[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=233591#comment-233591 ] 

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

Hi Luke,

I took a look at the attached pcap. Maybe a few of odd things to note, but not sure if any would be the problem:

1. I never see the outgoing leg of the call, just the incoming. For instance for the call sequence starting at No. 6378 I see the initial Invite (phone->asterisk) and then the invite with auth (phone->asterisk). After that I see a 183 progress and it appears audio begins to flow between the phone and asterisk. Later Asterisk sends the 200 OK and then a BYE.

2. Around that same call sequence around No. 7054 I see an RTCP Goodbye coming from the phone. Meaning the source is no longer active. It's hard to tell if that is associated specifically with the above call sequence though. If it is then for some reason the phone source goes away, so no audio. (note it happens just after Asterisk sends the 200 OK).

3. In both the 183 and 200 OK sent from Asterisk the SDP contains an empty fmtp line for opus. This is a known issue (ASTERISK-26520) that perhaps the phone can't handle properly. That issue should be worked fairly soon so you may want to keep an eye on it for when the fix goes in to see if that helps.

Unfortunately, at this time I don't have much else to recommend until there is more to go on. Except I'd suggest trying to narrow the scope a bit and start with a simple scenario if possible. For instance one that does not involve nat. That way things can start being ruled out. Also if you have another endpoint (hard/soft phone, browser) that can do opus that might tell you something as well.

Hope that helps some.

> 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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