[asterisk-bugs] [JIRA] (ASTERISK-24143) WebRTC callee not receiving BYE after caller hangup

Aleksei Kulakov (JIRA) noreply at issues.asterisk.org
Thu Jul 31 02:07:56 CDT 2014


Aleksei Kulakov created ASTERISK-24143:
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             Summary: WebRTC callee not receiving BYE after caller hangup
                 Key: ASTERISK-24143
                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-24143
             Project: Asterisk
          Issue Type: Bug
      Security Level: None
          Components: Channels/chan_pjsip
    Affects Versions: 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" 

chromium 35.0.1916.153(rev274914) (launch options: --use-fake-ui-for-media-stream --disable-webrtc-encryption)

SIPml-api.js?svn=224
            Reporter: Aleksei Kulakov


1. WebRTC callee accepts call from caller of any type.
2. Caller hangup.
3. WebRTC callee does not get BYE sip packet(staying "in call" forever) and asterisk dumps following message to log:
{quote}
WARNING[348] pjsip: tsx0xb68399d4 ...Failed to send Request msg BYE/cseq=9511 (tdta0xb6804798)! err=320053 (DNS "Name Error" (PJLIB_UTIL_EDNS_NXDOMAIN))
{quote}



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