[asterisk-bugs] [JIRA] (ASTERISK-26520) codec_opus: Generated fmtp line has no content

[scgm11 (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-26520?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=233499#comment-233499 ] 

scgm11 commented on ASTERISK-26520:
-----------------------------------

tests:

   size = ast_str_append(str, 0, "a=fmtp:%u ", payload);
   ast_log(LOG_WARNING, "(%i) TAMANO SIZE\n", size);

-->  opus_generate_sdp_fmtp: (11) TAMANO SIZE


and this:
 ast_log(LOG_WARNING, "STRLEN SIZE(%i)\n", ast_str_strlen(*str));
 if (size == ast_str_strlen(*str)) {
                ast_str_reset(*str);
        } else {
                ast_str_truncate(*str, -1);
                ast_str_append(str, 0, "\r\n");
        }

--> opus_generate_sdp_fmtp: STRLEN SIZE(38)


Never enters to ast_str_reset.


Other thing is no matter what I use in codecs.conf for  bitrate etc I get:

m=audio 18568 RTP/SAVPF 107 101
a=rtpmap:107 opus/48000/2
a=fmtp:107
a=rtpmap:101 telephone-event/8000




> codec_opus: Generated fmtp line has no content
> ----------------------------------------------
>
>                 Key: ASTERISK-26520
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-26520
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Codecs/codec_opus
>    Affects Versions: 13.12.0
>            Reporter: scgm11
>            Assignee: Unassigned
>         Attachments: fullverboselog.txt, LogSIPDEBUG.txt, newSDPs.txt, OSSDP.txt
>
>
> As reported at https://github.com/eface2face/rtcninja.js/issues/35 we somehow generated an fmtp line with no attributes. We should not do that.



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