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Summary:ASTERISK-29089: RTP Ports not cleared after hangup
Reporter:Ross Beer (rossbeer)Labels:
Date Opened:2020-09-21 08:51:24Date Closed:2020-09-23 09:56:47
Priority:MajorRegression?No
Status:Closed/CompleteComponents:Resources/res_rtp_asterisk
Versions:GIT Frequency of
Occurrence
Constant
Related
Issues:
Environment:CentOS 7Attachments:
Description:After Asterisk GIT-16-ab34417M some resources do not appear to be cleared after a call ends.

This has been confirmed on the #asterisk-dev chat where testing has also picked up the issue.

The following CLI output is displayed when the resources have been used up:
{quote}
[2020-09-21 13:37:48] ERROR[43037]: res_rtp_asterisk.c:3798 rtp_allocate_transport: Oh dear... we couldn't allocate a port for RTP instance '0x7fa3d4bf3350'
[2020-09-21 13:37:48] WARNING[43037]: res_rtp_asterisk.c:2790 dtls_srtp_stop_timeout_timer: Unable to cancel schedule ID 0.  This is probably a bug (res_rtp_asterisk.c: dtls_srtp_stop_timeout_timer, line 2790).
[2020-09-21 13:37:48] ERROR[43037]: res_pjsip_sdp_rtp.c:272 create_rtp: Unable to create RTP instance using RTP engine 'asterisk'
[2020-09-21 13:37:48] ERROR[43037]: res_pjsip_session.c:932 handle_incoming_sdp:  <endpoint>: Couldn't negotiate stream 0:audio-0:audio:sendrecv (nothing)
{quote}
Comments:By: Asterisk Team (asteriskteam) 2020-09-21 08:51:25.855-0500

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. Please note that log messages and other files should not be sent to the Sangoma Asterisk Team unless explicitly asked for. All files should be placed on this issue in a sanitized fashion as needed.

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.

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By: Joshua C. Colp (jcolp) 2020-09-21 08:55:05.003-0500

FYI, the error message "Couldn't negotiate stream" only exists in 18 - it is not present in 16.

By: Ross Beer (rossbeer) 2020-09-21 10:09:31.765-0500

That's strange as I did a show version on both servers and didn't notice it say 18. Checking the source code used it's also 16

By: Joshua C. Colp (jcolp) 2020-09-21 10:28:06.568-0500

Ah nevermind, case sensitive search got me. It is in 16 too.

By: Friendly Automation (friendly-automation) 2020-09-23 09:56:49.063-0500

Change 14982 merged by Joshua Colp:
res_pjsip_session: Fix session reference leak.

[https://gerrit.asterisk.org/c/asterisk/+/14982|https://gerrit.asterisk.org/c/asterisk/+/14982]

By: Friendly Automation (friendly-automation) 2020-09-23 10:01:09.124-0500

Change 14933 merged by Joshua Colp:
res_pjsip_session: Fix session reference leak.

[https://gerrit.asterisk.org/c/asterisk/+/14933|https://gerrit.asterisk.org/c/asterisk/+/14933]

By: Friendly Automation (friendly-automation) 2020-09-23 10:03:07.987-0500

Change 14983 merged by Joshua Colp:
res_pjsip_session: Fix session reference leak.

[https://gerrit.asterisk.org/c/asterisk/+/14983|https://gerrit.asterisk.org/c/asterisk/+/14983]

By: Friendly Automation (friendly-automation) 2020-09-23 10:04:00.101-0500

Change 14984 merged by Joshua Colp:
res_pjsip_session: Fix session reference leak.

[https://gerrit.asterisk.org/c/asterisk/+/14984|https://gerrit.asterisk.org/c/asterisk/+/14984]

By: Friendly Automation (friendly-automation) 2020-09-23 10:04:16.721-0500

Change 14985 merged by Joshua Colp:
res_pjsip_session: Fix session reference leak.

[https://gerrit.asterisk.org/c/asterisk/+/14985|https://gerrit.asterisk.org/c/asterisk/+/14985]