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

[Sean Bright (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-24779?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=226505#comment-226505 ] 

Sean Bright commented on ASTERISK-24779:
----------------------------------------

We're talking about passthrough, so no {{codec_opus}} is loaded. Asterisk defines the opus ast_codec struct at line ~700 of {{main/codec_builtin.c}} and doesn't include the {{samples_count}} callback (because the stream needs to be decoded to determine that information).

The part I was getting stuck on is why Asterisk needs to care about the number of samples for the purposes of passthrough, and I don't know enough about Asterisk's RTP stack to make an intelligent guess.

> 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: 13.2.0
>         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: PowerPBX
>         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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