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

[Luke Escude (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-26478?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=233367#comment-233367 ] 

Luke Escude commented on ASTERISK-26478:
----------------------------------------

Dial into an extension:

exten => 101,hint,SIP/101
same => 1,Answer
same => n,GotoIf($["${CURL("127.0.0.1/blacklist_check.php?user=5804f58837462&num=${CALLERID(num)}")}" != "1"]?noblock)
same => n,Busy
same => n,Hangup
same => n(noblock),noop
same => n,GoSub(recordcheck,s,1(${CALLERID(num)},101))
same => n,Set(CHANNEL(musicclass)=0)
same => n,GotoIf($["${CDR(cdr_direction)}" != ""]?keepDir)
same => n,Set(CDR(cdr_direction)=3)
same => n(keepDir),Set(CDR(cdr_internal_from_extension)=${CALLERID(num)})
same => n,Set(CDR(cdr_internal_to_extension)=101)
same => n,Set(CDR(cdr_application)=Dial)
same => n,Dial(SIP/101,20,W)
same => n,GotoIf($[${HANGUPCAUSE}=17]?skipA)
same => n,Set(__unavailable=u)
same => n,Goto(skipB)
same => n(skipA),Set(__unavailable=b)
same => n(skipB),noop(HANGUPCAUSE=${HANGUPCAUSE})
same => n,GotoIf($["${DIALSTATUS}" = "NOANSWER"]?101-NOANSWER,1)
same => n,GotoIf($["${DIALSTATUS}" = "BUSY"]?101-BUSY,1)
same => n,GotoIf($["${DIALSTATUS}" = "CHANUNAVAIL"]?101-CHANUNAVAIL,1)
same => n,Goto(101-CHANUNAVAIL,1)
exten => 101-CALLWAITING,1,Goto(*0101,1)
exten => 101-NOANSWER,1,Goto(*0101,1)
exten => 101-BUSY,1,Goto(*0101,1)
exten => 101-CHANUNAVAIL,1,Goto(*0101,1)
exten => _+XXXXXX./101,1,Goto(dial-out-101,${EXTEN:1},1)
exten => _XXXXXX./101,1,Goto(dial-out-101,${EXTEN},1)
exten => 911/101,1,Goto(dial-out-101,${EXTEN},1)
exten => 9911/101,1,Goto(dial-out-101,${EXTEN},1)
exten => **101,1,Pickup(101)
exten => *97/101,hint,Custom:vm101
same => 1,Set(CDR(cdr_application)=Voicemail)
same => n,GotoIf($["${CDR(cdr_direction)}" != ""]?keepDir)
same => n,Set(CDR(cdr_direction)=3)
same => n(keepDir),Set(CDR(cdr_internal_from_extension)=${CALLERID(num)})
same => n,Set(CDR(cdr_internal_to_extension)=*97/101)
same => n,StopMixMonitor
same => n,VoiceMailMain(101 at voicemail,s)

> 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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