[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=232814#comment-232814 ] 

Kevin Harwell commented on ASTERISK-26428:
------------------------------------------

This appears to be a problem with the Asterisk implementation of opus and how it handles forward error correction (FEC) when decoding. In the invite the browser specifies to use in band fec in the sdp's fmtp line for opus:
{noformat}
a=fmtp:111 minptime=10;useinbandfec=1
{noformat}
Due to the nature of the problem for now we are going to specify the decoder to always not do fec. That should go out in the next release of the codec. However, it's planned that a future version will be able to properly handle fec.

In the meantime, to work around this for now specify to not use fec in the sdp (either set it to 0 or just don't include it as it will default to 0) and audio should work again.



> 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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