Last 30 days

• Dec 05, 2009

  issue 89 (audio stops after 2 seconds) reported by vipkilla   -   hi, i've successfully installed red5phone on my linux box and it seems to work. the only problem is when i make a call the sound stops working after about 2 seconds. has anyone else had this problem? what could be causing this to happen?

  hi, i've successfully installed red5phone on my linux box and it seems to work. the only problem is when i make a call the sound stops working after about 2 seconds. has anyone else had this problem? what could be causing this to happen?
• Dec 04, 2009

  issue 88 (addToConf method not used) reported by tmlder   -   why method addToConf is not used in the Application.java and not caused from Flex-application?

  why method addToConf is not used in the Application.java and not caused from Flex-application?
• Dec 04, 2009

  issue 87 (How do red5phone worked in audio conferences) reported by tmlder   -   Hello! Tell me please, in general terms, how you can change red5phone enable it to serve audio conference? red5phone uses the library mjsip. But the library mjsip does not support the JOIN header, which is used to establish audio conferences! Maybe instead mjsip use another library or write your own work with the JOIN header? Please advise if. Thanks excuse for my english

  Hello! Tell me please, in general terms, how you can change red5phone enable it to serve audio conference? red5phone uses the library mjsip. But the library mjsip does not support the JOIN header, which is used to establish audio conferences! Maybe instead mjsip use another library or write your own work with the JOIN header? Please advise if. Thanks excuse for my english
• Nov 25, 2009

  issue 86 (Call Disconnects after 2 minutes) reported by costint   -   What steps will reproduce the problem? 1. Using Red5/Red5Phone to connect to TrixBox 2. Make the call using my setup 3. After almost exactly 2 minutes the call is disconnected by the TrixBox What is the expected output? What do you see instead? The call should not drop, but it's actually dropping. What version of the product are you using? On what operating system? Using Red5 0.8.0 with Red5Phone rev 39 from SVN on Windows 2003 server. Please provide any additional information below. On the TrixBox side, I have SIP Debugging on and I see this error: [Nov 25 19:43:08] VERBOSE[27663] logger.c: Scheduling destruction of SIP dialog '906880337308@10.248.90.9' in 32000 ms (Method: REGISTER) [Nov 25 19:43:40] VERBOSE[27663] logger.c: Really destroying SIP dialog '906880337308@10.248.90.9' Method: REGISTER I think that Red5Phone doesn't provide some type of answer that the TrixBox is looking for to keep the call active. Any assistance with this would be greatly appreciated. Thank you!

  What steps will reproduce the problem? 1. Using Red5/Red5Phone to connect to TrixBox 2. Make the call using my setup 3. After almost exactly 2 minutes the call is disconnected by the TrixBox What is the expected output? What do you see instead? The call should not drop, but it's actually dropping. What version of the product are you using? On what operating system? Using Red5 0.8.0 with Red5Phone rev 39 from SVN on Windows 2003 server. Please provide any additional information below. On the TrixBox side, I have SIP Debugging on and I see this error: [Nov 25 19:43:08] VERBOSE[27663] logger.c: Scheduling destruction of SIP dialog '906880337308@10.248.90.9' in 32000 ms (Method: REGISTER) [Nov 25 19:43:40] VERBOSE[27663] logger.c: Really destroying SIP dialog '906880337308@10.248.90.9' Method: REGISTER I think that Red5Phone doesn't provide some type of answer that the TrixBox is looking for to keep the call active. Any assistance with this would be greatly appreciated. Thank you!
• Nov 24, 2009

  issue 84 (How can you connect to a SIP phone without having a micropho...) commented on by lior.herman777   -   hello Tony when you wrote connect to sip using flex - what do you mean by that? where do you register too? is your red5 server locate on same lan as your sip proxy/asterisk? maybe a port/ pinhole on your firewall need to be open in order to get the stream and its happened after you transmit the first rtp. what i try to find is if you have some nat transversal between your red5 and your proxy. Lior

  hello Tony when you wrote connect to sip using flex - what do you mean by that? where do you register too? is your red5 server locate on same lan as your sip proxy/asterisk? maybe a port/ pinhole on your firewall need to be open in order to get the stream and its happened after you transmit the first rtp. what i try to find is if you have some nat transversal between your red5 and your proxy. Lior
• Nov 24, 2009

  issue 48 (possible - allow calling without login form? (auto-login)) commented on by olajide.dele   -   The code is still there ;-)

  The code is still there ;-)
• Nov 23, 2009

  issue 48 (possible - allow calling without login form? (auto-login)) commented on by lior.herman777   -   Hi Roslan there is other way to do auto login. i added to the code long time ago external function name login1. login1(obProxy1, myuid, phone1, username1, password1, realm1,server1,url1,conference1); add to your page a test button in the end of the html page <form name="demo"> <button onclick="red5phone.login(obProxy1, myuid, phone1, username1, password1, realm1,server1,url1,conference1);">Login</button> </form> it should work unless someone change it. Lior

  Hi Roslan there is other way to do auto login. i added to the code long time ago external function name login1. login1(obProxy1, myuid, phone1, username1, password1, realm1,server1,url1,conference1); add to your page a test button in the end of the html page <form name="demo"> <button onclick="red5phone.login(obProxy1, myuid, phone1, username1, password1, realm1,server1,url1,conference1);">Login</button> </form> it should work unless someone change it. Lior
• Nov 21, 2009

  issue 85 (Why do we need Red5 server?) reported by Alexi.zuo   -   I am a newbie to SIP phone. Can Flash phone talk to Asterisk directly? It seems red5phone use Red5 as a proxy. Can Flash work just like an ActiveX control and talk to Asterisk since Asterisk support SIP/RTMP protocol.

  I am a newbie to SIP phone. Can Flash phone talk to Asterisk directly? It seems red5phone use Red5 as a proxy. Can Flash work just like an ActiveX control and talk to Asterisk since Asterisk support SIP/RTMP protocol.
• Nov 18, 2009

  issue 48 (possible - allow calling without login form? (auto-login)) commented on by liweispace   -   Hi ruslan, open red5phone.mxml find <local:LoginCanvas id="loginCanvas" x="0" y="0" width="100%" height="100%" borderStyle="none" /> and modify to <local:LoginCanvas id="loginCanvas" x="0" y="0" width="100%" height="100%" borderStyle="none" visible="false"/> then recompile.

