Having Problem in Running red5-debug.bat
I am trying to run red5-debug.bat using command line as directed in document of Apache OpenMeetings named as Installing OM2.x on Windows 7. But unable to run the script properly. I am seeing the following window: c:\red5red5-debug.bat c:\red5if NOT DEFINED RED5_HOME set RED5_HOME=c:\red5\ c:\red5set JAVA_OPTS=-Xdebug -Xrunjdwp:transport=dt_socket,address=8787,server= y,suspend=n c:\red5C:\red5\\red5.bat c:\red5SETLOCAL c:\red5if NOT DEFINED RED5_HOME set RED5_HOME=C:\red5\ c:\red5if NOT DEFINED RED5_MAINCLASS set RED5_MAINCLASS=org.red5.server.Bootstr ap c:\red5if NOT DEFINED JAVA_HOME goto err c:\red5REM JAVA options c:\red5REM You can set JAVA_OPTS to add additional options if you want c:\red5REM Set up logging options c:\red5set LOGGING_OPTS=-Dlogback.ContextSelector=org.red5.logging.LoggingConte xtSelector -Dcatalina.useNaming=true c:\red5REM Set up security options c:\red5REM set SECURITY_OPTS= -Djava.security.debug=failure -Djava.security.man ager -Djava.security.policy=C:\red5\/conf/red5.policy c:\red5set SECURITY_OPTS=-Djava.security.debug=failure c:\red5set JAVA_OPTS=-Dlogback.ContextSelector=org.red5.logging.LoggingContextS elector -Dcatalina.useNaming=true -Djava.security.debug=failure -Xdebug -Xrunjdw p:transport=dt_socket,address=8787,server=y,suspend=n c:\red5set JYTHON_OPTS=-Dpython.home=lib c:\red5set RED5_CLASSPATH=C:\red5\\boot.jar;C:\red5\\conf; c:\red5if NOT DEFINED RED5_OPTS set RED5_OPTS= c:\red5goto launchRed5 c:\red5echo Starting Red5 Starting Red5 c:\red5C:\Program Files\Java\jdk1.7.0_13\bin\bin\java -Dpython.home=lib -Dlog back.ContextSelector=org.red5.logging.LoggingContextSelector -Dcatalina.useNamin g=true -Djava.security.debug=failure -Xdebug -Xrunjdwp:transport=dt_socket,addre ss=8787,server=y,suspend=n -cp C:\red5\\boot.jar;C:\red5\\conf; org.red5.serve r.bootstrap The system cannot find the path specified. c:\red5goto finally c:\red5ENDLOCAL c:\red5 Please help me about it to run the script successfully. Thanks in advance.
Re: SIP connectivity
Dear all, i have tested the asterisk sip integration as documented with the most recent instruction (http://openmeetings.apache.org/red5sip-integration_2.1.html) and it works just fine. The only thing i am missing is a way to get this working when i choose to rename the openmeetings context from http://yourcorp.com:5080/openmeetings to http://yourcorp.com:5080/yourmeetings Which settings do i have to modify so that red5sip functions even if the context name is changed? Regards Sascha Naderi Von: Maxim Solodovnik [solomax...@gmail.com] Gesendet: Samstag, 9. Februar 2013 02:32 Bis: Bart Coninckx Cc: user Betreff: Re: SIP connectivity All tables are created by OM automatically On Feb 9, 2013 5:46 AM, Bart Coninckx bart.conin...@telenet.bemailto:bart.conin...@telenet.be wrote: May I add that a portion is missing, since one explains how to configure Asterisk for Realtime, but one does not stipulate how to create the necessary tables. It's in my CentOS docs however (which I hope to post shortly). BC On 01/31/13 13:05, Maxim Solodovnik wrote: Hello Bart, I just take a look at your URL ... OM does not create/use sipfriends DB table (at least from version 2.1) only meetme table is used so I'm afraid there is nothing to change here Here is the most recent instruction: http://openmeetings.apache.org/red5sip-integration_2.1.html Will ask our SIP guru to review it one more time :) On Thu, Jan 31, 2013 at 5:25 PM, Maxim Solodovnik solomax...@gmail.commailto:solomax...@gmail.com wrote: OK will add it and notify you On Jan 31, 2013 5:05 PM, Bart Coninckx bart.conin...@telenet.bemailto:bart.conin...