[asterisk-bugs] [JIRA] (ASTERISK-26217) [patch] Codec 2 Mode 2400

[Asterisk Team (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-26217?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=231487#comment-231487 ] 

Asterisk Team commented on ASTERISK-26217:
------------------------------------------

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].

> [patch] Codec 2 Mode 2400
> -------------------------
>
>                 Key: ASTERISK-26217
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-26217
>             Project: Asterisk
>          Issue Type: Improvement
>      Security Level: None
>          Components: Codecs/General
>    Affects Versions: SVN
>            Reporter: Alexander Traud
>            Severity: Minor
>
> Asterisk already supports iLBC 30 and G.729. If you want to reduce further, you could [add AMR|http://github.com/traud/asterisk-amr]. However, to go for even more data reduction, this patch adds [Codec 2|http://www.rowetel.com/codec2.html] in mode [2400|http://lists.digium.com/pipermail/asterisk-dev/2015-December/075209.html]. Please, remember the overhead created by RTP, UDP, and IP, which might  over the alternatives.
> [This patch|https://github.com/traud/asterisk-codec2] is for Asterisk 13 and newer. If you use an older version:
> - [Asterisk 11…|http://svn.code.sf.net/p/freetel/code/codec2/branches/0.5/asterisk-11/]
> - [Asterisk 1.8…|http://svn.code.sf.net/p/freetel/code/codec2/branches/0.5/asterisk/]
> Thanks goes to [Antonio Eugenio Burriel|http://github.com/aeburriel/codec2/tree/master/asterisk-11], who provided the starting point with his module for Asterisk 11.



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