[asterisk-bugs] [JIRA] (ASTERISK-28584) Configure direct_media=yes in pjsip. Conf , don't valid, Media still flows asterisk

[Asterisk Team (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-28584?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=248396#comment-248396 ] 

Asterisk Team commented on ASTERISK-28584:
------------------------------------------

Thanks for creating a report! The issue has entered the triage process. That means the issue will wait in this status until a Bug Marshal has an opportunity to review the issue. Once the issue has been reviewed you will receive comments regarding the next steps towards resolution.

A good first step is for you to review the [Asterisk Issue Guidelines|https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines] if you haven't already. The guidelines detail what is expected from an Asterisk issue report.

Then, if you are submitting a patch, please review the [Patch Contribution Process|https://wiki.asterisk.org/wiki/display/AST/Patch+Contribution+Process].

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> Configure direct_media=yes in pjsip. Conf   ,don't valid, Media still flows asterisk
> ------------------------------------------------------------------------------------
>
>                 Key: ASTERISK-28584
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-28584
>             Project: Asterisk
>          Issue Type: Information Request
>      Security Level: None
>          Components: . I did not set the category correctly.
>    Affects Versions: 13.26.0
>         Environment: centos6.5 
>            Reporter: pengcheng wang
>              Labels: pjsip
>
> Configure direct_media=yes in pjsip. Conf.  The calling and called SDP have part of the same code, but they negotiate with asterisk respectively, resulting in different codes .In this case, direct communication between the two ends of direct media cannot be realized.  log content :    bridge_native_rtp.c: Bridge 'be1c247f-ced2-4728-9e22-3d9a702950c3': Channel codec0 = (ulaw) is not codec1 = (alaw), cannot native bridge in RTP .
> I want to make it happen:
> How to negotiate with asterisk SDP using the calling and called public codes?How should I modify the configuration?



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