@telenet.be wrote: It is for Asterisk 11 - don't know for other versions. You probably have no issues because of the 1.8 version. To be sure the .sql files in the Asterisk source should be compared across versions. this one is missing: `useragent` varchar(20) DEFAULT NULL, complete list (I think) is on: https://wiki.asterisk.org/wiki/display/AST/SIP+Realtime%2C+MySQL+table+structure If I bump into others, I'll report ASAP, BC On 01/31/13 06:21, Maxim Solodovnik wrote: Is the OM meetme table incomplete? My asterisk reports no issues :( could you provide me with missing fields and I'll add it. My purpose was to create table with required fields only. On Thu, Jan 31, 2013 at 4:45 AM, Bart Coninckx bart.conin...@telenet.bemailto:bart.conin...@telenet.be wrote: Openmeetings installed them for me, that's why I ended up with those. Using the Asterisk ones makes more sense to me. Maybe it's a good idea to have 'em removed from the install procedure. BC On 01/30/13 22:30, Jeff Clay wrote: Bart, If you look in the source directory of your asterisk tar file, under contrib/realtime/mysql you’ll find the .sql files required for all the realtime drivers. I never thought to use the ones with OM. Jeff Clay Network Administrator Infotech Enterprises America 870-215-5506 Ext. 1506 From: Bart Coninckx [mailto:bart.conin...@telenet.be] Sent: Wednesday, January 30, 2013 3:19 PM To: user@openmeetings.apache.orgmailto:user@openmeetings.apache.org Cc: Jeff Clay Subject: Re: SIP connectivity Well, I might have found one difference though: https://wiki.asterisk.org/wiki/display/AST/SIP+Realtime%2C+MySQL+table+structure dictates how the table should look like. I obviously used the one in the openmeetings mysql database, but this one seems to miss the table useragent. I discovered this because it showed up in the logfiles. BC On 01/29/13 14:41, Jeff Clay wrote: Bart, From an asterisk configuration standpoint there are very few differences between 1.8.x and 11.x. If memory serves, the only major changes that I ran into (in my production environment) was changes to SIP NAT values and the behavior of app_page() now uses confbridge instead of meetme to mix the audio. Also, TCP, TLS and app_confbridge got a major overhauling. There were of course many other changes and bug fixes, you can skim through the change log for full details, but I think that was the jist of it. Jeff Clay Network Administrator Infotech Enterprises America 870-215-5506 Ext. 1506 From: Bart Coninckx [mailto:bart.conin...@telenet.be] Sent: Tuesday, January 29, 2013 4:02 AM To: Maxim Solodovnik Cc: user Subject: Re: SIP connectivity I see - I'm willing to try the 11 version in the next fiew days if desired. BC On 01/29/13 10:57, Maxim Solodovnik wrote: I test the integration using Asterisk 1.8.13.1 (Ubuntu 12.10) On Tue, Jan 29, 2013 at 4:51 PM, Bart Coninckx bart.conin...@telenet.bemailto:bart.conin...@telenet.be wrote: That is amazing - I initially tried to do the same thing by using the new chan_motif driver in Asterisk 11 which connects to a XMPP server. Are you guys using Asterisk 11? This version is the newest LTS version and has the best video capabilities. Cheers, BC On 01/29/13 02:44, Maxim Solodovnik wrote: red5sip will create special OM user in
Re: SIP connectivity
