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

Frankie Chin (JIRA) noreply at issues.asterisk.org
Tue Sep 9 19:46:29 CDT 2014


    [ https://issues.asterisk.org/jira/browse/ASTERISK-24274?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=222598#comment-222598 ] 

Frankie Chin commented on ASTERISK-24274:
-----------------------------------------

Matt, referring to your previous comment #1, we want to be able to deliver 48 kHz high quality audio from one server to another. So I chose to use SLIN48 codec for the prototype I'm working on now. I'll take note of your other comments when creating the patch.

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.
 

> 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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