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

[George Joseph (JIRA)]

     [ https://issues.asterisk.org/jira/browse/ASTERISK-26564?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel ]

George Joseph updated ASTERISK-26564:
-------------------------------------

    Assignee: Sam For  (was: George Joseph)
      Status: Waiting for Feedback  (was: Open)

Hi Sam,

Can you do a test call again and while the call is in progress, do a "core show channel <channel name>"?  You can do a "core show channels" to get the specific channel or hit <TAB> after typing "core show channel ".  I want to see what path the transcoding is taking.

Does the issue only happen when transcoding to/from specific formats?

How about phone-to-phone or local only?

Finally is this chan_sip or chan_pjsip?

thanks!




> codec_silk: 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
>            Assignee: Sam For
>         Attachments: 02_silkTraud_ast13.7.2.pcap, 02_silkTraud_ast13.7.2.wav, 04_silkDigium_ast13.12.1.pcap, 04_silkDigium_ast13.12.1.wav
>
>
> 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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