[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=234982#comment-234982 ] 

Kevin Harwell edited comment on ASTERISK-26428 at 2/1/17 10:37 AM:
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A new version of codec_opus has not been released yet with the mentioned fix. However, in that time an appropriate handling of inband fec has been implemented in codec_opus. It will go out in the next release of the codec that should be out soon-ish.

[~rossbeer] In the negotiation between the g722->opus leg are you seeing "useinbandfec=1" in the sdp? If so then yeah probably the same issue. If not then it might be something else in which case a new issue should be created.


was (Author: kharwell):
A new version of codec_opus has not been released yet with the mentioned fix. However, in that time an appropriate handling of inband fec has been implemented in codec_opus. It will go out in the next release of the codec.

[~rossbeer] In the negotiation between the g722->opus leg are you seeing "useinbandfec=1" in the sdp? If so then yeah probably the same issue. If not then it might be something else in which case a new issue should be created.

> 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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