[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:
---------------------------------------

    Description: 
# Caller 6001(SIP softphone, but results for WebRTC or 'originate' caller is the same) calls WebRTC endpoint 354(SIPml)
# Callee places call on hold
# Callee resumes call from hold

After resuming there is 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.
Can be reproduced even when call is originated by asterisk('originate PJSIP/354 application musiconhold')

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?


  was:
# Caller 6001(SIP softphone, but results for WebRTC/SIPml caller is the same) calls WebRTC endpoint 354(SIPml)
# Callee places call on hold
# Callee resumes call from hold

After resuming there is 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?



> 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 6001(SIP softphone, but results for WebRTC or 'originate' caller is the same) calls WebRTC endpoint 354(SIPml)
> # Callee places call on hold
> # Callee resumes call from hold
> After resuming there is 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.
> Can be reproduced even when call is originated by asterisk('originate PJSIP/354 application musiconhold')
> 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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