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

[Aleksei Kulakov (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-24143?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=221557#comment-221557 ] 

Aleksei Kulakov edited comment on ASTERISK-24143 at 8/12/14 4:07 AM:
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>>I don't see any endpoints configured for a websocket transport in your config.
You're wrong, 354 and 355 configured as webrtc endpoints using [endpoint-web] template, 6001 configured for sip-softphone using [endpoint-std] template.

*Reproduced with same results using DTLS(at aster and chrome sides) on latest Chrome stable and unstable(38.0.2114.2 dev)*

PS: You guys don't intended to support(work on) "no encryption" case for webrtc? Or you just want to be sure that it is not encryption related?




was (Author: each):
>>I don't see any endpoints configured for a websocket transport in your config.
You're wrong, 354 and 355 configured as webrtc endpoints using [endpoint-web] template, 6001 configured for sip-softphone using [endpoint-std] template.

Same results with DTLS enabled(at aster and chrome sides) on latest Chrome stable and unstable(38.0.2114.2 dev).

PS: You guys don't intended to support(work on) "no encryption" case for webrtc? Or you just want to be sure that it is not encryption related?



> 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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