[jira] [Closed] (OPENMEETINGS-544) some bug with SIP

["Maxim Solodovnik (JIRA)" <ji...@apache.org>]

     [ https://issues.apache.org/jira/browse/OPENMEETINGS-544?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel ]

Maxim Solodovnik closed OPENMEETINGS-544.
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> 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: 2.1.0 Apache Release
>         Environment: OpenMeetings REV 1449605
>            Reporter: Sqli
>            Assignee: Maxim Solodovnik
>             Fix For: 3.0.0 Apache Release, 2.1.1 Apache Release
>
>
>  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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