[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=233642#comment-233642 ] 

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

{quote}
Also what are RTP checksums, and should they be enabled?
{quote}
I believe it used for checking if there are possible errors in a packet. From what I can tell it appears this is handled at the socket layer (i.e. the kernel checks to make sure the incoming packet checksum is equal to the calculated value). I'd gather that you would usually have this enabled.

However, since it can be optionally disabled there are probably some use cases where it would make sense to do so. That of course all depends on the type of problem one is having (firewall/NATs messing up the checksum potentially, not sure though).

> 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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