Summary:	ASTERISK-26797: res_pjsip: Crash when freeing pool of cloned message
Reporter:	Ross Beer (rossbeer)	Labels:
Date Opened:	2017-02-16 06:40:57.000-0600	Date Closed:	2020-01-14 11:13:52.000-0600
Priority:	Major	Regression?
Status:	Closed/Complete	Components:	Channels/chan_pjsip
Versions:	GIT	Frequency of Occurrence	Occasional
Related Issues:
Environment:	Fedora 23	Attachments:	( 0) backtrace_2016-02-16_1215_clean.txt
Description:	Segfault when releasing resources.
Comments:	By: Asterisk Team (asteriskteam) 2017-02-16 06:40:58.291-0600 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) 2017-02-20 06:19:23.457-0600 Thank you for taking the time to report this bug and helping to make Asterisk better. Unfortunately, we cannot work on this bug because your description did not include enough information. Please read over the Asterisk Issue Guidelines [1] which discusses the information necessary for your issue to be resolved and the format that information needs to be in. We would be grateful if you would then provide a more complete description of the problem. At a minimum, we need: 1. The specific steps or actions you took that caused you to encounter the problem. 2. The behavior you expected and the location of documentation that led you to that expectation. 3. The behavior you actually encountered. To demonstrate the issue in detail, please include Asterisk log files generated per the instructions on the wiki [2]. If applicable, please ensure that protocol-level trace debugging is enabled, e.g., 'sip set debug on' if the issue involves chan_sip, and configuration information such as dialplan and channel configuration. Thanks! [1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines [2] https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information By: Asterisk Team (asteriskteam) 2017-03-06 12:00:01.503-0600 Suspended due to lack of activity. This issue will be automatically re-opened if the reporter posts a comment. If you are not the reporter and would like this re-opened please create a new issue instead. If the new issue is related to this one a link will be created during the triage process. Further information on issue tracker usage can be found in the Asterisk Issue Guidlines [1]. [1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines By: Adagio (studioadagio) 2017-03-10 04:26:20.334-0600 Hi It seems we have the same problem. We have a business application that uses both conventional telephony and VoIP. We use the PJSIP library to make VoIP calls from mobile devices (Android & iOS). On server side we have Asterisk with PJSIP. Sometimes "Asterisk" process crash with "double free or corruption". This happens shortly after the INVITE transaction was finished (we hear about 0.5s of sound) and only if the call was started on Android device. We tried to reproduce the crash with other softphones (Zoiper, CSipSimple, Ekiga) and pjsua in CLI but it doesn't crash. Also it doesn't crash when iOS app is used. So, it seems that, the problem is with our Android implementation, but we don't know where to search for the solution. We tried workarounds from here: ASTERISK-25274 ASTERISK-25275 But nothing worked. This crash occur once in about 200 calls. After using Valgrind (valgrind.org) to analyze Asterisk memory, we restart Asterisk and crash is happening more often. Is there a link ? You will find backtrace and debug in attachments. We tried Asterisk versions: 13.14 and 14.2 PJSSIP versions: 2.5.5, 2.6 (We tried to change audio codec but nothing changed) Thanks a lot By: Joshua C. Colp (jcolp) 2017-03-10 04:52:19.431-0600 [~studioadagio] Please open a new issue with the information instead of attaching here. By: Adagio (studioadagio) 2017-03-10 07:31:28.481-0600 Oh sorry Joshua ! I just created it there : ASTERISK-26853