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

[Alexander Traud (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-29320?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=255933#comment-255933 ] 

Alexander Traud commented on ASTERISK-29320:
--------------------------------------------

Yes and no. The Asterisk Team interprets the Jira field ‘affected version’ as ‘reported with version’. Consequently, the latest current version *when* the report has been created is expected there. This is terrible confusing for external parties, I know, because other bug trackers track the version which introduced the issue. So, yes, normally it would be my job to update that field to Asterisk 18.0. However, no, in the world of Asterisk it stays at the version reported. Beside that I do not have the rights to edit that field. Nevertheless, there is an extension to that unwritten rule: If a new LTS branch was released and the bug still applies, its version is added. This makes sure, the bug is not deleted when the originally reported LTS branch gets end-of-life.

However, however, the causing issue ASTERISK-28756 could be linked. This would invite the contributor [~kHarwell]. I could patch it but I am totally unsure if the current code does some sort of filtering. If it does not, the easiest way might be to update the private {{get_codecs(.)}} and return an {{ast_format_cap}} structure directly, because within {{get_codecs(.)}} the order is iterated already.

> 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, 18.2.1
>         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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