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

[Abhay Gupta (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-24143?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=221563#comment-221563 ] 

Abhay Gupta commented on ASTERISK-24143:
----------------------------------------

This is when the other end disconnects and call is properly terminated 

-- Executing [101 at default:1] NoOp("PJSIP/102-0000000e", "") in new stack
    -- Executing [101 at default:2] Dial("PJSIP/102-0000000e", "PJSIP/101") in new stack
    -- Called PJSIP/101
    -- PJSIP/101-0000000f is ringing
    -- PJSIP/101-0000000f answered PJSIP/102-0000000e
    -- Channel PJSIP/102-0000000e joined 'simple_bridge' basic-bridge <6ba61987-c5d4-49db-9cea-a2e2987f1cf3>
    -- Channel PJSIP/101-0000000f joined 'simple_bridge' basic-bridge <6ba61987-c5d4-49db-9cea-a2e2987f1cf3>
       > 0x7f1c0809e9b0 -- Probation passed - setting RTP source address to 192.168.1.242:61054
[Aug 12 15:36:50] ERROR[2565]: pjsip:0 <?>: 	icess0x7f1c0c0 ..Error sending STUN request: Invalid argument
       > 0x7f1c080785b0 -- Probation passed - setting RTP source address to 192.168.1.161:58556
    -- Channel PJSIP/101-0000000f left 'simple_bridge' basic-bridge <6ba61987-c5d4-49db-9cea-a2e2987f1cf3>
    -- Channel PJSIP/102-0000000e left 'simple_bridge' basic-bridge <6ba61987-c5d4-49db-9cea-a2e2987f1cf3>
  == Spawn extension (default, 101, 2) exited non-zero on 'PJSIP/102-0000000e'

> 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
>            Assignee: Aleksei Kulakov
>         Attachments: 354ChromeConsole.log, debuglog.1, pjsip.conf
>
>
> 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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