[asterisk-bugs] [JIRA] (ASTERISK-26564) Very poor sound quality in SiLK implementation

[Asterisk Team (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-26564?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=233463#comment-233463 ] 

Asterisk Team commented on ASTERISK-26564:
------------------------------------------

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].

> Very poor sound quality in SiLK implementation
> ----------------------------------------------
>
>                 Key: ASTERISK-26564
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-26564
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Codecs/codec_silk
>    Affects Versions: 13.12.1
>         Environment: Ubuntu 14.04. Asterisk 13.12.1.
>            Reporter: Sam For
>
> Using Asterisk 13.7.2 with the traud/asterisk-silk patch gives very good sound quality on transcoded audio.
> Using Asterisk 13.12.1 with the digium-silk-binary gives very poor sound  quality on transcoded audio.
> The quality difference can easily be heard by listening to the two different sound files I've attached.
> I have attached the following to this ticket:
> 1) PCAPs for the calls on versions 13.7 and 13.12.1
> 2) Audio file recordings for each version
> I used SLN16 sounds version 1.4.27 for both tests.
> I used the following dialplan:
> exten =>       5004,n,Answer
> exten =>       5004,n,Wait(3)
> exten =>       5004,n,Playback(priv-callee-options)
> exten =>       5004,n,Hangup
> I used Jitsi as a client with SiLK/16000.
> This was the codec setting for codecs.conf for 13.12.1:
> [silk16]
> type=silk
> samprate=16000
> fec=true
> packetloss_percentage=10
> maxbitrate=24000
> dtx=true
> As it stands now, the Digium SiLK codec binary is unusable unfortunately.



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