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

[Beytullah ARSLAN (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-25845?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=230111#comment-230111 ] 

Beytullah ARSLAN commented on ASTERISK-25845:
---------------------------------------------

[yerel-telefon-sablonu](!)
type=endpoint
direct_media=yes
disable_direct_media_on_nat=yes
message_context=mesajlar
disallow=all
allow=g722
allow=g729
allow=ulaw
allow=alaw
allow=h264
allow_subscribe=yes
sub_min_expiry=30
transport=transport-udp

[auth-userpass](!)
type=auth
auth_type=userpass

[aor-single-reg](!)
type=aor
remove_existing=yes     ; Determines whether new contacts replace existing ones
max_contacts=2
qualify_frequency=5
authenticate_qualify=yes


;=====
[4160](yerel-telefon-sablonu)
context=custom-uluslararasi
auth=4160
aors=4160
callerid="caller id" <4160>
call_group= 10
pickup_group= 10
allow_subscribe=yes
sub_min_expiry=30
[4160](auth-userpass)
username=4160
password=pass
[4160](aor-single-reg)
;=====

> 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
>            Assignee: Joshua Colp
>            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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