  Hi ruslan, open red5phone.mxml find <local:LoginCanvas id="loginCanvas" x="0" y="0" width="100%" height="100%" borderStyle="none" /> and modify to <local:LoginCanvas id="loginCanvas" x="0" y="0" width="100%" height="100%" borderStyle="none" visible="false"/> then recompile.
• Nov 17, 2009

  issue 84 (How can you connect to a SIP phone without having a micropho...) commented on by tony.chappell   -   [Updated 11/17/09] What is the expected output? What do you see instead? What is expected is Red5phone should still connect to the SIP and the user hears audio from phone. What happens is the Red5phone is waiting for audio input from user before publishing any SIP audio.

  [Updated 11/17/09] What is the expected output? What do you see instead? What is expected is Red5phone should still connect to the SIP and the user hears audio from phone. What happens is the Red5phone is waiting for audio input from user before publishing any SIP audio.
• Nov 16, 2009

  issue 84 (How can you connect to a SIP phone without having a micropho...) reported by tony.chappell   -   What steps will reproduce the problem? 1. Connect to SIP using Flex Red5phone client app 2. Either disconnect microphone or click "deny" setting on Flash settings panel 3. What is the expected output? What do you see instead? Red5phone should still connect to SIP and the user hears audio from phone. The Red5phone will connect publish SIP audio until it receives audio from user from microphone. What version of the product are you using? On what operating system? Red5 server 0.8, Red5phone r29 Please provide any additional information below.

  What steps will reproduce the problem? 1. Connect to SIP using Flex Red5phone client app 2. Either disconnect microphone or click "deny" setting on Flash settings panel 3. What is the expected output? What do you see instead? Red5phone should still connect to SIP and the user hears audio from phone. The Red5phone will connect publish SIP audio until it receives audio from user from microphone. What version of the product are you using? On what operating system? Red5 server 0.8, Red5phone r29 Please provide any additional information below.
• Nov 16, 2009

  issue 5 (red5 can not run when useing (sip)) commented on by tony.chappell   -   You need to use a proxy server. Red5phone app has problems running on the same server with Red5. Try using a proxy or having either the Red5phone or Red5 on a different server.

  You need to use a proxy server. Red5phone app has problems running on the same server with Red5. Try using a proxy or having either the Red5phone or Red5 on a different server.

Earlier this year

• Nov 16, 2009

  issue 83 (File handles leak) reported by stoyanov   -   What steps will reproduce the problem? 1. To observe the problem monitor the number of open files by java process: lsof -p PROCPID | wc -l, where PROCPID is the pid of the java process running red5. Do this before and after calling. If server is used intensively it starts throwing exceptions with message "Too many open files" and practically stops functioning. What is the expected output? What do you see instead? It is expected after each call the number of open files (file handles) to be restored to the number before the call. The number of available file handles is spent quickly and the red5/red5phone becomes unusable. The process starts throwing exceptions with message "Too many open files". What version of the product are you using? On what operating system? Operating system is Debian / GNU Linux 2.6. Java 1.5 and Java 6 produce the same result. Using Red5 0.8 and Red5 0.9RC1 with latest sip.zip builds. Please provide any additional information below. While the number of limited file handles (sockets, files, threads) can increase the time before the system becomes unusable, it does not solve the stability problem. Practically, on every call the java process opens around 15-20 file handles and restores only 2 after call hangup. If a server is more intensively used the system may become unusable within minutes. Setting socket timeout value to a lower number (defaults to 60) also doesn't help (as suggested for projects that use Mina for NIO).

  What steps will reproduce the problem? 1. To observe the problem monitor the number of open files by java process: lsof -p PROCPID | wc -l, where PROCPID is the pid of the java process running red5. Do this before and after calling. If server is used intensively it starts throwing exceptions with message "Too many open files" and practically stops functioning. What is the expected output? What do you see instead? It is expected after each call the number of open files (file handles) to be restored to the number before the call. The number of available file handles is spent quickly and the red5/red5phone becomes unusable. The process starts throwing exceptions with message "Too many open files". What version of the product are you using? On what operating system? Operating system is Debian / GNU Linux 2.6. Java 1.5 and Java 6 produce the same result. Using Red5 0.8 and Red5 0.9RC1 with latest sip.zip builds. Please provide any additional information below. While the number of limited file handles (sockets, files, threads) can increase the time before the system becomes unusable, it does not solve the stability problem. Practically, on every call the java process opens around 15-20 file handles and restores only 2 after call hangup. If a server is more intensively used the system may become unusable within minutes. Setting socket timeout value to a lower number (defaults to 60) also doesn't help (as suggested for projects that use Mina for NIO).
• Nov 13, 2009

  issue 82 (New Features Request) reported by dharmesh.mevawala   -   It will be nice if we can have following features 1)Phone-book integration 2)Only one button should be there for Dial/Hangup. When call is not connected it should say “Dail” with green background and if call is connected it should say “Hangup” with red background. 3)Video support

  It will be nice if we can have following features 1)Phone-book integration 2)Only one button should be there for Dial/Hangup. When call is not connected it should say “Dail” with green background and if call is connected it should say “Hangup” with red background. 3)Video support
• Nov 04, 2009

  issue 81 (Transfer error solution) reported by mollamusaoglu   -   What steps will reproduce the problem? 0. For example at least B is red5phone user ; 1. Call from A to B, accept in B 2. Transfer from B to C, accept in C 3. Transfer from A or C to B, accept in B Error: Call is not established, only "incoming call ***" message appears What is the expected output? What do you see instead? Call should be established What version of the product are you using? On what operating system? red5 0.8.0, red5phone r39, Debian Lenny Please provide any additional information below. Solution: Hangup correctly after transfer; Two Solutions; 1--) In "Red5Manager.as" add "netConnection.call("hangup", null, uid)" inside doTransfer() function -> public function doTransfer(transferTo:String):void { netConnection.call("transfer", null, uid, transferTo); netConnection.call("hangup", null, uid); // add this line } 2--) Or in "SipUserAgent.java" add "hangup()" inside transfer() function -> public void transfer( String transfer_to ){ printLog("REFER/TRANSFER", "Init..." ); try { if (call!=null && call.isOnCall()) { call.transfer(transfer_to); hangup(); // add this line } } catch (Exception e) { printLog("transfer: ", e.toString());} } Hope this helps.

