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

[Abhay Gupta (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-27826?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=247064#comment-247064 ] 

Abhay Gupta commented on ASTERISK-27826:
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I was checking one of the PCAP of 19th sep and on analysing on wireshark i saw that while chrome is saying 

Failed to unprotect RTP packet: size=172, seqnum=13388, SSRC=135391816

The data on wireshark is indeed a invalid data and is not even detected as a RTP and in fact the data is 

80:00:34:a2:18:81:88:40:08:11:ea:48:7e:65:74:ed:5d:5f:70:70:ed:ef:6c:6b:61:6b:ec:f2:f0:de:d7:eb:60:59:5e:6f:63:5a:78:e9:78:e1:f0:69:f6:f6:db:dc:ef:d6:fa:4d:54:58:fc:d0:d2:ce:de:6b:64:4f:59:72:50:45:48:58:6a:e5:f2:f6:cf:cd:cc:ce:cf:db:6a:66:58:5c:56:4f:5f:63:f2:eb:76:e1:dd:fa:78:6f:5e:7a:fc:6c:7c:6d:69:57:55:5c:5b:ed:db:d7:d4:dd:e2:ed:72:5b:5e:eb:e8:ed:e6:f0:7a:72:5e:74:e8:fe:7a:7c:78:5c:50:59:64:70:e2:de:fc:67:61:6a:70:fe:74:72:e5:e8:f6:72:5f:68:6f:f8:f6:7e:ec:f8:f6:f4:fa:7c:76:ef:ef:eb:fa:70:fe:69:5f:67:6f:f4

Now chrome is sending the right data but the recording shows that instead of decrypting , the encrypted data itself is recorded leading to noise in recording .

> 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, app_install_list.txt, asterisk_config_log.txt, asterisk-console-latest-call.log, b6-19-09-2018-asterisk-debug.log, b6-19-09-2018-asterisk-side.pcap, b6-19-09-2018-chrome-logs.log, b6-19-09-2018-chrome-side.pcap, bad_call.mp3, chrome_bad_call_log.txt, chrome-debug-latest-call.log, chrome-logs.txt, config.log, dump, dump.pcap, fragment, good_call.mp3, installed.txt, res_srtp.txt, res_srtp.txt, webrtc-at-asterisk-latest.pcap, webrtc-at-asterisk-latest-udp-only.pcap, webrtc-at-chrome-latest.pcap
>
>
> 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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