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

[PowerPBX (JIRA)]

PowerPBX created ASTERISK-24779:
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             Summary: 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 x86
pjsip v2.3 compiles from source according to Asterisk recommendations
Asterisk v13.2.0 compiled from source
            Reporter: PowerPBX


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.  I tried PhonerLite v2.21 and v2.22 beta as well as Xlite v4.7

When I switched the extension to chan_sip they were able to communicate with each other via OPUS codec.  There is no OPUS codec installed on Asterisk.

The following errors were observed from CLI

res_pjsip_sdp_rtp.c:247 set_caps: No joint capabilities for 'audio' media stream between our configuration((ulaw|alaw|gsm|g726|speex|opus)) and incoming SDP((nothing))

chan_pjsip.c:530 chan_pjsip_answer: Unable to push answer task to the threadpool. Cannot answer call



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