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

[Dan Jenkins (JIRA)]

Dan Jenkins created ASTERISK-26428:
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             Summary: 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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