  What steps will reproduce the problem? 0. For example at least B is red5phone user ; 1. Call from A to B, accept in B 2. Transfer from B to C, accept in C 3. Transfer from A or C to B, accept in B Error: Call is not established, only "incoming call ***" message appears What is the expected output? What do you see instead? Call should be established What version of the product are you using? On what operating system? red5 0.8.0, red5phone r39, Debian Lenny Please provide any additional information below. Solution: Hangup correctly after transfer; Two Solutions; 1--) In "Red5Manager.as" add "netConnection.call("hangup", null, uid)" inside doTransfer() function -> public function doTransfer(transferTo:String):void { netConnection.call("transfer", null, uid, transferTo); netConnection.call("hangup", null, uid); // add this line } 2--) Or in "SipUserAgent.java" add "hangup()" inside transfer() function -> public void transfer( String transfer_to ){ printLog("REFER/TRANSFER", "Init..." ); try { if (call!=null && call.isOnCall()) { call.transfer(transfer_to); hangup(); // add this line } } catch (Exception e) { printLog("transfer: ", e.toString());} } Hope this helps.
• Nov 02, 2009

  issue 80 (DTMF codes still aren't working) reported by costint   -   What steps will reproduce the problem? 1. Connect to a call, like a conferencing provider 2. Push the keypad buttons to send DTMF codes What is the expected output? What do you see instead? I expect the conferencing provider to receive the dtmf codes and process them. However, no dtmf codes are received by the provider. What version of the product are you using? On what operating system? Red5Phone Rev 39. Please provide any additional information below.

  What steps will reproduce the problem? 1. Connect to a call, like a conferencing provider 2. Push the keypad buttons to send DTMF codes What is the expected output? What do you see instead? I expect the conferencing provider to receive the dtmf codes and process them. However, no dtmf codes are received by the provider. What version of the product are you using? On what operating system? Red5Phone Rev 39. Please provide any additional information below.
• Nov 02, 2009

  issue 78 (Transfer error in Flex example) commented on by mollamusaoglu   -   Ok I got r39 revision from svn. Now it works. Thanks...

  Ok I got r39 revision from svn. Now it works. Thanks...
• Nov 01, 2009

  issue 79 (Red5phone Call Hang) commented on by luisrolo   -   i'm using asteriskwin32 0.66b too.

  i'm using asteriskwin32 0.66b too.
• Oct 30, 2009

  issue 79 (Red5phone Call Hang) reported by luisrolo   -   What steps will reproduce the problem? 1.I can successfully register 2.I can make a call to X-LITE 3.When I accept the call it hangs, so i can't answer the call. What is the expected output? What do you see instead? answer the call without hanging up. What version of the product are you using? On what operating system? windows xp SP3 Red5 0.8.0 final openfire 3.6.4 asterisk-im 1.4.0 red5 plugin 0.1.11 Please provide any additional information below.

  What steps will reproduce the problem? 1.I can successfully register 2.I can make a call to X-LITE 3.When I accept the call it hangs, so i can't answer the call. What is the expected output? What do you see instead? answer the call without hanging up. What version of the product are you using? On what operating system? windows xp SP3 Red5 0.8.0 final openfire 3.6.4 asterisk-im 1.4.0 red5 plugin 0.1.11 Please provide any additional information below.
• Oct 30, 2009

  issue 78 (Transfer error in Flex example) reported by mollamusaoglu   -   What steps will reproduce the problem? 1. Pushing transfer button (destination for example '997') What is the expected output? What do you see instead? Transfer process should done when pressing transfer button What version of the product are you using? On what operating system? red5 server 0.8.0 & red5phone r29 Please provide any additional information below. Error output: [ERROR] [NioProcessor-1] org.red5.server.service.ServiceInvoker - Method transfer with parameters [A6C898C8-D0A8-99A8-D8CE-A63387AFB79A, 997] not found in org.red5.server.webapp.sip.Application@789d63

  What steps will reproduce the problem? 1. Pushing transfer button (destination for example '997') What is the expected output? What do you see instead? Transfer process should done when pressing transfer button What version of the product are you using? On what operating system? red5 server 0.8.0 & red5phone r29 Please provide any additional information below. Error output: [ERROR] [NioProcessor-1] org.red5.server.service.ServiceInvoker - Method transfer with parameters [A6C898C8-D0A8-99A8-D8CE-A63387AFB79A, 997] not found in org.red5.server.webapp.sip.Application@789d63
• Oct 30, 2009

  issue 32 (Problems with g729 audio codec) commented on by seyean   -   Can you tell me, how can to make red5phone support with codec g729?

  Can you tell me, how can to make red5phone support with codec g729?
• Oct 29, 2009

  issue 76 (I want to know that How RTMPUser receieve the RTMP stream pu...) commented on by olajide.dele   -   look at red-screenshare to see how to handle video in RTMPUser class

  look at red-screenshare to see how to handle video in RTMPUser class
• Oct 29, 2009

  issue 77 (Service Invocation error in Red5 on Hangup remote end) commented on by niraj874u   -   Hey I have the same problem. When One user hangup , This error comes. I don't have the solution yet. If you find solution please mail me to : niraj874u[at]gmail[dot]com Thanks, Niraj

  Hey I have the same problem. When One user hangup , This error comes. I don't have the solution yet. If you find solution please mail me to : niraj874u[at]gmail[dot]com Thanks, Niraj
• Oct 28, 2009

  issue 23 (Latest fixes for Red5Phone on openfire project) commented on by brook.patten   -   Looking at the trunk it doesn't look as if these changes have made it in yet. I'm having some of these issues. Would you mind forwarding me the changes or pointing me to the right branch?

