[asterisk-bugs] [JIRA] (ASTERISK-25845) res_pjsip_sdp_rtp: Wrong audio codec used when video enabled

[Joshua Colp (JIRA)]

     [ https://issues.asterisk.org/jira/browse/ASTERISK-25845?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel ]

Joshua Colp updated ASTERISK-25845:
-----------------------------------

    Description: 
The codec is negotiated with the call initiator when the initator places a call and a codec is choosed. For instance g722 was choosed between the caller and Asterisk.
Between called party and Asterisk a suitable another codec for instance ulaw was choosed.
When the sound starts from the caller, for the translation, Asterisk looks for the list of available codecs of the called party's allowed codec list and if it founds the caller's codec (g722) it uses it. INSTEAD of using the negotiated codec (ulaw) af called party. And this a problem that can be simulated easily with different codecs also.
And I think you can fix this problem easily.


  was:
The codec is negotiated with the call initiator when the initator places a call and a codec is choosed. For instance g722 was choosed between the caller and Asterisk.
Between called party and Asterisk a suitable another codec for instance ulaw was choosed.
When the sound starts from the caller, for the translation, Asterisk looks for the list of available codecs of the called party's allowed codec list and if it founds the caller's codec (g722) it uses it. INSTEAD of using the negotiated codec (ulaw) af called party. And this a problem that can be simulated easily with different codecs also.
And I think you can fix this problem easily.
pjsip set logger on output bellow link.
https://www.dropbox.com/s/7wziye4n2mif8xy/pjsipsetloggeron.txt?dl=0



> res_pjsip_sdp_rtp: Wrong audio codec used when video enabled
> ------------------------------------------------------------
>
>                 Key: ASTERISK-25845
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-25845
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Resources/res_pjsip_sdp_rtp
>    Affects Versions: 13.7.2
>         Environment: Gento-Linux, 4.1.12 Asterisk compiled directly from source with pjsip 2.4.5 support
>            Reporter: Beytullah ARSLAN
>            Severity: Minor
>         Attachments: pjsipsetloggeron.txt
>
>
> The codec is negotiated with the call initiator when the initator places a call and a codec is choosed. For instance g722 was choosed between the caller and Asterisk.
> Between called party and Asterisk a suitable another codec for instance ulaw was choosed.
> When the sound starts from the caller, for the translation, Asterisk looks for the list of available codecs of the called party's allowed codec list and if it founds the caller's codec (g722) it uses it. INSTEAD of using the negotiated codec (ulaw) af called party. And this a problem that can be simulated easily with different codecs also.
> And I think you can fix this problem easily.



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