[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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