| Summary: | ASTERISK-29104: func_jitterbuffer: Calling JITTERBUFFER multiple times does not work properly | ||
| Reporter: | Robert Sutton (rsutton@noojee.com.au) | Labels: | |
| Date Opened: | 2020-10-04 22:15:42 | Date Closed: | |
| Priority: | Minor | Regression? | No |
| Status: | Open/New | Components: | Core/Jitterbuffer |
| Versions: | 16.11.1 16.13.0 | Frequency of Occurrence | Constant |
| Related Issues: | |||
| Environment: | Ubuntu 18.04, Docker | Attachments: | ( 0) debug_log_123456.txt |
| Description: | It seems that a PJSIP channel spanning multiple bridges where jitter buffer is enabled on it between each bridge has accumulating lag (abount 250ms per call) and eventually, after about 10 bridges no audio will pass.
This is the command being used Set(JITTERBUFFER(adaptive)=500,,100) The process: The Primary channel is brought up via manager Originate and placed into dial plan where jitter buffer is enabled and then on to AGI. A second channel is brought up via manager Originate and jitter buffer is enabled and placed into AGI via dial-plan. AGI on the second channel then bridges the two channels, when the second channel hangs up the Primary channel returns to dial-plan where jitter buffer is enabled again and then on to AGI to wait for another channel to be bridged. With each successive cycle the audio delay increases, and after about 10 cycles audio only passes intermittently. This can be mitigated by using a channel variable on the Primary channel and ensuring that jitter buffer is enabled only once during it's life. I've tried to reproduce this using only dial plan but am not able to, I'd like to be able to provide some useful diagnostic information but there doesn't seem to be any ability to log jitter buffer behavior? Please suggest how to capture diagnostic information. | ||
| Comments: | By: Asterisk Team (asteriskteam) 2020-10-04 22:15:43.308-0500 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: Joshua C. Colp (jcolp) 2020-10-05 05:40:28.207-0500 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: Robert Sutton (rsutton@noojee.com.au) 2020-10-06 22:20:05.719-0500 logs as requested | ||