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

[Dan Jenkins (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-26428?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=232530#comment-232530 ] 

Dan Jenkins commented on ASTERISK-26428:
----------------------------------------

Asterisk 13 git branch - no audio, rtp set debug on shows receiving packets

Got  RTP packet from    <ip>:61476 (type 111, seq 029908, ts 1136137639, len 000003)

core show channel <id> shows 
NativeFormats: (opus)
    WriteFormat: opus
     ReadFormat: slin
 WriteTranscode: No
  ReadTranscode: Yes (opus at 48000)->(slin at 48000)->(slin at 8000)

After removing the other codecs which are now installable through the same means so that its exactly the same as my asterisk 13.11.2 install in terms of configing install, same result.


For some reason I now can't get my asterisk 13.11.2 thats patched with the other version of the opus codec to install/compile. Keeps failing on pjproject but I'll figure that out in a bit - but yes basically, same issue with 13 with the digium opus codec as 14

> 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: Unassigned
>         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 



--
This message was sent by Atlassian JIRA
(v6.2#6252)