[asterisk-bugs] [JIRA] (ASTERISK-26478) codec_opus: Opus transcoding one-way audio issue

[Luke Escude (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-26478?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=233601#comment-233601 ] 

Luke Escude commented on ASTERISK-26478:
----------------------------------------

Kevin,

First off, I HIGHLY appreciate the work and time you've put into this, and I know my customers do too.

I'm able to see both legs of the call... Let me clarify the IP addresses for you.

207.91.156.217 --> My PBX server (testbed2.primevox.net)
97.77.119.234 --> My handset with Opus enabled, behind NAT (ubiquiti edgerouter with no SIP ALG, works flawlessly with uLaw).
216.115.69.114 --> The Flowroute SIP proxy (kamailio) - no media handled here
74.120.93.199 --> One of Flowroute's many media handling servers. Media travels here. uLaw forced.

> codec_opus: Opus transcoding one-way audio issue
> ------------------------------------------------
>
>                 Key: ASTERISK-26478
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-26478
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Codecs/codec_opus
>    Affects Versions: 14.0.2
>         Environment: Centos x64
>            Reporter: Luke Escude
>            Assignee: Unassigned
>            Severity: Minor
>         Attachments: 2out1in.pcap, Console-1.rtf, pcap0 3.pcap, PCAP2.pcap
>
>
> Are there any known issues with transcoding with the Opus codec from Digium?
> About 1 in every 4 handsets is having one-way audio problems on Opus, during outbound calls (handset -> opus -> Asterisk 14 -> uLaw -> Trunk)
> It could be the handsets themselves as well, since Grandstream only recently started supporting opus.



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