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

[Kevin Harwell (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-26478?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=233391#comment-233391 ] 

Kevin Harwell commented on ASTERISK-26478:
------------------------------------------

Unfortunately nothing stands directly out in your dialplan or config to me. Looking at the pcaps you linked however I did notice in flowroute-256788-cap2.pcap the media between the mentioned call seems to be off. For instance call sequence starting at No. 21084:
{noformat}
Invite - source ip = 76.186.126.55 dest ip = 206.126.62.152 media ip = 10.20.74.143
183 Session Progress and 200 OK - source ip = 206.126.62.152 dest ip = 76.186.126.55 media = 206.126.62.152
{noformat}
If I understand correctly media should flow between 10.20.74.143 and 206.126.62.152. After the 183 Session Progress (No. 21306) media appears to flow between those two addresses for a few frames. Then it appears to swap to flowing between 76.186.126.55 and 206.126.62.152 after the phone sends media to asterisk for some reason even though it negotiated a different media address.

Something to try if you haven't already to potentially narrow it down to the phone: Does the problem occur with other phones? Try swapping the phone out for the same endpoint and see if the problem still happens. Also you may investigate why the media address differs from the phone in the initial invite (but that all depends on your setup).


> 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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