression?
Status:	Closed/Complete	Components:	Channels/chan_sip/General
Versions:	13.5.0	Frequency of Occurrence
Related Issues:
Environment:		Attachments:
Description:	Unwanted sip INVITES to my asterisk 13 go to the default context [default] where I do just Hangup. Asterisk then sends Trying and Decline, but asterisk also temporarily opens rtp ports. As a consequence, if I get N unwanted invites within a short time and I have less than 2N ports available, then for a short time all ports will be blocked, so that the next legitimate invite will get rejected. Is there a way around this? I may have misunderstood parts of this process; if so please correct me.
Comments:	By: Asterisk Team (asteriskteam) 2016-02-15 07:02:49.517-0600 Thanks for creating a report! The issue has entered the triage process. That means the issue will wait in this status until a Bug Marshal has an opportunity to review the issue. Once the issue has been reviewed you will receive comments regarding the next steps towards resolution. A good first step is for you to review the [Asterisk Issue Guidelines|https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines] if you haven't already. The guidelines detail what is expected from an Asterisk issue report. Then, if you are submitting a patch, please review the [Patch Contribution Process|https://wiki.asterisk.org/wiki/display/AST/Patch+Contribution+Process]. By: Joshua C. Colp (jcolp) 2016-02-15 10:47:19.379-0600 chan_sip initializes RTP instances when the dialog itself is created, to defer this would require additional work in the module itself. There's no ability to configure that. If you'd like to provide a change to add support for this it could be reviewed and included, but the fact it does this is not a bug itself.