[asterisk-bugs] [JIRA] (ASTERISK-27624) WebRTC regression with Asterisk and Chrome 64 to receive calls

[Asterisk Team (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-27624?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=241809#comment-241809 ] 

Asterisk Team commented on ASTERISK-27624:
------------------------------------------

The module you are reporting the issue against is no longer supported as a core module but your issue is in the queue. Your patience is appreciated as a community developer may work the issue when time and resources become available.

Asterisk is an open source project and community members work the issues on a voluntary basis. You are welcome to develop your own patches and submit them to the project.[1]

If you are not a programmer and you are in a hurry to see a patch provided then you might try rallying support on the Asterisk users mailing list or forums.[2] Another alternative is offering a bug bounty on the asterisk-dev mailing list.[3] Often a little incentive can go a long way.

[1]: https://wiki.asterisk.org/wiki/display/AST/Patch+Contribution+Process
[2]: http://www.asterisk.org/community/discuss
[3]: https://wiki.asterisk.org/wiki/display/AST/Asterisk+Bug+Bounties

> WebRTC regression with Asterisk and Chrome 64 to receive calls
> --------------------------------------------------------------
>
>                 Key: ASTERISK-27624
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-27624
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Resources/res_rtp_asterisk
>    Affects Versions: 15.2.0
>         Environment: Debian 9 64 bits, libsrtp 2.0.0
>            Reporter: Ludovic Gasc (Eyepea)
>              Labels: webrtc
>
> Hi,
> With the new release of Chrome 64 two days ago, we have now a regression with Chrome 64+WebRTC+chan_sip, for the incoming calls on a WebRTC endpoint.
> When the webphone picks up the incoming call, Asterisk hangs up.
> We have this log message in Asterisk console:
> {quote}
> [Jan 25 20:04:52] ERROR[2431][C-00000006]: res_rtp_asterisk.c:2818 __rtp_recvfrom: DTLS failure occurred on RTP instance '0x7fe6140046b8' due to reason 'unexpected message', terminating
> [Jan 25 20:04:52] WARNING[2431][C-00000006]: res_rtp_asterisk.c:5802 ast_rtp_read: RTP Read error: Unspecified.  Hanging up.
> {quote}
> I have also the DTLS+STUN trace for this incoming call.
> It works pretty well with Firefox or an older version of Chrome.
> Do you think it's necessary to open an issue on the bug tracker of Chrome or it's something missing in Asterisk ?
> Do you need more details ?
> Thanks for your help.



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