  Looking at the trunk it doesn't look as if these changes have made it in yet. I'm having some of these issues. Would you mind forwarding me the changes or pointing me to the right branch?
• Oct 26, 2009

  issue 77 (Service Invocation error in Red5 on Hangup remote end) reported by ad...@wvg-tele.com   -   What steps will reproduce the problem? 1. When remote user hang-up remote sip user 2. does not happen when u hang up on red5phone What is the expected output? What do you see instead? [ERROR] [NioProcessor-1] org.red5.server.service.ServiceInvoker - Error executing call: Service: null Method: publish Num Params: 1 0: null [ERROR] [NioProcessor-1] org.red5.server.service.ServiceInvoker - Service invocation error java.lang.reflect.InvocationTargetException: null at sun.reflect.NativeMethodAccessorImpl.invoke0(Native Method) [na:1.6.0_16] at sun.reflect.NativeMethodAccessorImpl.invoke(NativeMethodAccessorImpl.java:39) [na:1.6.0_16] at sun.reflect.DelegatingMethodAccessorImpl.invoke(DelegatingMethodAccessorImpl.java:25) [na:1.6.0_16] at java.lang.reflect.Method.invoke(Method.java:597) [na:1.6.0_16] at org.red5.server.service.ServiceInvoker.invoke(ServiceInvoker.java:200) [red5.jar:na] at org.red5.server.net.rtmp.RTMPHandler.invokeCall(RTMPHandler.java:186) [red5.jar:na] at org.red5.server.net.rtmp.RTMPHandler.onInvoke(RTMPHandler.java:380) [red5.jar:na] at org.red5.server.net.rtmp.BaseRTMPHandler.messageReceived(BaseRTMPHandler.java:146) [red5.jar:na] at org.red5.server.net.rtmp.RTMPMinaIoHandler.messageReceived(RTMPMinaIoHandler.java:193) [red5.jar:na] at org.apache.mina.core.filterchain.DefaultIoFilterChain$TailFilter.messageReceived(DefaultIoFilterChain.java:713) [mina-core-2.0.0-M7-SNAPSHOT.jar:na] at org.apache.mina.core.filterchain.DefaultIoFilterChain.callNextMessageReceived(DefaultIoFilterChain.java:434) [mina-core-2.0.0-M7-SNAPSHOT.jar:na] at org.apache.mina.core.filterchain.DefaultIoFilterChain.access$1200(DefaultIoFilterChain.java:46) [mina-core-2.0.0-M7-SNAPSHOT.jar:na] at org.apache.mina.core.filterchain.DefaultIoFilterChain$EntryImpl$1.messageReceived(DefaultIoFilterChain.java:793) [mina-core-2.0.0-M7-SNAPSHOT.jar:na] at org.apache.mina.filter.codec.ProtocolCodecFilter$ProtocolDecoderOutputImpl.flush(ProtocolCodecFilter.java:375) [mina-core-2.0.0-M7-SNAPSHOT.jar:na] at org.apache.mina.filter.codec.ProtocolCodecFilter.messageReceived(ProtocolCodecFilter.java:229) [mina-core-2.0.0-M7-SNAPSHOT.jar:na] at org.apache.mina.core.filterchain.DefaultIoFilterChain.callNextMessageReceived(DefaultIoFilterChain.java:434) [mina-core-2.0.0-M7-SNAPSHOT.jar:na] at org.apache.mina.core.filterchain.DefaultIoFilterChain.access$1200(DefaultIoFilterChain.java:46) [mina-core-2.0.0-M7-SNAPSHOT.jar:na] at org.apache.mina.core.filterchain.DefaultIoFilterChain$EntryImpl$1.messageReceived(DefaultIoFilterChain.java:793) [mina-core-2.0.0-M7-SNAPSHOT.jar:na] at org.apache.mina.core.filterchain.IoFilterAdapter.messageReceived(IoFilterAdapter.java:119) [mina-core-2.0.0-M7-SNAPSHOT.jar:na] at org.apache.mina.core.filterchain.DefaultIoFilterChain.callNextMessageReceived(DefaultIoFilterChain.java:434) [mina-core-2.0.0-M7-SNAPSHOT.jar:na] at org.apache.mina.core.filterchain.DefaultIoFilterChain.fireMessageReceived(DefaultIoFilterChain.java:426) [mina-core-2.0.0-M7-SNAPSHOT.jar:na] at org.apache.mina.core.polling.AbstractPollingIoProcessor.read(AbstractPollingIoProcessor.java:632) [mina-core-2.0.0-M7-SNAPSHOT.jar:na] at org.apache.mina.core.polling.AbstractPollingIoProcessor.process(AbstractPollingIoProcessor.java:592) [mina-core-2.0.0-M7-SNAPSHOT.jar:na] at org.apache.mina.core.polling.AbstractPollingIoProcessor.process(AbstractPollingIoProcessor.java:581) [mina-core-2.0.0-M7-SNAPSHOT.jar:na] at org.apache.mina.core.polling.AbstractPollingIoProcessor.access$400(AbstractPollingIoProcessor.java:59) [mina-core-2.0.0-M7-SNAPSHOT.jar:na] at org.apache.mina.core.polling.AbstractPollingIoProcessor$Processor.run(AbstractPollingIoProcessor.java:945) [mina-core-2.0.0-M7-SNAPSHOT.jar:na] at org.apache.mina.util.NamePreservingRunnable.run(NamePreservingRunnable.java:64) [mina-core-2.0.0-M7-SNAPSHOT.jar:na] at java.util.concurrent.ThreadPoolExecutor$Worker.runTask(ThreadPoolExecutor.java:886) [na:1.6.0_16] at java.util.concurrent.ThreadPoolExecutor$Worker.run(ThreadPoolExecutor.java:908) [na:1.6.0_16] at java.lang.Thread.run(Thread.java:619) [na:1.6.0_16] Caused by: java.lang.NullPointerException: null at org.red5.server.stream.StreamService.publish(StreamService.java:343) [red5.jar:na] ... 30 common frames omitted What version of the product are you using? On what operating system? Win2003 , Red 8 Please provide any additional information below. Can somebody explain me why this happens ? Is this problem ?

