[asterisk-bugs] [JIRA] (ASTERISK-27636) One way audio on Chrome 64 webrtc chan_sip
Asterisk Team (JIRA)
noreply at issues.asterisk.org
Mon Jan 29 09:58:13 CST 2018
[ https://issues.asterisk.org/jira/browse/ASTERISK-27636?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=241882#comment-241882 ]
Asterisk Team commented on ASTERISK-27636:
------------------------------------------
Thanks for creating a report! The issue has entered the triage process. That means the issue will wait in this status until a Bug Marshal has an opportunity to review the issue. Once the issue has been reviewed you will receive comments regarding the next steps towards resolution.
A good first step is for you to review the [Asterisk Issue Guidelines|https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines] if you haven't already. The guidelines detail what is expected from an Asterisk issue report.
Then, if you are submitting a patch, please review the [Patch Contribution Process|https://wiki.asterisk.org/wiki/display/AST/Patch+Contribution+Process].
> One way audio on Chrome 64 webrtc chan_sip
> ------------------------------------------
>
> Key: ASTERISK-27636
> URL: https://issues.asterisk.org/jira/browse/ASTERISK-27636
> Project: Asterisk
> Issue Type: Bug
> Security Level: None
> Components: Channels/chan_sip/General
> Affects Versions: 13.19.0
> Environment: Ubuntu 16.04
> Reporter: Sebastian Gutierrez
> Labels: webrtc
>
> After updating chrome to 64 (didn't happen on 63) we are having one way audio for an inbound call, does not happen on outbound calls.
> WebRTC, JSSIP
> I think may be related to some issue on the sdp, I will attach now the sip debug client side that is possible to see one with a chrome 63 that works and the other with 64 that wont work ok, I will upload a server side log
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