| Summary: | ASTERISK-30071: rtp: Usage of rtp_timeout on WebRTC causes failure | ||||
| Reporter: | nappsoft (nappsoft) | Labels: | webrtc | ||
| Date Opened: | 2022-05-19 00:57:59 | Date Closed: | 2022-06-02 12:00:01 | ||
| Priority: | Minor | Regression? | Yes | ||
| Status: | Closed/Complete | Components: | Resources/res_pjsip_sdp_rtp Resources/res_rtp_asterisk | ||
| Versions: | 18.12.0 | Frequency of Occurrence | Constant | ||
| Related Issues: |
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| Environment: | Attachments: | ||||
| Description: | We recently migrated from asterisk 16.19.0 (with security patches) to asterisk 18.12.0 and PJSIP 2.12.
We now have a problem with connections to ConfBridges over websockets as soon as the the user connects with audio and video. If the user only connects with audio or if we set rtp_timeout to 0, everything works as expected. We didn't have this behavior with asterisk 16.19.0. | ||||
| Comments: | By: Asterisk Team (asteriskteam) 2022-05-19 00:58:00.843-0500 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]. Please note that once your issue enters an open state it has been accepted. As Asterisk is an open source project there is no guarantee or timeframe on when your issue will be looked into. If you need expedient resolution you will need to find and pay a suitable developer. Asking for an update on your issue will not yield any progress on it and will not result in a response. All updates are posted to the issue when they occur. Please note that by submitting data, code, or documentation to Sangoma through JIRA, you accept the Terms of Use present at [https://www.asterisk.org/terms-of-use/|https://www.asterisk.org/terms-of-use/]. By: nappsoft (nappsoft) 2022-05-19 02:19:25.770-0500 Reverting the following changeset solves the issue: https://github.com/asterisk/asterisk/commit/5875c7bb6c21a09d7cc38fbba8765561b258cefd By: Joshua C. Colp (jcolp) 2022-05-19 03:41:06.017-0500 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: Asterisk Team (asteriskteam) 2022-06-02 12:00:00.989-0500 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 | ||||