[asterisk-bugs] [JIRA] (ASTERISK-25659) DTLS failure occurred on RTP instance

[Joshua Colp (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-25659?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=228842#comment-228842 ] 

Joshua Colp commented on ASTERISK-25659:
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There is logic in configure.ac which determines whether the underlying function is available. It assumes that if the function is available from OpenSSL then it can be used. As to why the OpenSSL function isn't working in your case I have no idea. I'd focus on that instead, as it should be.

> 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 : 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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