[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=233363#comment-233363 ] 

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

Very interesting ideas... I'll pull dial plan right now.

Until then, here are a couple more packet captures:

http://provisioning.primevox.net/flowroute-256788-cap2.pcap
Toward the end, extension 102 tries to call 817-233-1397. Time: 5617.926352

http://provisioning.primevox.net/flowroute-256788-cap1.pcap
"Scott <102>" to "2143954681" start time: 4515.353396.
You'll see that Hardenbrook "2143954681" called him back after, and the audio was fine.

I'll pull dialplan now. This issue seems to happen more on outbound calls instead of inbound calls.

> 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: 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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