[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=243678#comment-243678 ] 

Sean Bright edited comment on ASTERISK-27826 at 9/15/18 2:48 PM:
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I think we're also going to need the browser side as the trace and Asterisk side doesn't provide the full picture and the needed information to understand precisely what is going on. Passing the following arguments to Chrome will spit out all the WebRTC debugging and may provide the information, please attach it as a text file:

{{--enable-logging --v=1 --vmodule=\*/webrtc/\*=2,\*=-2 --enable-logging=stderr}}


was (Author: jcolp):
I think we're also going to need the browser side as the trace and Asterisk side doesn't provide the full picture and the needed information to understand precisely what is going on. Passing the following arguments to Chrome will spit out all the WebRTC debugging and may provide the information, please attach it as a text file:

--enable-logging --v=1 --vmodule=*/webrtc/*=2,*=-2 --enable-logging=stderr

> 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: Unassigned
>              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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