please try red5sip rev. 76 it has additional parameter: om.context On Thu, Feb 14, 2013 at 2:59 AM, Naderi, Sascha snad...@datus.com wrote: Dear all, ** i have tested the asterisk sip integration as documented with the most recent instruction ( http://openmeetings.apache.org/red5sip-integration_2.1.html) and it worksjust fine . The only thing i am missing is a way to get this working when i choose to rename the openmeetings context from http://yourcorp.com:5080/openmeetings to http://yourcorp.com:5080/yourmeetings Which settings do i have to modify so that red5sip functions even if the context name is changed? ** ** Regards Sascha Naderi -- *Von:* Maxim Solodovnik [solomax...@gmail.com] *Gesendet:* Samstag, 9. Februar 2013 02:32 *Bis:* Bart Coninckx *Cc:* user *Betreff:* Re: SIP connectivity All tables are created by OM automatically On Feb 9, 2013 5:46 AM, Bart Coninckx bart.conin...@telenet.be wrote: May I add that a portion is missing, since one explains how to configure Asterisk for Realtime, but one does not stipulate how to create the necessary tables. It's in my CentOS docs however (which I hope to post shortly). BC On 01/31/13 13:05, Maxim Solodovnik wrote: Hello Bart, I just take a look at your URL ... OM does not create/use sipfriends DB table (at least from version 2.1) only meetme table is used so I'm afraid there is nothing to change here Here is the most recent instruction: http://openmeetings.apache.org/red5sip-integration_2.1.html Will ask our SIP guru to review it one more time :) On Thu, Jan 31, 2013 at 5:25 PM, Maxim Solodovnik solomax...@gmail.comwrote: OK will add it and notify you On Jan 31, 2013 5:05 PM, Bart Coninckx bart.conin...@telenet.be wrote: It is for Asterisk 11 - don't know for other versions. You probably have no issues because of the 1.8 version. To be sure the .sql files in the Asterisk source should be compared across versions. this one is missing: `useragent` varchar(20) DEFAULT NULL, complete list (I think) is on: https://wiki.asterisk.org/wiki/display/AST/SIP+Realtime%2C+MySQL+table+structure If I bump into others, I'll report ASAP, BC On 01/31/13 06:21, Maxim Solodovnik wrote: Is the OM meetme table incomplete? My asterisk reports no issues :( could you provide me with missing fields and I'll add it. My purpose was to create table with required fields only. On Thu, Jan 31, 2013 at 4:45 AM, Bart Coninckx bart.conin...@telenet.be wrote: Openmeetings installed them for me, that's why I ended up with those. Using the Asterisk ones makes more sense to me. Maybe it's a good idea to have 'em removed from the install procedure. BC On 01/30/13 22:30, Jeff Clay wrote: Bart, If you look in the source directory of your asterisk tar file, under contrib/realtime/mysql you’ll find the .sql files required for all the realtime drivers. I never thought to use the ones with OM. Jeff Clay Network Administrator Infotech Enterprises America 870-215-5506 Ext. 1506 *From:* Bart Coninckx [mailto:bart.conin...@telenet.bebart.conin...@telenet.be] *Sent:* Wednesday, January 30, 2013 3:19 PM *To:* user@openmeetings.apache.org *Cc:* Jeff Clay *Subject:* Re: SIP connectivity Well, I might have found one difference though: https://wiki.asterisk.org/wiki/display/AST/SIP+Realtime%2C+MySQL+table+structure dictates how the table should look like. I obviously used the one in the openmeetings mysql database, but this one seems to miss the table useragent. I discovered this because it showed up in the logfiles. BC On 01/29/13 14:41, Jeff Clay wrote: Bart, From an asterisk configuration standpoint there are very few differences between 1.8.x and 11.x. If memory serves, the only major changes that I ran into (in my production environment) was changes to SIP NAT values and the behavior of app_page() now uses confbridge instead of meetme to mix the audio. Also, TCP, TLS and app_confbridge got a major overhauling. There were of course many other changes and bug fixes, you can skim through the change log for full details, but I think that was the jist of it. Jeff Clay Network Administrator Infotech Enterprises America 870-215-5506 Ext. 1506 *From:* Bart Coninckx [mailto:bart.conin...@telenet.bebart.conin...@telenet.be] *Sent:* Tuesday, January 29, 2013 4:02 AM *To:* Maxim Solodovnik *Cc:* user *Subject:* Re: SIP connectivity I see - I'm willing to try the 11 version in the next fiew days if desired. BC On 01/29/13 10:57, Maxim Solodovnik wrote: I test the integration using Asterisk 1.8.13.1 (Ubuntu 12.10) On Tue, Jan 29, 2013 at 4:51 PM, Bart Coninckx bart.conin...@telenet.be wrote: That is amazing - I initially tried to do the same thing by using the new chan_motif driver in Asterisk
