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

Luke Escude (JIRA) noreply at issues.asterisk.org
Thu Nov 10 12:56:10 CST 2016


    [ https://issues.asterisk.org/jira/browse/ASTERISK-26478?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=233577#comment-233577 ] 

Luke Escude edited comment on ASTERISK-26478 at 11/10/16 12:55 PM:
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You can ignore the stream with 1001 and 875, those are a result of us taking down the firewall for testing.

There's definitely some weirdness going on. You can tell the server hears both sides of the call, but the Opus handset simply isn't receiving any audio. This only happens on outbound calls.

It should be using 48,000 Opus since that's all we support.

If you would like access to the cloud PBX, I can give you a login for the web UI and give you SSH access to get into the dial plan and whatnot.


was (Author: lukeescude):
You can ignore the stream with 1001 and 875, those are a result of us taking down the firewall for testing.

There's definitely some weirdness going on. You can tell the server hears both sides of the call, but the Opus handset simply isn't receiving any audio. This only happens on outbound calls.

It should be using 48,000 Opus since that's all we support.

> 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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