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

Kevin Harwell (JIRA) noreply at issues.asterisk.org
Wed Nov 9 17:57:10 CST 2016


    [ https://issues.asterisk.org/jira/browse/ASTERISK-26478?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=233547#comment-233547 ] 

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

You can't do it via the dialplan, but you can configure it through conf files. For instance in codecs.conf something like the following needs exist:
{noformat}
[myopus]
type=opus
max_playback_rate=8000
bitrate=8000
max_bandwidth=narrow
{noformat}
And then in your sip.conf or pjsip.conf you'll need to specify "myopus" in the allow line for a given endpoint (note example snippet from a pjsip.conf file):
{noformat}
[109]
type=endpoint
aors=109
allow=!all,myopus
{noformat}
When the sdp gets negotiated between asterisk and that endpoint then the endpoint should hopefully limit sending data to that of the negotiated rates. 

> 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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