[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) noreply at issues.asterisk.org
Fri Mar 13 00:08:34 CDT 2015


     [ https://issues.asterisk.org/jira/browse/ASTERISK-24858?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel ]

Frankie Chin updated ASTERISK-24858:
------------------------------------

    Description: 
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.



  was:
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. 

Test 1:
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.

Test 2:
Change Asterisk A and B to use "ulaw" as well. And this time, the voice coming out from Softphone B is normal. 


> 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
>
> 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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