[asterisk-bugs] [JIRA] (ASTERISK-25684) noise instead of audio on slin codec with translation

[Rusty Newton (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-25684?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=229020#comment-229020 ] 

Rusty Newton commented on ASTERISK-25684:
-----------------------------------------

It may be unnecessary, but we like to have all the info we can up front. Please follow the instructions on the wiki [1] for how to collect debugging information from Asterisk during a call. For expediency, where possible, attach the debug with a '.txt' file extension so that the debug will be usable for further analysis.

Thanks!

[1] https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information



> noise  instead of audio on slin codec with translation
> ------------------------------------------------------
>
>                 Key: ASTERISK-25684
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-25684
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Codecs/General
>    Affects Versions: 13.1.0, 13.4.0, 13.6.0
>         Environment: Ubuntu 14.4 i686 3.19.0-42 on VMware WS 10
> FreeBSD 10.2 i386 on VirtualBox 5.0.10
> Ubuntu 14.4 i686 3.19.0 on bare hw (intel core i5)
>            Reporter: Victor Sverdlin
>            Assignee: Victor Sverdlin
>            Severity: Minor
>         Attachments: slin_noise_01.pcap
>
>
> Then one peer is slin (L16/8000) or slin16 (L16/16000) and second peer is other codec (tested with ulaw, gsm and speex) slin peer listen only noise. Noise disappear if other peer muted. Slin peer can listen voice if signal is gained down by AGC set to 10 and less (anyway sound is distorted).
> Both peers use sip channel.
> Testes with several peers:
> - slin: MicroSIP 3.10.9, custom HW device
> - other: antiSIP 4.2.9 (Android), SFLphone 1.3.0, Zoiper 3.3.25608
> Asterisk compiled with gcc 4.8.5(FreeBSD) and 4.8.4(Ubuntu).
> On Ubuntu Asterisk compiled with DONT_OPTIMIZE option.
> RTP capture will be attached.



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