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Summary:ASTERISK-30051: res_pjsip: No video after un-hold with moh_passthrough=yes
Reporter:Maximilian Fridrich (mfridrich)Labels:
Date Opened:2022-05-10 03:04:47Date Closed:2022-06-02 11:33:21
Priority:MinorRegression?No
Status:Closed/CompleteComponents:Resources/res_pjsip
Versions:16.16.0 18.11.1 19.3.3 Frequency of
Occurrence
Frequent
Related
Issues:
Environment:Linux 5.13.0, 20.04.1-Ubuntu, x86_64Attachments:( 0) debug_log_30051.txt
( 1) extensions.conf
( 2) pjsip.conf
( 3) unhold_video_bug_sip.pcapng
Description:During a call with audio and video, when one participant holds the call and then un-holds, the participant that initiated the hold sees no video.

The issue occurs only when {{moh_passthrough=yes}} is set in pjsip.conf.

It occurs most of the time, however sometimes there is video as expected. Furthermore, when Asterisk is built with the DEBUG_THREADS option, the issue does not occur at all.

The attached trace shows the un-hold re-INVITE of the UA (packet no. 31) with _sendrecv/sendrecv_, Asterisk sending _sendonly/sendonly_ to the second call leg (no. 33), then sending another re-INVITE to the second call leg with _sendrecv/sendonly_ (no. 36) . So Asterisk is ultimately setting the video stream of the second call leg sendonly from its perspective, which is why the UA of the first leg does not receive video.
Comments:By: Asterisk Team (asteriskteam) 2022-05-10 03:04:49.192-0500

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.

Then, if you are submitting a patch, please review the [Patch Contribution Process|https://wiki.asterisk.org/wiki/display/AST/Patch+Contribution+Process].

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By: Joshua C. Colp (jcolp) 2022-05-10 04:31:19.551-0500

We require additional debug to continue with triage of your issue. Please follow the instructions on the wiki [1] for how to collect debugging information from Asterisk. For expediency, where possible, attach the debug with a '.txt' file extension so that the debug will be usable for further analysis.

Thanks!

[1] https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information



By: Friendly Automation (friendly-automation) 2022-06-02 11:33:21.829-0500

Change 18571 merged by Friendly Automation:
chan_pjsip: Only set default audio stream on hold.

[https://gerrit.asterisk.org/c/asterisk/+/18571|https://gerrit.asterisk.org/c/asterisk/+/18571]

By: Friendly Automation (friendly-automation) 2022-06-02 11:37:36.320-0500

Change 18586 merged by Joshua Colp:
chan_pjsip: Only set default audio stream on hold.

[https://gerrit.asterisk.org/c/asterisk/+/18586|https://gerrit.asterisk.org/c/asterisk/+/18586]

By: Friendly Automation (friendly-automation) 2022-06-02 11:37:50.137-0500

Change 18584 merged by Joshua Colp:
chan_pjsip: Only set default audio stream on hold.

[https://gerrit.asterisk.org/c/asterisk/+/18584|https://gerrit.asterisk.org/c/asterisk/+/18584]

By: Friendly Automation (friendly-automation) 2022-06-02 11:38:15.589-0500

Change 18585 merged by Joshua Colp:
chan_pjsip: Only set default audio stream on hold.

[https://gerrit.asterisk.org/c/asterisk/+/18585|https://gerrit.asterisk.org/c/asterisk/+/18585]