[asterisk-bugs] [JIRA] (ASTERISK-24858) Asterisk 13 PJSIP sends RTP packets in wrong byte order on Intel platform when using slin codec

[Frankie Chin (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-24858?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=225381#comment-225381 ] 

Frankie Chin commented on ASTERISK-24858:
-----------------------------------------

Attached configuration and log files.
Softphone A is the endpoint 2000 in pjsip_A.conf
Softphone B is the endpoint 1000 in pjsip_B.conf
Asterisk A is the endpoint peer_1_238 in pjsip_B.conf
Asterisk B is the endpoint peer_2_178 in pjsip_A.conf
full_corrupt is the debug log file.

> Asterisk 13 PJSIP sends RTP packets in wrong byte order on Intel platform when using slin codec
> -----------------------------------------------------------------------------------------------
>
>                 Key: ASTERISK-24858
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-24858
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Codecs/General, Core/RTP, Resources/res_rtp_asterisk
>    Affects Versions: 13.0.0
>            Reporter: Frankie Chin
>            Assignee: Frankie Chin
>         Attachments: extensions_A.conf, extensions_B.conf, full_corrupt, full_good, pjsip_A.conf, pjsip_B.conf
>
>
> Setup:
> I have two Asterisks registered to one another via PJSIP using only "slin" codec.   One PhonerLite softphone A is registered to Asterisk A using "ulaw" codec. Another softphone B is registered to Asterisk B using "ulaw" codec. Note: PhonerLite doesn't support slinear codec.
> Test:
> Softphone A dials an extension into Asterisk A, which will dial another extension into Asterisk B, which eventually dials Softphone B. The voice coming out from Softphone B is corrupted.



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