  What steps will reproduce the problem? 1. When remote user hang-up remote sip user 2. does not happen when u hang up on red5phone What is the expected output? What do you see instead? [ERROR] [NioProcessor-1] org.red5.server.service.ServiceInvoker - Error executing call: Service: null Method: publish Num Params: 1 0: null [ERROR] [NioProcessor-1] org.red5.server.service.ServiceInvoker - Service invocation error java.lang.reflect.InvocationTargetException: null at sun.reflect.NativeMethodAccessorImpl.invoke0(Native Method) [na:1.6.0_16] at sun.reflect.NativeMethodAccessorImpl.invoke(NativeMethodAccessorImpl.java:39) [na:1.6.0_16] at sun.reflect.DelegatingMethodAccessorImpl.invoke(DelegatingMethodAccessorImpl.java:25) [na:1.6.0_16] at java.lang.reflect.Method.invoke(Method.java:597) [na:1.6.0_16] at org.red5.server.service.ServiceInvoker.invoke(ServiceInvoker.java:200) [red5.jar:na] at org.red5.server.net.rtmp.RTMPHandler.invokeCall(RTMPHandler.java:186) [red5.jar:na] at org.red5.server.net.rtmp.RTMPHandler.onInvoke(RTMPHandler.java:380) [red5.jar:na] at org.red5.server.net.rtmp.BaseRTMPHandler.messageReceived(BaseRTMPHandler.java:146) [red5.jar:na] at org.red5.server.net.rtmp.RTMPMinaIoHandler.messageReceived(RTMPMinaIoHandler.java:193) [red5.jar:na] at org.apache.mina.core.filterchain.DefaultIoFilterChain$TailFilter.messageReceived(DefaultIoFilterChain.java:713) [mina-core-2.0.0-M7-SNAPSHOT.jar:na] at org.apache.mina.core.filterchain.DefaultIoFilterChain.callNextMessageReceived(DefaultIoFilterChain.java:434) [mina-core-2.0.0-M7-SNAPSHOT.jar:na] at org.apache.mina.core.filterchain.DefaultIoFilterChain.access$1200(DefaultIoFilterChain.java:46) [mina-core-2.0.0-M7-SNAPSHOT.jar:na] at org.apache.mina.core.filterchain.DefaultIoFilterChain$EntryImpl$1.messageReceived(DefaultIoFilterChain.java:793) [mina-core-2.0.0-M7-SNAPSHOT.jar:na] at org.apache.mina.filter.codec.ProtocolCodecFilter$ProtocolDecoderOutputImpl.flush(ProtocolCodecFilter.java:375) [mina-core-2.0.0-M7-SNAPSHOT.jar:na] at org.apache.mina.filter.codec.ProtocolCodecFilter.messageReceived(ProtocolCodecFilter.java:229) [mina-core-2.0.0-M7-SNAPSHOT.jar:na] at org.apache.mina.core.filterchain.DefaultIoFilterChain.callNextMessageReceived(DefaultIoFilterChain.java:434) [mina-core-2.0.0-M7-SNAPSHOT.jar:na] at org.apache.mina.core.filterchain.DefaultIoFilterChain.access$1200(DefaultIoFilterChain.java:46) [mina-core-2.0.0-M7-SNAPSHOT.jar:na] at org.apache.mina.core.filterchain.DefaultIoFilterChain$EntryImpl$1.messageReceived(DefaultIoFilterChain.java:793) [mina-core-2.0.0-M7-SNAPSHOT.jar:na] at org.apache.mina.core.filterchain.IoFilterAdapter.messageReceived(IoFilterAdapter.java:119) [mina-core-2.0.0-M7-SNAPSHOT.jar:na] at org.apache.mina.core.filterchain.DefaultIoFilterChain.callNextMessageReceived(DefaultIoFilterChain.java:434) [mina-core-2.0.0-M7-SNAPSHOT.jar:na] at org.apache.mina.core.filterchain.DefaultIoFilterChain.fireMessageReceived(DefaultIoFilterChain.java:426) [mina-core-2.0.0-M7-SNAPSHOT.jar:na] at org.apache.mina.core.polling.AbstractPollingIoProcessor.read(AbstractPollingIoProcessor.java:632) [mina-core-2.0.0-M7-SNAPSHOT.jar:na] at org.apache.mina.core.polling.AbstractPollingIoProcessor.process(AbstractPollingIoProcessor.java:592) [mina-core-2.0.0-M7-SNAPSHOT.jar:na] at org.apache.mina.core.polling.AbstractPollingIoProcessor.process(AbstractPollingIoProcessor.java:581) [mina-core-2.0.0-M7-SNAPSHOT.jar:na] at org.apache.mina.core.polling.AbstractPollingIoProcessor.access$400(AbstractPollingIoProcessor.java:59) [mina-core-2.0.0-M7-SNAPSHOT.jar:na] at org.apache.mina.core.polling.AbstractPollingIoProcessor$Processor.run(AbstractPollingIoProcessor.java:945) [mina-core-2.0.0-M7-SNAPSHOT.jar:na] at org.apache.mina.util.NamePreservingRunnable.run(NamePreservingRunnable.java:64) [mina-core-2.0.0-M7-SNAPSHOT.jar:na] at java.util.concurrent.ThreadPoolExecutor$Worker.runTask(ThreadPoolExecutor.java:886) [na:1.6.0_16] at java.util.concurrent.ThreadPoolExecutor$Worker.run(ThreadPoolExecutor.java:908) [na:1.6.0_16] at java.lang.Thread.run(Thread.java:619) [na:1.6.0_16] Caused by: java.lang.NullPointerException: null at org.red5.server.stream.StreamService.publish(StreamService.java:343) [red5.jar:na] ... 30 common frames omitted What version of the product are you using? On what operating system? Win2003 , Red 8 Please provide any additional information below. Can somebody explain me why this happens ? Is this problem ?
• Oct 26, 2009

  issue 62 (RTP end to end point) commented on by alex.madmind   -   The scheme is actually like this: Webphone <----> Red5 Flash Server <----> Re5phone SIP UA <----> Called party So as you see the call is not controlled via SIP in the webphone. So no direct connection is possible.

