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

[Frankie Chin (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-24274?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=222437#comment-222437 ] 

Frankie Chin edited comment on ASTERISK-24274 at 8/27/14 6:38 PM:
------------------------------------------------------------------

I modified the "rtp_engine.c" in the "ast_rtp_engine_init()" to add in the following two lines...

set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_SLINEAR48, 0), 0, "audio", "L16", 48000);
add_static_payload(120, ast_format_set(&tmpfmt, AST_FORMAT_SLINEAR48, 0), 0);

I also modified "frame.c" in the "ast_codec_get_samples()" to add another switch case for slin48:
        
        case AST_FORMAT_SLINEAR:
        case AST_FORMAT_SLINEAR16:
        case AST_FORMAT_SLINEAR48:       
                samples = f->datalen / 2;
                break;

Now I'm able to get both servers talking using "slin48", which I verified using "sip show channels".


was (Author: fchin):
I modified the "rtp_engine.c" in the "ast_rtp_engine_init()" to add in the following two lines...

set_next_mime_type(ast_format_set(&tmpfmt, AST_FORMAT_SLINEAR48, 0), 0, "audio", "L16", 48000);
add_static_payload(120, ast_format_set(&tmpfmt, AST_FORMAT_SLINEAR48, 0), 0);

I also modified "frame.c" in the "ast_codec_get_samples()" to add another switch case for slin48:
        
        case AST_FORMAT_SLINEAR:
        case AST_FORMAT_SLINEAR16:
        case AST_FORMAT_SLINEAR48:       
                samples = f->datalen / 2;
                break;

Now I'm able to get both servers talking using "slin48".

> 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: Channels/chan_sip/General
>    Affects Versions: 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.



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