| Summary: | ASTERISK-27851: app_confbridge: Opus participants have bad quality in confbridge audio conference with non-20ms mixing interval | ||
| Reporter: | Aleksandr Salanov (havier163) | Labels: | pjsip |
| Date Opened: | 2018-05-11 07:21:49 | Date Closed: | |
| Priority: | Minor | Regression? | |
| Status: | Open/New | Components: | Applications/app_confbridge Codecs/codec_opus |
| Versions: | 13.18.0 13.20.0 13.21.0 15.4.0 | Frequency of Occurrence | Constant |
| Related Issues: | |||
| Environment: | PowerEdge R630/2 x Intel Xeon E5-2667 v4 3.2GHz()/RAM 32Gb/ | Attachments: | |
| Description: | We caught some strange behavior of the Asterisk that works as the audio conference bridge.
We have a solution: 1) The Hardware server 2) OS version is Linux version 3.10.0-693.21.1.el7.x86_64 (mockbuild@x86-ol7-builder-02.us.oracle.com) (gcc version 4.8.5 20150623 (Red Hat 4.8.5-16) 3) Asterisk 13.18 with confbridge and chan_sip 4) Last version of opus codec 1.3.0 5) The confbridge has settings: [default_bridge] type=bridge video_mode=none mixing_interval=40 sound_join=en/beep sound_only_person=en/beep sound_leave=en/nc_custom/confbridge-leave The scenario is: 1) More then one opus participants join to a the same conference bridge 2) Bad quality occur if one of them on mute (not server mute) or just silent. it affect only who on mute/silent. 3) The quality is good when both of them are speaking at the same time. Testing other version of the Asterisk: I tried pjsip with 13.18,13.20,13.21 and last 15.4 version. I got the same results. Workaround is: The issue has been resolved by changing mixing_interval to 20. I read documentation and found that 40 ms should cover sample rates 8-96 kHz. So, opus has 48 kHz but by some reason it doesn’t work properly. Is this the expected behavior or a bug? | ||
| Comments: | By: Asterisk Team (asteriskteam) 2018-05-11 07:21:50.616-0500 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. 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]. By: Aleksandr Salanov (havier163) 2018-06-22 15:14:20.238-0500 Hello Team, Do you have any updates? By: Richard Mudgett (rmudgett) 2018-06-22 15:27:47.955-0500 Your issue is in the queue. Your patience is appreciated as a developer may work the issue when time and resources become available. Asterisk is an open source project and community members work the issues on a voluntary basis. You are welcome to develop your own patches and submit them to the project.[1] If you are not a programmer and you are in a hurry to see a patch provided then you might try rallying support on the Asterisk users mailing list or forums.[2] Another alternative is offering a bug bounty on the asterisk-dev mailing list.[3] Often a little incentive can go a long way. [1]: https://wiki.asterisk.org/wiki/display/AST/Patch+Contribution+Process [2]: http://www.asterisk.org/community/discuss [3]: https://wiki.asterisk.org/wiki/display/AST/Asterisk+Bug+Bounties Any updates will get posted on this issue. | ||