RE: GSoC project ideas wanted
Hi Sebastian, My idea for GSOC-2013 is to add recurrence events support for OpenMeetings calendar. Best regards, Irina. 2013/2/12 seba.wag...@gmail.com seba.wag...@gmail.com Hi Ed, thanks for your ideas. About idea no 1: That is an interesting idea, however it won't be possible that you provide a free to choose bandwidth for each user. The background is: Every stream that any client consumes has to be created somewhere. So what could be realized is that every stream that is broadcasted from one user via webcam to Red5/OpenMeetings will be re-transcoded into multiple streams (high, middle, low) bandwidth. So there might be some limitations to that: - high quality will never be better then the original material. We can't make a picture better then the original. So all re-transcoding will only make the original to lower quality, never to higher. - Re-transcoding has to happen on the server side (and number of streams are limited, we can't provide a stream on the required bandwidth on-demand for each user, or only with very big effort) - it will require real-time transcoding on server side which is possible with FFMPEG and some integration into Red5. But we would need a very specialized student that is keen and very motiviated as there is hardly any documentation on that available in the internet. What a project makes a success is if all participant know the potential outcome and the tools and methods that are needed to realize that. I would be happy to put this project on our list but it will be difficult to find somebody with the needed skills. Sebastian 2013/2/12 BBS Technik dormiti...@gmx.de Hi all, I think, one of the gratest liminations for satisfactory video conferencing with om is the limited bandwidth of internet connections of the clients . Therefore I would like to suggest the following ideas for a GSoC project : 1. The image size of the videos transferred from the om server to the clients should be adapted to the video window size set in the recipient client. Thus the recipient client itself could influence the transferred amount of data to it. Then all the participants achieve the best possible result for them. 2. A second proposal concerns that the screensharing bandwidth requirements has an great impact on the overall quality of the video conference. Here, in a project the existing function of sreen sharing could be expanded and enhanced. For example, the possibility for the transfer on only one application window, regardless of its size. Or the possibility of shared browsing with a locally installed browser. Moreover, certainly an improvement of the used compression method would be a very good project topic. I would be happy if the subject of bandwidth consumption would plays a role in the selected GSoC project . Best regards Ed Original-Nachricht Datum: Tue, 12 Feb 2013 08:44:54 +1300 Von: seba.wag...@gmail.com seba.wag...@gmail.com An: dev d...@openmeetings.apache.org CC: user@openmeetings.apache.org Betreff: GSoC project ideas wanted Google Summer of Code is about to start soon! Google sponsors every student with 4500USD. Plus 500 for the Apache Foundation. We are searching for ideas what porential students can do. Ideas from Non-Developers are welcome too! We will add the ideas to JIRA then with a special label so students can find it. Sebastian -- Sebastian Wagner https://twitter.com/#%21/dead_lock https://twitter.com/#!