[asterisk-bugs] [JIRA] (ASTERISK-25684) codec: Translation of slin16 results in noise

[Joshua Colp (JIRA)]

     [ https://issues.asterisk.org/jira/browse/ASTERISK-25684?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel ]

Joshua Colp updated ASTERISK-25684:
-----------------------------------

    Status: Open  (was: Triage)

> codec: Translation of slin16 results in noise
> ---------------------------------------------
>
>                 Key: ASTERISK-25684
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-25684
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Codecs/General, Resources/res_rtp_asterisk
>    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: Unassigned
>            Severity: Minor
>         Attachments: cli_output.txt, myDebugLog.txt, slin_noise_01.pcap, slin_noise_15jan2015.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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