Summary: | ASTERISK-30354: chan_iax2: Lack of formats prior to receiving voice frames causes jitterbuffer to stall | ||
Reporter: | N A (InterLinked) | Labels: | |
Date Opened: | 2022-12-14 09:58:57.000-0600 | Date Closed: | 2023-02-28 07:44:29.000-0600 |
Priority: | Minor | Regression? | |
Status: | Closed/Complete | Components: | Channels/chan_iax2 |
Versions: | 20.0.1 | Frequency of Occurrence | Constant |
Related Issues: | |||
Environment: | Attachments: | ||
Description: | Currently, chan_iax2 only calls jb_get to read frames from the jitterbuffer when pvt->voiceformat has been set, which only happens after receiving a voice frame from the other side. This means if the jitterbuffer is enabled, non-voice frames get stalled in the jitterbuffer until a voice frame is received.
The workaround is to disable the jitterbuffer completely, but that may not be desirable. This is kind of silly, since when the call is negotiated, both sides know what format the call is going to use (or at least start with). To fix this, we now fallback to using peerformat if voiceformat hasn't been set yet, which will only happen prior to receiving voice frames. | ||
Comments: | By: Asterisk Team (asteriskteam) 2022-12-14 09:59:02.690-0600 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. 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: Friendly Automation (friendly-automation) 2023-02-28 07:44:29.804-0600 Change 19919 merged by Friendly Automation: chan_iax2: Fix jitterbuffer regression prior to receiving audio. [https://gerrit.asterisk.org/c/asterisk/+/19919|https://gerrit.asterisk.org/c/asterisk/+/19919] By: Friendly Automation (friendly-automation) 2023-02-28 07:55:17.459-0600 Change 19940 merged by George Joseph: chan_iax2: Fix jitterbuffer regression prior to receiving audio. [https://gerrit.asterisk.org/c/asterisk/+/19940|https://gerrit.asterisk.org/c/asterisk/+/19940] By: Friendly Automation (friendly-automation) 2023-02-28 07:55:47.867-0600 Change 19712 merged by George Joseph: chan_iax2: Fix jitterbuffer regression prior to receiving audio. [https://gerrit.asterisk.org/c/asterisk/+/19712|https://gerrit.asterisk.org/c/asterisk/+/19712] |