Summary:	ASTERISK-29455: Local channels (dialed using Originate dialplan application) play back gsm files over ulaw files when both exist
Reporter:	N A (InterLinked)	Labels:	patch
Date Opened:	2021-05-27 12:38:21	Date Closed:	2022-11-08 09:16:01.000-0600
Priority:	Minor	Regression?
Status:	Closed/Complete	Components:	Codecs/General
Versions:	18.4.0	Frequency of Occurrence	Constant
Related Issues:
Environment:		Attachments:	( 0) translate.diff
Description:	Strange issue with codec conversion that I noticed today. Despite the translation costs of slin192 <-> ulaw and slin192 <-> gsm being identical, an slin192 will pick a gsm audio file if both a gsm and ulaw candidate exist for a specific file. It plays ulaw files correctly if there isn't a candidate (same name) gsm file. The correct behavior should be that ulaw is preferred over gsm, since the costs are identical, and ulaw is the higher quality codec, and being both uncompressed and raw audio, closer to slin format than gsm would be. Yet, it falls back to the lower quality codec for some reason which seems arbitrary. An analysis of the ast_translator_build_path() function in channel.c along with DEBUG messages confirms the problem occurs here, and likely the actual bug lies with the ast_translator_best_choice function in translate.c. It appears this function looks only at translation costs and the choice between two codecs can be arbitrary if costs are identical.
Comments:	By: Asterisk Team (asteriskteam) 2021-05-27 12:38:25.546-0500 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: N A (InterLinked) 2021-05-27 13:30:50.899-0500 The attached diff successfully resolves my *specific* issue. Some algorithm for determining if one algorithm is (from a quality perspective) superior to the other would need to be called here. By: Friendly Automation (friendly-automation) 2022-11-08 09:16:01.985-0600 Change 15953 merged by Friendly Automation: translate.c: Prefer better codecs upon translate ties. [https://gerrit.asterisk.org/c/asterisk/+/15953|https://gerrit.asterisk.org/c/asterisk/+/15953]