[asterisk-bugs] [JIRA] (ASTERISK-27624) WebRTC regression with Asterisk 15 and Chrome 64 to receive calls

[Ludovic Gasc (Eyepea) (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-27624?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=242383#comment-242383 ] 

Ludovic Gasc (Eyepea) commented on ASTERISK-27624:
--------------------------------------------------

I have tried again to migrate from chan_sip to chan_pjsip for WebRTC with Asterisk 15.3-rc1.
We had no audio with Asterisk 15.2+pjsip+Chrome 64, but with Asterisk 15.3-rc1 it's OK now.

We still have a delay (5-10 seconds) to pickup calls with incoming calls, we need to investigate more before to open an eventual issue: I don't know yet if it's a problem in the webphone or a real bug in Asterisk. But the situation is clearly better than before ;-)

For this bug itself, I will close.

> WebRTC regression with Asterisk 15 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)
>            Assignee: Unassigned
>              Labels: webrtc
>         Attachments: dtls_error_unexpected_message.pcapng.gz
>
>
> 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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