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

Ludovic Gasc (Eyepea) (JIRA) noreply at issues.asterisk.org
Thu Mar 15 16:29:13 CDT 2018


    [ https://issues.asterisk.org/jira/browse/ASTERISK-27624?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=242603#comment-242603 ] 

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

> We still have a delay (5-10 seconds) to pickup calls with incoming calls

In case of other people will face to this problem, the solution we have found is to reduce the ICE timeout at Web browser side.
With sip.js, it's with the parameter: iceCheckingTimeout.

You can keep this issue closed, we confirm it's clearly better to use chan_pjsip instead of chan_sip for WebRTC with Asterisk 15.

> 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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