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

[Kevin Harwell (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-26428?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=232780#comment-232780 ] 

Kevin Harwell commented on ASTERISK-26428:
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[~djenkins], I was able to get the app installed and have it make a call. However, I don't see the error(s) you saw and actually don't seem to be getting any audio from it in the bridge. I want to make sure I have not missed something.

>From the app's webpage I click "call". I have it dialing into an extension that places it into an ARI app. From there I add the channel (A) to a mixing bridge, place another channel (B) in the bridge with it, and then hear no audio. I place a third channel (C) in the bridge and can hear audio between B and C.

As a side note, I can hear audio (what sounds like MOH) playing from the computer hosting your app. I'm assuming that is the audio that I should be hearing in the bridge? Is there something else I am missing?



> codec_opus: 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
>          Components: Codecs/codec_opus
>    Affects Versions: 14.0.1
>            Reporter: Dan Jenkins
>            Assignee: Kevin Harwell
>         Attachments: gistfile1.txt
>
>
> 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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