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

Luke Escude (JIRA) noreply at issues.asterisk.org
Wed Nov 2 17:53:10 CDT 2016


    [ https://issues.asterisk.org/jira/browse/ASTERISK-26478?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=233365#comment-233365 ] 

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

Trunk config:

username=12345
fromuser=12345
secret=password1
host=sip.flowroute.com
type=friend
context=from-trunk
insecure=port,invite
trustrpid=yes
sendrpid=yes
directmedia=no
qualify=yes
keepalive=45
nat=force_rport,comedia
dtmfmode=rfc2833
disallow=all
allow=ulaw
t38pt_udptl=yes,redundancy,maxdatagram=400
defaultexpiry=120
registertimeout=1
registerattempts=0

Peer Config:
[101]
secret=password2
dial=SIP/101
mailbox=101 at voicemail
callerid=101 <101>
deny=0.0.0.0/0.0.0.0
permit=0.0.0.0/0.0.0.0
disallow=all
allow=ulaw
dtmfmode=rfc2833
canreinvite=no
context=from-internal
host=dynamic
trustrpid=yes
sendrpid=pai
type=friend
nat=force_rport,comedia
port=5060
qualify=10000
qualifyfreq=15
transport=udp,tcp,tls
avpf=no
force_avp=no
icesupport=no
encryption=no
namedcallgroup=
namedpickupgroup=
callcounter=yes
faxdetect=no
cc_monitor_policy=generic

> 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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