  The scheme is actually like this: Webphone <----> Red5 Flash Server <----> Re5phone SIP UA <----> Called party So as you see the call is not controlled via SIP in the webphone. So no direct connection is possible.
• Oct 23, 2009

  issue 76 (I want to know that How RTMPUser receieve the RTMP stream pu...) reported by vincent19850927   -   As title, because I am now planning to add the video part in the red5phone. Is the class PlayNetStream inside the RTMPUser can receive the RTMP VideoData already? In class PlayNetStream, inside the method dispatchEvent(IEvent event): if ( rtmpEvent instanceof VideoData ) { IoBuffer videoData = ( (IStreamData) rtmpEvent ).getData().asReadOnlyBuffer(); Is the videoData can receive the RTMP videoData already? Thanks, Vincent

  As title, because I am now planning to add the video part in the red5phone. Is the class PlayNetStream inside the RTMPUser can receive the RTMP VideoData already? In class PlayNetStream, inside the method dispatchEvent(IEvent event): if ( rtmpEvent instanceof VideoData ) { IoBuffer videoData = ( (IStreamData) rtmpEvent ).getData().asReadOnlyBuffer(); Is the videoData can receive the RTMP videoData already? Thanks, Vincent
• Oct 21, 2009

  issue 75 (why RegisterAgent pick the ip of localhost 27.0.0.2:5070) commented on by lisakuepper   -   sorry there is one typing mistake: If I put red5 Server and Asterisk togehter in the same machine. It would NOT be a problem.

  sorry there is one typing mistake: If I put red5 Server and Asterisk togehter in the same machine. It would NOT be a problem.
• Oct 21, 2009

  issue 75 (why RegisterAgent pick the ip of localhost 27.0.0.2:5070) reported by lisakuepper   -   I have started the red5 Server with red5phone, have given my user data and have logged in. It schown the "connection success" but later got a Reg. Failure time out. I saw in the log in shell with the message "RegisterAgent: Registering contact <sip:xxxx@127.0.0.2:5070>" If I put red5 Server and Asterisk togehter in the same machine. It would be a problem. Why Red5 Server don't pick the ip 192.168.x.x I use openSUSE 11.1, standalone Red5 Server and Red5phone sip_r29.zip

  I have started the red5 Server with red5phone, have given my user data and have logged in. It schown the "connection success" but later got a Reg. Failure time out. I saw in the log in shell with the message "RegisterAgent: Registering contact <sip:xxxx@127.0.0.2:5070>" If I put red5 Server and Asterisk togehter in the same machine. It would be a problem. Why Red5 Server don't pick the ip 192.168.x.x I use openSUSE 11.1, standalone Red5 Server and Red5phone sip_r29.zip
• Oct 21, 2009

  issue 73 (8 new established socket pairs that never closed after each ...) commented on by eleo...@yahoo.com   -   I noticed this problem too...it keeps growing and will only close if you restart red5. I'm using windows, does linux have this issue too?

  I noticed this problem too...it keeps growing and will only close if you restart red5. I'm using windows, does linux have this issue too?
• Oct 19, 2009

  issue 74 (RTMP not working) reported by mera2solutions   -   Hello, I install red5 server successfully. What is the expected output? What do you see instead? (1441) Connections: true | true (4280) connected?: true (7396) NetConnection.onStatus: level = error code = NetConnection.Connect.Failed [WARN] [Red5_Scheduler_Worker-1] org.red5.server.net.rtmp.RTMPConnection - Closing RTMPMinaConnection from 80.96.194.52 : 49601 to 173.45.114.196 (in: 3425 out 3209 ), with id 5104638 due to long handshake What version of the product are you using? On what operating system? red 8.0.0 Debian 4.0 etch java 6.0 Please provide any additional information below. everything looks running ok, but the phone gives me this error: Failed to connect to Red5 Thanks, O

  Hello, I install red5 server successfully. What is the expected output? What do you see instead? (1441) Connections: true | true (4280) connected?: true (7396) NetConnection.onStatus: level = error code = NetConnection.Connect.Failed [WARN] [Red5_Scheduler_Worker-1] org.red5.server.net.rtmp.RTMPConnection - Closing RTMPMinaConnection from 80.96.194.52 : 49601 to 173.45.114.196 (in: 3425 out 3209 ), with id 5104638 due to long handshake What version of the product are you using? On what operating system? red 8.0.0 Debian 4.0 etch java 6.0 Please provide any additional information below. everything looks running ok, but the phone gives me this error: Failed to connect to Red5 Thanks, O
• Oct 19, 2009

  issue 73 (8 new established socket pairs that never closed after each ...) reported by chengweichao   -   We had found a strange problem when we tested the red5phone. After starting the red5phone server, we use "netstat" to monitor network port usage. (ps. we had use "netstat -b" to confirm these sockets are established by java process) When we make phone calls from the web page, the established socket pairs continue growing. It seems to establish 8 new socket pairs that never closed after each phone call. after 1 call TCP 127.0.0.1:54336 james:54337 ESTABLISHED TCP 127.0.0.1:54337 james:54336 ESTABLISHED TCP 127.0.0.1:54338 james:54339 ESTABLISHED TCP 127.0.0.1:54339 james:54338 ESTABLISHED TCP 127.0.0.1:54340 james:54341 ESTABLISHED TCP 127.0.0.1:54341 james:54340 ESTABLISHED TCP 127.0.0.1:54342 james:54343 ESTABLISHED TCP 127.0.0.1:54343 james:54342 ESTABLISHED after the 2nd call TCP 127.0.0.1:54336 james:54337 ESTABLISHED TCP 127.0.0.1:54337 james:54336 ESTABLISHED TCP 127.0.0.1:54338 james:54339 ESTABLISHED TCP 127.0.0.1:54339 james:54338 ESTABLISHED TCP 127.0.0.1:54340 james:54341 ESTABLISHED TCP 127.0.0.1:54341 james:54340 ESTABLISHED TCP 127.0.0.1:54342 james:54343 ESTABLISHED TCP 127.0.0.1:54343 james:54342 ESTABLISHED TCP 127.0.0.1:54354 james:54355 ESTABLISHED TCP 127.0.0.1:54355 james:54354 ESTABLISHED TCP 127.0.0.1:54356 james:54357 ESTABLISHED TCP 127.0.0.1:54357 james:54356 ESTABLISHED TCP 127.0.0.1:54358 james:54359 ESTABLISHED TCP 127.0.0.1:54359 james:54358 ESTABLISHED TCP 127.0.0.1:54360 james:54361 ESTABLISHED TCP 127.0.0.1:54361 james:54360 ESTABLISHED Does anyone notice this problem and know the reason?

