[asterisk-bugs] [JIRA] (ASTERISK-27826) res_rtp_asterisk: DTLS negotiation fails when it should succeed, causing SRTP failure

[Sean Bright (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-27826?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=244819#comment-244819 ] 

Sean Bright edited comment on ASTERISK-27826 at 9/16/18 6:19 AM:
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[~mivanov at lanta-net.ru], please attach the following (As {{.txt}} files):

- {{config.log}} from your Asterisk build directory
- The output of {{apt list --installed}}
- The output of {{ldd /usr/lib/asterisk/modules/res_srtp.so}}

[~agupta], if you are on Debian/Ubuntu, please do the same.


was (Author: seanbright):
[~mivanov at lanta-net.ru], please attach the following:

- {{config.log}} from your Asterisk build directory
- The output of {{apt list --installed}}
- The output of {{ldd /usr/lib/asterisk/modules/res_srtp.so}}

[~agupta], if you are on Debian/Ubuntu, please do the same.

> res_rtp_asterisk: DTLS negotiation fails when it should succeed, causing SRTP failure
> -------------------------------------------------------------------------------------
>
>                 Key: ASTERISK-27826
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-27826
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Resources/res_rtp_asterisk
>    Affects Versions: 15.3.0
>            Reporter: Mikhail Ivanov
>            Assignee: Mikhail Ivanov
>              Labels: fax, pjsip, webrtc
>         Attachments: 1205-5191-01.pcap, 1205-5191-02.pcap, chrome_bad_call_log.txt, dump, fragment
>
>
> I have a problem with incoming (may be with outgoing too, not sure) calls to WebRTC clients (based on jssip.net library)
> Sometimes (2-5% of all incoming calls) I have no sound (on both sides) on incoming calls.
> RTP is going fine in both sides (local network)
> If I turn on mixMonitor on Asterisk, I can see only noise in call (looks like a problem with srtp keys, but not sure)
> https://www.dropbox.com/s/41nmwqhg0chcwl7/cf626000ac4601445d6cee3cd909188d.mp3?dl=1
> Asterisk 15.3.0, JsSIP 3.2.8, tested in Chrome, Chromium and Firefox
> If I turn off rtp encryption 
> webrtc = no 
> rtcp_mux = yes 
> use_avpf = yes 
> ice_support = yes 
> media_encryption = no
> and 
> --disable-webrtc-encryption in Chrome (Chromium)
> everything is fine, yes, it's workaround but not a solution



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