[asterisk-bugs] [JIRA] (ASTERISK-26428) No Audio when using Codec Opus from Chrome WebRTC Web Audio Stream

[Asterisk Team (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-26428?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=232468#comment-232468 ] 

Asterisk Team commented on ASTERISK-26428:
------------------------------------------

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].

> No Audio when using Codec Opus from Chrome WebRTC Web Audio Stream
> ------------------------------------------------------------------
>
>                 Key: ASTERISK-26428
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-26428
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>    Affects Versions: 14.0.1
>            Reporter: Dan Jenkins
>
> In terms of logging errors, Asterisk 99% of the time doesn't log that anything has gone wrong. However, one time, it did and heres the log
> https://gist.github.com/danjenkins/73bbabbbe06bab2a2b2c82131549fbc6
> I can't be sure if this would help or not, or if its even related (I can only presume it is)
> Scenario:
> Connect Chrome up using WebRTC to Asterisk and using a Web Audio Stream as the mediaStream into WebRTC, putting the extension into a bridge using ARI and other extens join from the PSTN (SRTP and RTP)
> I've used Asterisk 13.11.2 using the open source Opus patches available and everything works brilliantly.
> Upgraded to Asterisk 14 with Opus (core show channels shows opus on that channel) but no audio makes it into the bridge - I want to say that rtp debug on also doesn't show any data but with limited data connectivity while in the US is hampering testing this properly.
> I can help out with a basic scenario to reproduce this
> Like I say, everything worked perfectly with Opus on Ast 13.11.2 but not with Opus supplied with 14.0.1 



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