[asterisk-bugs] [JIRA] (ASTERISK-25265) DTLS Failure when calling WebRTC-peer on Firefox 39

Mark Duncan (JIRA) noreply at issues.asterisk.org
Fri Jul 24 02:29:32 CDT 2015


    [ https://issues.asterisk.org/jira/browse/ASTERISK-25265?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=227028#comment-227028 ] 

Mark Duncan edited comment on ASTERISK-25265 at 7/24/15 2:28 AM:
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I can confirm this in asterisk-13.1-cert2 (with the small patch to fix the DTLS handshake on newer OpenSSL versions manually applied).  I have tried with Firefox and Firefox Developer Edition and both show the exact same issue.  An incoming call is immediately dropped when attempting to answer.  Chrome works as expected.  The error shown in the Asterisk console is as follows

`[Jul 24 07:24:48] ERROR[1336]: res_rtp_asterisk.c:2042 __rtp_recvfrom: DTLS failure occurred on RTP instance '0x7f77bc007888' due to reason 'no shared cipher', terminating`


was (Author: mwduncan):
I can confirm this in asterisk-13.1-cert2 (with the small patch to fix the DTLS handshake on newer OpenSSL versions manually applied).  I have tried with Firefox and Firefox Developer Edition and both show the exact same issue.  Chrome works as expected. 

> DTLS Failure when calling WebRTC-peer on Firefox 39 
> ----------------------------------------------------
>
>                 Key: ASTERISK-25265
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-25265
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>    Affects Versions: 13.1.0
>            Reporter: Stefan Engström
>
> This issue has already been reported by http://forums.asterisk.org/viewtopic.php?f=1&t=95417
> Whenever calling a webrtc peer which uses firefox version 39 (or 40 beta), I get  error messages like "res_rtp_asterisk.c: DTLS failure occurred on RTP instance '0x7fefe800e9e8' due to reason 'no shared cipher', terminating" after the SDP exchange, and the call terminates.
> Hopefully you can reproduce it yourself on the latest version of asterisk by using
> http://www.sipml5.org/call.htm (I'm not sure if asterisk is doing anything wrong or just firefox/sipml5)
> I will provide more info if it's not easily reproducable.



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