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https://issues.apache.org/jira/browse/OPENMEETINGS-544?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=13607730#comment-13607730
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Maxim Solodovnik commented on OPENMEETINGS-544:
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Just have checked that and it works as expected .... (tested on latest 2.1
build)
Not sure what else can I do :(
> some bug with SIP
> -----------------
>
> Key: OPENMEETINGS-544
> URL: https://issues.apache.org/jira/browse/OPENMEETINGS-544
> Project: Openmeetings
> Issue Type: Bug
> Components: VoIP/SIP
> Affects Versions: 2.1 Apache Release
> Environment: OpenMeetings REV 1449605
> Reporter: Sqli
> Assignee: Timur Tleukenov
>
> Hi,
> I have some problems with red5sip's configuration.
> The connection with asterisk works.
> I use "alaw" codec, and I can join conference with PhoneIP.
> 1 - Sip Dialer dialer doesn't work.
> 2 - Sip transport sometimes crashes and reconnects immediately.
> I have this error:
> " [WARN] o.r.s.n.r.BaseRTMPClientHandler: Stream data was null for client id:
> 3"
> " [ERROR] o.r.s.a.RTPStreamForMultiplex: Stream %d buffer overflow. Buffer is
> cleared"
> 3 - When I start red5sip I have this warning message.
> " [WARN ] n.s.e.c.ConfigurationFactory: No configuration found. Configuring
> ehcache from ehcache-failsafe.xml found in the classpath:
> jar:file:/opt/red5sip/lib/ehcache-2.2.0.jar!/ehcache-failsafe.xml"
> 4 - During a conference, the sound is correct but not very 'confortable':
> example: crackling, voice lag (1/2secondes delay)
> Do you have an "optimum" configuration to link openmeetings and asterisk?
> I would like to test with another codec, what codec can Red5sip support?
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