/dead_lock http://www.webbase-design.de http://www.webbase-design.de http://www.wagner-sebastian.com http://www.wagner-sebastian.com mailto:seba.wag...@gmail.com seba.wag...@gmail.com -- Sebastian Wagner https://twitter.com/#%21/dead_lock https://twitter.com/#!/dead_lock http://www.webbase-design.de http://www.webbase-design.de http://www.wagner-sebastian.com http://www.wagner-sebastian.com mailto:seba.wag...@gmail.com seba.wag...@gmail.com -- Daniel Ascher, M.Ed. President A+ Test Prep and Tutoring Creating Bright Futures 505 York Road, Suite 6 Jenkintown, PA 19046 Office: 215.886.9188 Direct: 267.242.9640 www.aplustutoring.com http://www.aplustutoring.com/ Image removed by sender. image001.jpg
Re: Please Help to Install OM
Yes following the instructions of https://cwiki.apache.org/confluence/display/OPENMEETINGS/Tutorials+for+installing+OpenMeetings+and+Tools I completed all the 19 steps now at Step 20 and trying to run the script red5-debug.bat but viewing the error like: Listening for transport dt_socket at address: 8787 Error: Could not find or load main class org.red5.server.bootstrap What's it and how to resolve? From: George Kirkham gkirk...@co2crc.com.au To: user@openmeetings.apache.org Sent: Thursday, February 14, 2013 10:56 AM Subject: RE: Please Help to Install OM Aysha, Are you using Stephen Cottham’s installation instructions from; https://cwiki.apache.org/confluence/display/OPENMEETINGS/Tutorials+for+installing+OpenMeetings+and+Tools Step 20: Running OM 2.X for the first time Locate this folder C:\red5, and run the script called “red5-debug.bat” Now open the browser and go to the following link. http://%3cyourIP%3e:5080/openmeetings/install Thanks, George Kirkham From:Aysha Saddiqa [mailto:ayshasadd...@yahoo.com] Sent: Thursday, 14 February 2013 4:41 PM To: User Open Meeting Subject: Please Help to Install OM Please help me in installation of OM 2.x.
Having Problem in Installation
I am following the instructions of https://cwiki.apache.org/confluence/display/OPENMEETINGS/Tutorials+for+installing+OpenMeetings+and+Tools I completed all the 19 steps now at Step 20 and trying to run the script red5-debug.bat but viewing the error like: Listening for transport dt_socket at address: 8787 Error: Could not find or load main class org.red5.server.bootstrap What's it and how to resolve?
AW: SIP connectivity
Dear Jeff, dear all, not the asterisk but the red5sip. I get the following error message from red5sip after the directory name (url) of openmeetings was changed. 13 Feb 08:29:39 - [INFO ] o.r.s.n.r.BaseRTMPClientHandler: rtmp://127.0.0.1:1935/openmeetings/0 13 Feb 08:29:39 - [INFO ] o.r.s.n.r.c.RTMPProtocolDecoder: Action _result 13 Feb 08:29:39 - [INFO ] o.r.s.a.RTMPControlClient: service call result: Service: null Method: connect No params 13 Feb 08:29:39 - [INFO ] o.r.s.a.RTMPControlClient: connect 13 Feb 08:29:39 - [ERROR] o.r.s.n.r.BaseRTMPHandler: Error while executing callback org.red5.sip.app.Application$2@3debe8abmailto:org.red5.sip.app.Application$2@3debe8ab java.lang.IllegalThreadStateException 13 Feb 08:29:39 - [WARN ] o.r.s.n.r.RTMPMinaIoHandler: Exception caught Connection reset by peer 13 Feb 08:29:39 - [INFO ] o.r.s.a.RTMPControlClient: service call result: Service: null Method: getActiveRoomIds No params 13 Feb 08:29:39 - [INFO ] o.r.s.a.RTMPControlClient: getActiveRoomIds Regards Sascha Von: Jeff Clay [jeff.c...@infotech-enterprises.com] Gesendet: Mittwoch, 13. Februar 2013 21:02 Bis: user@openmeetings.apache.org Cc: Maxim Solodovnik [solomax...