[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=248397#comment-248397 ] 

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

We appreciate the difficulties you are facing, however information request type issues would be better served in a different forum.

The Asterisk community provides support over IRC, mailing lists, and forums as described at http://asterisk.org/community. The Asterisk issue tracker is used specifically to track issues concerning bugs and documentation errors.

If this issue is actually a bug please use the Bug issue type instead.

Please see the Asterisk Issue Guidelines [1] for instruction on the intended use of the Asterisk issue tracker.

Thanks!

[1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines

> 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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