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

[Aleksei Kulakov (JIRA)]

Aleksei Kulakov created ASTERISK-24002:
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             Summary: 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: 12.3.0, 12.2.0, 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


# 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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