[asterisk-bugs] [JIRA] (ASTERISK-24274) Codec Format Is Not Included in the SDP Media Attributes When SLIN48 Codec Is Used

[Rusty Newton (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-24274?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=222818#comment-222818 ] 

Rusty Newton commented on ASTERISK-24274:
-----------------------------------------

{quote}
Rusty, this is what I plan to do to create the patch:

* Download the source from http://svn.asterisk.org/svn/asterisk/tags/12.4.0/
* Make the changes in rtp_engine.c and frame.c and create the patch.
* Sign the License Agreement and follow the Code Review process to submit the patch.

Does it sound right? Thanks.
{quote}

Yes, except you probably want to grab the latest head of SVN for the 12 and 11 branch. Once you have a patch working in one, you'll want to validate it works in the other branch, as well as Trunk.

The Code Review process will have you put it on Reviewboard at that point for others to review and test. Thanks.

> Codec Format Is Not Included in the SDP Media Attributes When SLIN48 Codec Is Used
> ----------------------------------------------------------------------------------
>
>                 Key: ASTERISK-24274
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-24274
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Core/RTP
>    Affects Versions: 11.12.0, 12.4.0
>            Reporter: Frankie Chin
>         Attachments: sip_server_A.conf, sip_server_B.conf
>
>
> I first submitted this question in the forum and was later asked to submit it as an issue: http://forums.digium.com/viewtopic.php?f=13&t=91190&start=0&hilit=Confbridge&sid=9ee675f5b376c518edec9952fb0de9b5
> I described the issue background and my findings in the forum. So I think I don't need to repeat it here. In summary:
> 1) When using SLIN16, the media attributes are: "m=audio 19942 RTP/AVP 118 101"
> 2) When using SLIN48, the media attributes are: "m=audio 17868 RTP/AVP 101"
> Please find the attached sip.conf of Server A and B.
> Another thing I noticed in Server A's debug log is that the Joint capabilities are (nothing). This is the same when using "slin16" as well.
> [Aug 27 02:30:30] DEBUG[6124] chan_sip.c: *** Our native formats are (slin48) 
> [Aug 27 02:30:30] DEBUG[6124] chan_sip.c: *** Joint capabilities are (nothing) 
> [Aug 27 02:30:30] DEBUG[6124] chan_sip.c: *** Our capabilities are (slin48) 
> [Aug 27 02:30:30] DEBUG[6124] chan_sip.c: *** AST_CODEC_CHOOSE formats are slin48 
> [Aug 27 02:30:30] DEBUG[6124] chan_sip.c: *** Our preferred formats from the incoming channel are (slin) 
> [Aug 27 02:30:30] DEBUG[6124] chan_sip.c: This channel will not be able to handle video.
> What other settings I need to configure in order to get some valid joint capabilities?



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