[asterisk-bugs] [JIRA] (ASTERISK-24868) Bad audio with silk if connection is bridged over third server and g722 on one side

[Peter Katzmann (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-24868?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=225486#comment-225486 ] 

Peter Katzmann commented on ASTERISK-24868:
-------------------------------------------

Szenario
3 Server : lnx06-asteriskdev1, lnx06-asteriskdev2, 10.0.0.50
2 Users: 2203 (on lnx06-asteriskdev2), 1000 (on 10.0.0.50)

lnx06-asteriskdev1 know lnx06-asteriskdev2 and 10.0.0.50 and can talk with each other.

lnx06-asteriskdev2 and 10.0.0.50 can't talk with each other directly, only over lnx06-asteriskdev1

Codecs allowed from to lnx06-asteriskdev1:
lnx06-asteriskdev2 allow=g722, alaw, silk16, silk8, speex
10.0.0.50 allow=silk16,silk8, speex

2203 uses a OpenStage 60 with g722 enabled
1000 uses a Snom 370 with alaw enabled

When i make a call from 1000 to 2203 over lnx06-asteriskdev1 transcoding in the bridging server is done despite the fact that write through is enough.

I attach also a trace of a call in from 2203 to 1000 because in this case lnx06-asteriskdev1 chooses h261 as codec.



> Bad audio with silk if connection is bridged over third server and g722 on one side
> -----------------------------------------------------------------------------------
>
>                 Key: ASTERISK-24868
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-24868
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Channels/chan_sip/General, Codecs/codec_g722
>    Affects Versions: 11.16.0
>         Environment: ubuntu 12
>            Reporter: Peter Katzmann
>            Assignee: Peter Katzmann
>         Attachments: h261.tar.bz2, silk-bridge-err.tar.bz2
>
>
> We experience distorted audio in the following scenario:
> 3 Servers, 
> Phone 1 g722
> Phone 2 alaw
> The call is over wan from server 1 over server 2 to server 3
> If the call is direct between server 1 and server 2 audio is ok.
> If i force speex (server 2 silk codec disabled) audio is ok.
> It seems that server 2 does some transcoding when we use silk, because the media path says that audio goes in from silk to slin and out from slin to silk.
> In opposition, when i force speex no transcoding occurs.
> media path in is speex and out also.
> So the question is, why does serer 2 transcode to silk instead of transparently connecting the in and out path between server 1 and server 3.



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