[asterisk-bugs] [JIRA] (ASTERISK-25659) DTLS failure occurred on RTP instance
Edwin Vandamme (JIRA)
noreply at issues.asterisk.org
Tue Jan 5 08:16:33 CST 2016
[ https://issues.asterisk.org/jira/browse/ASTERISK-25659?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=228830#comment-228830 ]
Edwin Vandamme commented on ASTERISK-25659:
-------------------------------------------
Joshua,
Just to make sure I also tried on a Windows 8 with Chrome, but to no avail.
Could it be in the generation of the certificate request ?
# Create CSR and key file.
openssl req -new -newkey rsa:4096 -nodes -out ${KFN}/${KNM}.csr -keyout ${KFN}/${KNM}.key -subj "/C=${CO}/ST=${ST}/L=${LO}/O=${OR}/OU=${OU}/CN=${CN}/emailAddress=${EM}"
${} values are the corresponding parameter values.
> DTLS failure occurred on RTP instance
> -------------------------------------
>
> Key: ASTERISK-25659
> URL: https://issues.asterisk.org/jira/browse/ASTERISK-25659
> Project: Asterisk
> Issue Type: Bug
> Security Level: None
> Affects Versions: 13.6.0
> Environment: Using the following on the server :
> CentOS 7.2 2015-11
> Asterisk 13.6 2015-10
> jansson 2.7 2014-10-02
> PJSIP (pjproject) 2.4.5 2015-08-12
> sipML5 2.0.2 2015-12
> Using the following on the client :
> CentOS 7.2 KDE desktop
> Chrome Version 47.0.2526.106 (64-bit)
> Reporter: Edwin Vandamme
> Assignee: Unassigned
> Severity: Minor
> Attachments: asterisk.log
>
>
> This issue has been on the forum for over a week, but I did not get any feedback, http://forums.asterisk.org/viewtopic.php?f=1&t=96461&sid=528c724d236a38e60e868817462c6f26, so I have now escalated this as a bug report.
> Using the described environment, I get the following error in my Asterisk log :
> res_rtp_asterisk.c: DTLS failure occurred on RTP instance '0x7fe8c8024178' due to reason 'missing tmp ecdh key', terminating
> res_rtp_asterisk.c: RTP Read error: Unspecified. Hanging up.
> An earlier bug report listed this as a problem on FireFox : https://issues.asterisk.org/jira/browse/ASTERISK-25265
> It is said to be fixed in 13.6
> WebRTC is not yet in production on my system, due to the constant changes, but in earlier tests everything worked fine. As far as I can tell, it all started when Chrome forced the usage of https over http.
> Dialing from a WebRTC peer to Asterisks works just fine.
> For various reasons I use sip.conf, not pjsip.conf.
> Certificates used are propper certificates, not self signed versions.
> I attached (asterisk.log) part of the Asterisk log file with "sip debug on", start of call till failure.
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