[asterisk-bugs] [JIRA] (ASTERISK-29320) res_pjsip_sdp_rtp: Codec preference order of remote is not correct on unhold

[Richard Mudgett (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-29320?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=256756#comment-256756 ] 

Richard Mudgett edited comment on ASTERISK-29320 at 11/3/21 11:00 AM:
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since this only applies to outgoing_call_offer_pref I think this issue should actually be linked to this issue ASTERISK-28777



was (Author: f.floimair):
since this only applies to outgoing_call_offer_pref I think this issue should actually be linked to this issue https://issues.asterisk.org/jira/browse/ASTERISK-28777


> res_pjsip_sdp_rtp: Codec preference order of remote is not correct on unhold
> ----------------------------------------------------------------------------
>
>                 Key: ASTERISK-29320
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-29320
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Resources/res_pjsip_sdp_rtp
>    Affects Versions: GIT, 18.0.0, 18.2.1
>         Environment: CentOS 7
>            Reporter: Ross Beer
>            Assignee: Unassigned
>         Attachments: debug.txt, Flow.jpg
>
>
> When having two endpoints configured with the allow set to 'alaw,gsm' and then calling from one endpoint to another the call is set up with 'alaw' and there is two-way audio. 
> If a call is then put on hold and then re-connected there is either one way or no audio. This looks to be caused by the incorrect codec order in the 200 response from Asterisk:
> {noformat}
> Media Attribute (a): rtpmap:3 GSM/8000
> Media Attribute (a): rtpmap:8 PCMA/8000
> Media Attribute (a): rtpmap:101 telephone-event/8000
> Media Attribute (a): fmtp:101 0-16
> Media Attribute (a): ptime:20
> Media Attribute (a): maxptime:150
> Media Attribute (a): recvonly
> {noformat}
> If you set the endpoint to a single codec the issue is resolved.



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