[asterisk-bugs] [JIRA] (ASTERISK-24002) No audio after WebRTC callee resumes call from hold

[Aleksei Kulakov (JIRA)]

     [ https://issues.asterisk.org/jira/browse/ASTERISK-24002?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel ]

Aleksei Kulakov updated ASTERISK-24002:
---------------------------------------

    Attachment: pjsip.conf
                extensions.conf
                http.conf
                noAudioAfterWebRtcCalleeUnhold.pcapng
                noAudioAfterWebRtcCalleeUnhold.log
                calleeChromeConsole.txt
                calleeChromeWebrtcDump.txt

sip.conf deleted entirely,
rtp.conf unchanged,
extensions.conf just added 2 dial rules  
http.conf - default changes for WebRTC support



> No audio after WebRTC callee resumes call from hold
> ---------------------------------------------------
>
>                 Key: ASTERISK-24002
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-24002
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Channels/chan_pjsip
>    Affects Versions: 12.2.0, 12.3.0, 12.4.0
>         Environment: Ubuntu 14.04
> PJProject(https://github.com/asterisk/pjproject 06/JUN/14)
> --enable-shared --disable-sound --disable-resample --disable-video --disable-opencore-amr --with-external-srtp CFLAGS="-g -DNDEBUG"
> Chrome M35
> SIPml-api.js?svn=224 with SDES patch from https://code.google.com/p/sipml5/issues/detail?id=183
>            Reporter: Aleksei Kulakov
>         Attachments: calleeChromeConsole.txt, calleeChromeWebrtcDump.txt, extensions.conf, http.conf, noAudioAfterWebRtcCalleeUnhold.log, noAudioAfterWebRtcCalleeUnhold.pcapng, pjsip.conf
>
>
> # Caller (SIP softphone or SIPml) calls WebRTC endpoint(SIPml)
> # Callee places call on hold
> # Callee resumes call from hold
> After resuming there is no no audio on both ends and many of 'SRTP unprotect failed with: authentication failure 110' messages in asterisk log.
> Issue reproducible with chan_pjsip(logs for that case) and chan_sip.
> Notes: When endpoints in pjsip.conf configured to use DTLS, resuming call from hold fails with '488 Not acceptable here'. Should i upload logs for this case too?



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