@gmail.com] Betreff: RE: SIP connectivity I do not believe that the asterisk context is related to the url of openmeetings. Jeff Clay Network Administrator Infotech Enterprises America 870-215-5506 Ext. 1506 From: Naderi, Sascha [mailto:snad...@datus.com] Sent: Wednesday, February 13, 2013 2:00 PM To: user@openmeetings.apache.org Cc: Maxim Solodovnik [solomax...@gmail.com] Subject: Re: SIP connectivity Dear all, i have tested the asterisk sip integration as documented with the most recent instruction (http://openmeetings.apache.org/red5sip-integration_2.1.html) and it works just fine. The only thing i am missing is a way to get this working when i choose to rename the openmeetings context from http://yourcorp.com:5080/openmeetings to http://yourcorp.com:5080/yourmeetings Which settings do i have to modify so that red5sip functions even if the context name is changed? Regards Sascha Naderi Von: Maxim Solodovnik [solomax...@gmail.com] Gesendet: Samstag, 9. Februar 2013 02:32 Bis: Bart Coninckx Cc: user Betreff: Re: SIP connectivity All tables are created by OM automatically On Feb 9, 2013 5:46 AM, Bart Coninckx bart.conin...@telenet.bemailto:bart.conin...@telenet.be wrote: May I add that a portion is missing, since one explains how to configure Asterisk for Realtime, but one does not stipulate how to create the necessary tables. It's in my CentOS docs however (which I hope to post shortly). BC On 01/31/13 13:05, Maxim Solodovnik wrote: Hello Bart, I just take a look at your URL ... OM does not create/use sipfriends DB table (at least from version 2.1) only meetme table is used so I'm afraid there is nothing to change here Here is the most recent instruction: http://openmeetings.apache.org/red5sip-integration_2.1.html Will ask our SIP guru to review it one more time :) On Thu, Jan 31, 2013 at 5:25 PM, Maxim Solodovnik solomax...@gmail.commailto:solomax...@gmail.com wrote: OK will add it and notify you On Jan 31, 2013 5:05 PM, Bart Coninckx bart.conin...@telenet.bemailto:bart.conin...@telenet.be wrote: It is for Asterisk 11 - don't know for other versions. You probably have no issues because of the 1.8 version. To be sure the .sql files in the Asterisk source should be compared across versions. this one is missing: `useragent` varchar(20) DEFAULT NULL, complete list (I think) is on: https://wiki.asterisk.org/wiki/display/AST/SIP+Realtime%2C+MySQL+table+structure If I bump into others, I'll report ASAP, BC On 01/31/13 06:21, Maxim Solodovnik wrote: Is the OM meetme table incomplete? My asterisk reports no issues :( could you provide me with missing fields and I'll add it. My purpose was to create table with required fields only. On Thu, Jan 31, 2013 at 4:45 AM, Bart Coninckx bart.conin...@telenet.bemailto:bart.conin...@telenet.be wrote: Openmeetings installed them for me, that's why I ended up with those. Using the Asterisk ones makes more sense to me. Maybe it's a good idea to have 'em removed from the install procedure. BC On 01/30/13 22:30, Jeff Clay wrote: Bart, If you look in the source directory of your asterisk tar file, under contrib/realtime/mysql you’ll find the .sql files required for all the realtime drivers. I never thought to use the ones with OM. Jeff Clay Network Administrator Infotech Enterprises America 870-215-5506 Ext. 1506 From: Bart Coninckx [mailto:bart.conin...@telenet.be] Sent: Wednesday, January 30, 2013 3:19 PM To: user@openmeetings.apache.orgmailto:user@openmeetings.apache.org Cc: Jeff Clay Subject: Re: SIP connectivity Well, I might have found one difference though: https://wiki.asterisk.org/wiki/display/AST/SIP+Realtime%2C+MySQL+table+structure dictates how the
