[asterisk-bugs] [JIRA] (ASTERISK-24779) Passthrough OPUS codec not working with chan_pjsip

[Alexander Traud (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-24779?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=228177#comment-228177 ] 

Alexander Traud commented on ASTERISK-24779:
--------------------------------------------

Sean, thank you for your patch. I took it over for review because it works here and I need it. I hope, you do not mind. About that warning message {{ast_codec_samples_count}}: Please, open a new issue and describe your setup in greater detail. Because, I am not able to reproduce that warning yet (chan_pjsip pass-through, chan_pjsip transcoding, internally or bridged to chan_sip). I would be very interested in that, because I have two transcoding modules myself which (would) need the amount of samples calculated from actual data in the payload. So, I am not sure either why pass-through needs that information.

> Passthrough OPUS codec not working with chan_pjsip
> --------------------------------------------------
>
>                 Key: ASTERISK-24779
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-24779
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Channels/chan_pjsip
>    Affects Versions: 13.6.0
>         Environment: CentOS 6.6 x86
> pjsip v2.3 compiled from source according to Asterisk recommendations
> Asterisk v13.2.0 compiled from source
> opus-devel-1.1-1.el6.i686.rpm installed from epel repo (if that matters?)
>            Reporter: PowerPBX
>            Assignee: Alexander Traud
>         Attachments: asterisk-24779.patch, full, full_pjsip
>
>
> With 2 extensions and no NAT operating as direct_media=yes with no "r" in dial option (ie.  Passthrough codec mode) I am unable to communicate from one extension to another using the build in Opus Codec in the 2 extensions with Opus as the only active codec on the extensions.  I tried PhonerLite v2.21 and v2.22 beta as well as Xlite v4.7
> When I switched the extensions from using chan_pjsip to chan_sip they were able to communicate with each other via OPUS codec.  There is no OPUS codec installed on Asterisk so passthrough is the only possible way they can communicate using that codec.
> The following errors were observed from CLI
> {noformat}
> res_pjsip_sdp_rtp.c:247 set_caps: No joint capabilities for 'audio' media stream between our configuration((speex|opus)) and incoming SDP((nothing))
> chan_pjsip.c:530 chan_pjsip_answer: Unable to push answer task to the threadpool. Cannot answer call
> {noformat}



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