| Summary: | ASTERISK-28346: useless transcoding | ||
| Reporter: | Thomas Sevestre (to) | Labels: | |
| Date Opened: | 2019-03-22 09:31:46 | Date Closed: | 2020-01-14 11:13:56.000-0600 |
| Priority: | Major | Regression? | |
| Status: | Closed/Complete | Components: | Codecs/codec_opus |
| Versions: | 13.25.0 | Frequency of Occurrence | Constant |
| Related Issues: | |||
| Environment: | Attachments: | ||
| Description: | Here is the revelant informations :
/etc/asterisk/sip.conf : [100] disallow=all allow=alaw allow=opus ... [200] disallow=all allow=alaw allow=opus ... /etc/asterisk/extensions.conf : I force opus codec with SIP_CODEC and SIP_CODEC_OUTBOUND variables The call is established with Dial application. In asterisk console, both channels have the following transcoding information : NativeFormats: (opus) WriteFormat: slin48 ReadFormat: slin48 WriteTranscode: Yes (slin@48000)->(opus@48000) ReadTranscode: Yes (opus@48000)->(slin@48000) If I force alaw on both channels, it works as expected without transcoding. If I change codec order in sip.conf : [100] disallow=all allow=opus allow=alaw ... [200] disallow=all allow=opus allow=alaw ... Then I can force opus passthrough but when I force alaw there is a useless transcoding : NativeFormats: (alaw) WriteFormat: slin48 ReadFormat: slin48 WriteTranscode: Yes (slin@48000)->(slin@8000)->(alaw@8000) ReadTranscode: Yes (alaw@8000)->(slin@8000)->(slin@48000) Is it a known issue? | ||
| Comments: | By: Asterisk Team (asteriskteam) 2019-03-22 09:31:46.439-0500 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]. 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. By: Joshua C. Colp (jcolp) 2019-03-26 04:39:20.700-0500 You're going to need to provide full configuration to replicate this. As well when you say you force ulaw - is that for both channels, or only one? By: Thomas Sevestre (to) 2019-03-26 05:36:17.014-0500 I force the same codec on both channels. The idea is to prevent asterisk transcoding. From the outside, asterisk behave as expected (requested codecs are used) But when I force codecs that are not the first ones in order of preference, asterisk does transcode. I'll try to provide a full configuration but it will take some time By: Asterisk Team (asteriskteam) 2019-04-11 12:00:01.304-0500 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 | ||