[OPENMEETINGS-544] some bug with SIP

• Type: Bug
• Status: Closed
• Priority: Major
• Resolution: Cannot Reproduce
• Affects Version/s: 2.1.0 Apache Release
• Fix Version/s: 2.1.1 Apache Release, 3.0.0
• Component/s: VoIP/SIP
• Environment: OpenMeetings REV 1449605
• Assignee: Maxim Solodovnik
• Reporter: Sqli
• Created: 26/Feb/13 16:56
• Updated: 14/Aug/13 07:06
• Resolved: 03/Jul/13 04:21

Description

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?