  We had found a strange problem when we tested the red5phone. After starting the red5phone server, we use "netstat" to monitor network port usage. (ps. we had use "netstat -b" to confirm these sockets are established by java process) When we make phone calls from the web page, the established socket pairs continue growing. It seems to establish 8 new socket pairs that never closed after each phone call. after 1 call TCP 127.0.0.1:54336 james:54337 ESTABLISHED TCP 127.0.0.1:54337 james:54336 ESTABLISHED TCP 127.0.0.1:54338 james:54339 ESTABLISHED TCP 127.0.0.1:54339 james:54338 ESTABLISHED TCP 127.0.0.1:54340 james:54341 ESTABLISHED TCP 127.0.0.1:54341 james:54340 ESTABLISHED TCP 127.0.0.1:54342 james:54343 ESTABLISHED TCP 127.0.0.1:54343 james:54342 ESTABLISHED after the 2nd call TCP 127.0.0.1:54336 james:54337 ESTABLISHED TCP 127.0.0.1:54337 james:54336 ESTABLISHED TCP 127.0.0.1:54338 james:54339 ESTABLISHED TCP 127.0.0.1:54339 james:54338 ESTABLISHED TCP 127.0.0.1:54340 james:54341 ESTABLISHED TCP 127.0.0.1:54341 james:54340 ESTABLISHED TCP 127.0.0.1:54342 james:54343 ESTABLISHED TCP 127.0.0.1:54343 james:54342 ESTABLISHED TCP 127.0.0.1:54354 james:54355 ESTABLISHED TCP 127.0.0.1:54355 james:54354 ESTABLISHED TCP 127.0.0.1:54356 james:54357 ESTABLISHED TCP 127.0.0.1:54357 james:54356 ESTABLISHED TCP 127.0.0.1:54358 james:54359 ESTABLISHED TCP 127.0.0.1:54359 james:54358 ESTABLISHED TCP 127.0.0.1:54360 james:54361 ESTABLISHED TCP 127.0.0.1:54361 james:54360 ESTABLISHED Does anyone notice this problem and know the reason?
• Oct 14, 2009

  r39 (Adding RTP and SIP ports configuration and management Adding...) committed by rafaelfc12   -   Adding RTP and SIP ports configuration and management Adding Volume Normalization configuration

  Adding RTP and SIP ports configuration and management Adding Volume Normalization configuration
• Oct 13, 2009

  r38 (Updating .class files;) committed by rafaelfc12   -   Updating .class files;

  Updating .class files;
• Oct 13, 2009

  r37 (Fix call hangup; Why we need more than one Call class instan...) committed by rafaelfc12   -   Fix call hangup; Why we need more than one Call class instance?

  Fix call hangup; Why we need more than one Call class instance?
• Oct 13, 2009

  r36 (Fix call disconnect) committed by rafaelfc12   -   Fix call disconnect

  Fix call disconnect
• Oct 13, 2009

  r35 (DTMF fix) committed by rafaelfc12   -   DTMF fix

  DTMF fix
• Oct 13, 2009

  r34 (Enforces security passing A1 MD5 SIP Authentication Digest P...) committed by rafaelfc12   -   Enforces security passing A1 MD5 SIP Authentication Digest Parameter instead of plain text password

  Enforces security passing A1 MD5 SIP Authentication Digest Parameter instead of plain text password
• Oct 13, 2009

  r33 (Adding Config class, firstly to store audio codecs precedenc...) committed by rafaelfc12   -   Adding Config class, firstly to store audio codecs precedence

  Adding Config class, firstly to store audio codecs precedence
• Oct 13, 2009

  r32 (Downgrade to java 1.5;) committed by rafaelfc12   -   Downgrade to java 1.5;

  Downgrade to java 1.5;
• Oct 13, 2009

  r31 (Creating a branch for my improvements) committed by rafaelfc12   -   Creating a branch for my improvements

  Creating a branch for my improvements
• Oct 13, 2009

  r30 (Creating a branch for my improvements) committed by rafaelfc12   -   Creating a branch for my improvements

  Creating a branch for my improvements
• Oct 10, 2009

  issue 72 (calling to the same user that called you not working - solut...) commented on by olajide.dele   -   Marcin, you are a star (#). Thanks for fixing that.

  Marcin, you are a star (#). Thanks for fixing that.
• Oct 10, 2009

  sip_r29.zip (Includes Fix for issue 71 Reported by Marcin.Balcer) file uploaded by olajide.dele
• Oct 10, 2009

  r29 (Fix Reported by Marcin.Balcer) committed by olajide.dele   -   Fix Reported by Marcin.Balcer

  Fix Reported by Marcin.Balcer
• Oct 09, 2009

  issue 72 (calling to the same user that called you not working - solut...) reported by Marcin.Balcer   -   What steps will reproduce the problem? 1. login and register 2 different users. Lets call the A and B 2. call from A to B 3. accept in B 4) hangup in B 5) try to call from B to A Error -> no incoming call. What is the expected output? What do you see instead? Incoming call should appear. What version of the product are you using? On what operating system? red5 0.8.0 Please provide any additional information below. The solution: in SIPUserAgent.java class you need to do following changes: 1) add call cancel as below: public void call( String target_url ) { printLog( "call", "Init..." ); changeStatus( UA_OUTGOING_CALL ); if (call != null) { printLog( "call", "cancelling old object:" + this.call); this.call.cancel(); } call = new ExtendedCall( sipProvider, userProfile.fromUrl, userProfile.contactUrl, userProfile.username, userProfile.realm, userProfile.passwd, this ); 2) add call cancell as below: public void listen() { printLog( "listen", "Init..." ); changeStatus( UA_IDLE ); if (call != null) { printLog( "listen", "cancelling old object:" + this.call); this.call.cancel(); } call = new ExtendedCall( sipProvider, userProfile.fromUrl, userProfile.contactUrl, userProfile.username, userProfile.realm, userProfile.passwd, this );

