[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=247335#comment-247335 ] 

Abhay Gupta commented on ASTERISK-27826:
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Have found out that we do not get the following Debug line in case of call with a issue 

res_srtp.c: Adding new policy for SSRC

In successful call we have the following sequence 
1. On DailBegin we get 
DEBUG[24451] res_srtp.c: local_key64 rp9hxgtp0q3LwWW5XlNnnRgYNbuEHn0dA81+cs6e len 40
2. On DialEnd and answer we get
DEBUG[32672] res_srtp.c: Adding new policy for SSRC 400086938

On problem call we do not get the line Adding new policy as in point 2 but we do get local_key64 line .

> 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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