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

[Mark Michelson (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-26520?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=233775#comment-233775 ] 

Mark Michelson commented on ASTERISK-26520:
-------------------------------------------

I've discovered the reason why the empty fmtp line is there. The Opus format attribute module is assuming that it is being given an empty string into which to place format attributes. However, chan_sip gives it the entire SDP it has built so far. When the format attribute module attempts to determine if it added any attributes, it does not properly calculate what it did, and so it ends up leaving the empty fmtp line in the SDP.

Regarding your codec.conf issue, there is a bug in codec_opus where if you name the section in codecs.conf "opus", your options will end up getting overwritten by the default attributes for Opus. You can instead name the section something like "myopus", and then in your sip.conf, you can set "allow=myopus" in order to use those attributes.

> 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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