  What steps will reproduce the problem? 1. login and register 2 different users. Lets call the A and B 2. call from A to B 3. accept in B 4) hangup in B 5) try to call from B to A Error -> no incoming call. What is the expected output? What do you see instead? Incoming call should appear. What version of the product are you using? On what operating system? red5 0.8.0 Please provide any additional information below. The solution: in SIPUserAgent.java class you need to do following changes: 1) add call cancel as below: public void call( String target_url ) { printLog( "call", "Init..." ); changeStatus( UA_OUTGOING_CALL ); if (call != null) { printLog( "call", "cancelling old object:" + this.call); this.call.cancel(); } call = new ExtendedCall( sipProvider, userProfile.fromUrl, userProfile.contactUrl, userProfile.username, userProfile.realm, userProfile.passwd, this ); 2) add call cancell as below: public void listen() { printLog( "listen", "Init..." ); changeStatus( UA_IDLE ); if (call != null) { printLog( "listen", "cancelling old object:" + this.call); this.call.cancel(); } call = new ExtendedCall( sipProvider, userProfile.fromUrl, userProfile.contactUrl, userProfile.username, userProfile.realm, userProfile.passwd, this );
• Oct 09, 2009

  issue 71 (openlaszlo red5phone not working - I know the sollution) commented on by Marcin.Balcer   -   Sorry for typo "sollution" -> solution.

  Sorry for typo "sollution" -> solution.
• Oct 09, 2009

  issue 71 (openlaszlo red5phone not working - I know the sollution) reported by Marcin.Balcer   -   What steps will reproduce the problem? 1. open index.html 2. click on "Openlaszlo Phone Template" 3. connect, register 4. try to make a call from other instance to registered user. Error -> no incoming call What is the expected output? What do you see instead? incoming call should be seen in logs. Also after registration "registrationSucess" should be visible in the openlaszlo logs. What version of the product are you using? On what operating system? red5 0.8.0 Please provide any additional information below. Here is the code that works fine (you need to move some part of code from init to on_nc handler): <method name="init"> <![CDATA[ super.init(); this.myMic = Microphone.get(); this.myMic.setRate(8); ]]> </method> <handler name="on_nc"><![CDATA[ this._nc.registrationSucess = function(msg){ // if (debug) Debug.write("SIP: registrationSucess:", msg); this.t.registrationSucess.sendEvent(msg); } this._nc.registrationFailure = function(msg){ // if (debug) Debug.write("SIP: registrationFailure:", msg); this.t.registrationFailure.sendEvent(msg); } this._nc.incoming = function(iSource, iSourceName, iDestination, iDestinationName){ this.t.source = iSource; this.t.sourceName = iSourceName; this.t.destination = iDestination; this.t.destinationName = iDestinationName; //if (debug) Debug.write("SIP: callIncoming:", source); this.t.callIncoming.sendEvent(source, sourceName, destination, destinationName); } this._nc.connected = function(publishName, playName){ this.t.incomingNetStream = new NetStream(_nc); this.t.outgoingNetStream = new NetStream(_nc); this.t.outgoingNetStream.attachAudio(myMic); this.t.incomingNetStream.play(playName); this.t.outgoingNetStream.publish(publishName, "live"); //if (debug) Debug.write("SIP: callConnected:", publishName + "->" + playName); this.t.callConnected.sendEvent(publishName, playName); } _nc.callState = function(msg){ //if (debug) Debug.write("SIP: callState:", msg); if (msg == "onUaCallClosed") { if (this.t.incomingNetStream != null) { this.t.incomingNetStream.play(false); this.t.outgoingNetStream.attachAudio(null); this.t.outgoingNetStream.publish(false); this.t.incomingNetStream = null; this.t.outgoingNetStream = null; } } this.t.callState.sendEvent(msg); } ]]> </handler>

  What steps will reproduce the problem? 1. open index.html 2. click on "Openlaszlo Phone Template" 3. connect, register 4. try to make a call from other instance to registered user. Error -> no incoming call What is the expected output? What do you see instead? incoming call should be seen in logs. Also after registration "registrationSucess" should be visible in the openlaszlo logs. What version of the product are you using? On what operating system? red5 0.8.0 Please provide any additional information below. Here is the code that works fine (you need to move some part of code from init to on_nc handler): <method name="init"> <![CDATA[ super.init(); this.myMic = Microphone.get(); this.myMic.setRate(8); ]]> </method> <handler name="on_nc"><![CDATA[ this._nc.registrationSucess = function(msg){ // if (debug) Debug.write("SIP: registrationSucess:", msg); this.t.registrationSucess.sendEvent(msg); } this._nc.registrationFailure = function(msg){ // if (debug) Debug.write("SIP: registrationFailure:", msg); this.t.registrationFailure.sendEvent(msg); } this._nc.incoming = function(iSource, iSourceName, iDestination, iDestinationName){ this.t.source = iSource; this.t.sourceName = iSourceName; this.t.destination = iDestination; this.t.destinationName = iDestinationName; //if (debug) Debug.write("SIP: callIncoming:", source); this.t.callIncoming.sendEvent(source, sourceName, destination, destinationName); } this._nc.connected = function(publishName, playName){ this.t.incomingNetStream = new NetStream(_nc); this.t.outgoingNetStream = new NetStream(_nc); this.t.outgoingNetStream.attachAudio(myMic); this.t.incomingNetStream.play(playName); this.t.outgoingNetStream.publish(publishName, "live"); //if (debug) Debug.write("SIP: callConnected:", publishName + "->" + playName); this.t.callConnected.sendEvent(publishName, playName); } _nc.callState = function(msg){ //if (debug) Debug.write("SIP: callState:", msg); if (msg == "onUaCallClosed") { if (this.t.incomingNetStream != null) { this.t.incomingNetStream.play(false); this.t.outgoingNetStream.attachAudio(null); this.t.outgoingNetStream.publish(false); this.t.incomingNetStream = null; this.t.outgoingNetStream = null; } } this.t.callState.sendEvent(msg); } ]]> </handler>
• Oct 08, 2009

  sip_r28.zip (Red5phone ver R28) file uploaded by olajide.dele   -  

  Labels: red5phone red5 sip

  Labels: red5phone red5 sip
• Oct 08, 2009

  r28 (Timestamp fix by Rafael on RTPSender Transfer feature by Li...) committed by olajide.dele   -   Timestamp fix by Rafael on RTPSender Transfer feature by Lior Conference feature using transfer by dele

  Timestamp fix by Rafael on RTPSender Transfer feature by Lior Conference feature using transfer by dele