[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=244834#comment-244834 ] 

Sean Bright edited comment on ASTERISK-27826 at 9/17/18 2:54 PM:
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I have tried _at least_ 100 calls and have been unable to reproduce, so this is what we need to move this forward:

# Packet capture with all traffic
# Asterisk console output with debug at level 9 and {{pjsip set logger on}} _- We need to be able to see all of the signalling and a pcap is not going to show us that_
# chrome_debug.txt

All 3 of these need to include the same failing call in order to be able to correlate the information between them.

Additionally - this should be done against either the Asterisk 15 branch from Git, or the latest Asterisk 15 release which is 15.6.0. If you cannot upgrade to that, please indicate the exact output of {{core show version}}.


was (Author: seanbright):
I have tried _at least_ 100 calls and have been unable to reproduce, so this is what we need to move this forward:

# Packet capture with all traffic
# Asterisk console output with debug at level 9 and {{pjsip set logger on}} _- We need to be able to see all of the signalling and a pcap is not going to show us that_
# chrome_debug.txt

All 3 of these need to include the same failing call in order to be able to correlate the information between them.

Additionally - this should be done against the latest Asterisk 15 release which is 15.6.0. If you cannot upgrade to that, please indicate the exact output of {{core show version}}.

> 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, app_install_list.txt, asterisk_config_log.txt, bad_call.mp3, chrome_bad_call_log.txt, chrome-logs.txt, config.log, dump, dump.pcap, fragment, good_call.mp3, installed.txt, res_srtp.txt, res_srtp.txt
>
>
> 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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