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From: apurv t. <apu...@gm...> - 2012-06-20 15:57:27
|
Hi everyone, I am building an interface for an application where I think cairo project can help. The application needs to be such that I can call into using a SIP phone. Now, the application needs to take in audio and send back audio to the caller. Currently, the application can send audio over TCP socket and similarly receive audio too. I want to using cario SIP server to be able to bridge the connection between the SIP caller and this server. Is it possible to do that using cairo project? I was looking at BargeInClient.java code in this project, but the demo only use TTS for speech synthesis. Could some one please let me know how to send media from a socket connection or from an audio file? Thanks, - Apurv Tiwari |
From: x.liu <x....@hw...> - 2012-05-08 14:21:38
|
Hello, Just joined to this mailing list. I'd like to use Cairo MRCP server with FreeSwitch. I am wondering if it is straightforward or do I need a FreeSwitch plug-in module for Cairo? Has anyone tried to connect Cairo with FreeSwitch? Do you have any example of the configuration files? I know FreeSwitch works with PocketSphinx via its built-in plug-in module. The reason I want to use Cairo is that it wraps the Sphinx4 which I may want to modify/extend later on. Many thanks! Xing -- Heriot-Watt University is the Sunday Times Scottish University of the Year 2011-2012 Heriot-Watt University is a Scottish charity registered under charity number SC000278. |
From: spencer l. <spe...@gm...> - 2010-11-19 02:10:12
|
Hi Chris, I recently switched to windows 7 and it is working ok on my new machine. So I dont think that is it. Though I have had some problems with other apps on windows7. I get he same exception you are getting, but as I said before it does not effect the functionality. That exception always occurs at the end of the mrcp session -- it really is a warning from the mrcp library that cairo should cleanup the session in a more elegant manner. I included what I see in the demo and transmitter windows below. Is this same as you? Can you tell me again what the difference between xp and windows7 output looks like? We may need to turn on logging. It does appear that the audio was synthesized based on this from your earlier email C:\temp\cairo\basePromptDir\12c4f7a65f6@speechsynth\1289823631677.au<12c4f7a65f6@speechsynth%5C1289823631677.au> The problem must be in the RTP streaming from the server to the client. Or with the client taking the stream and playing it on the machines speaker. Spencer This is what I get in my synthclient.bat window 2010-11-18 17:40:56,355 INFO {main} org.speechforge.cairo.demo.tts.SpeechSynthClient Sending a SIP invitation to the cairo server. 2010-11-18 17:40:57,096 WARN {EventScannerThread} org.speechforge.cairo.sip.SipListenerImpl Received an unhandled SIP response status code (ignoring it): 183 : Session progress 2010-11-18 17:40:57,272 INFO {main} org.speechforge.cairo.demo.tts.SpeechSynthClient Received the SIP Response. 2010-11-18 17:41:10,200 INFO {Thread-1} org.speechforge.cairo.sip.SimpleSipAgent Sent a SIP BYE. This is what i get in the transmitter window 2010-11-18 17:39:43,938 INFO {main} org.speechforge.cairo.server.resource.TransmitterResource looking up: rmi://spencers-pc/ResourceRegistry 2010-11-18 17:39:45,426 INFO {main} org.speechforge.cairo.server.resource.TransmitterResource binding transmitter resource... 2010-11-18 17:39:45,491 INFO {main} org.speechforge.cairo.server.resource.TransmitterResource Resource bound and waiting... Wrote synthesized speech to C:\temp\cairo\basePromptDir\12c61c96549@speechsynth\1290130857612.au 2010-11-18 17:41:11,316 WARN {IoThreadPool-1} org.mrcp4j.server.SESSION EXCEPTION: An existing connection was forcibly closed by the remote host java.io.IOException: An existing connection was forcibly closed by the remote host at sun.nio.ch.SocketDispatcher.read0(Native Method) at sun.nio.ch.SocketDispatcher.read(SocketDispatcher.java:25) at sun.nio.ch.IOUtil.readIntoNativeBuffer(IOUtil.java:237) at sun.nio.ch.IOUtil.read(IOUtil.java:204) at sun.nio.ch.SocketChannelImpl.read(SocketChannelImpl.java:236) at org.apache.mina.io.socket.SocketIoProcessor.read(SocketIoProcessor.java:265) at org.apache.mina.io.socket.SocketIoProcessor.processSessions(SocketIoProcessor.java:238) at org.apache.mina.io.socket.SocketIoProcessor.access$200(SocketIoProcessor.java:42) at org.apache.mina.io.socket.SocketIoProcessor$Worker.run(SocketIoProcessor.java:555) On Thu, Nov 18, 2010 at 2:05 AM, Christian Schulz <chs...@df...> wrote: > Hi Spencer, > > so I was trying to start the demo with my home machine. And there I did not > get the reported error. I cannot tell why. > The only difference between the machines I can think of is that at home I > have windows xp running instead of windows 7. > So there is also no speech synthesis generated in the demo running on > windows 7. By the way, just want to clarify I am only referring to different > variants of the operating system to make it easier to talk about the issue. > So it is not the case that i am thinking that that difference causes the > issue. > > Nevertheless further hints are welcome! > > Chris > > Am 15.11.2010 19:08, schrieb spencer lord: > > This exception in the transmitter is ok too. It is low priority known > problem, regarding the way the mrcp channels are shutdown. MRCP4J complains > with this message if things are not closed in certain way -- but it handles > the situation. At the very least you should be able to run the demos. > > Are you sure you do not hear any audio when you run synthClient? I think > it should be working. Is it possible the volume is turned down? > > On Mon, Nov 15, 2010 at 9:53 AM, Christian Schulz <chs...@df...>wrote: > >> For the transmitter: >> >> Resource bound and waiting... >> Wrote synthesized speech to C:\temp\cairo\basePromptDir\ >> 12c4f7a65f6@speechsynth\1289823631677.au<12c4f7a65f6@speechsynth%5C1289823631677.au> >> 2010-11-15 13:20:45,349 WARN {IoThreadPool-1} org.mrcp4j.server.SESSION >> EXCEPTION: Eine vorhandene Verbindung wurde vom Remotehost geschlossen >> >> java.io.IOException: Eine vorhandene Verbindung wurde vom Remotehost >> geschlossen >> >> at sun.nio.ch.SocketDispatcher.read0(Native Method) >> at sun.nio.ch.SocketDispatcher.read(SocketDispatcher.java:25) >> at sun.nio.ch.IOUtil.readIntoNativeBuffer(IOUtil.java:233) >> at sun.nio.ch.IOUtil.read(IOUtil.java:200) >> at sun.nio.ch.SocketChannelImpl.read(SocketChannelImpl.java:236) >> at >> org.apache.mina.io.socket.SocketIoProcessor.read(SocketIoProcessor.java:265) >> at >> org.apache.mina.io.socket.SocketIoProcessor.processSessions(SocketIoProcessor.java:238) >> at >> org.apache.mina.io.socket.SocketIoProcessor.access$200(SocketIoProcessor.java:42) >> at >> org.apache.mina.io.socket.SocketIoProcessor$Worker.run(SocketIoProcessor.java:555) >> >> For the receiver: >> >> Resource bound and waiting... >> 2010-11-15 13:05:27,967 WARN {RMI TCP Connection(2)-134.96.189.199} >> org.speechforge.cairo.server.recog.RTPRecogChannel >> No recengine to return to pool! >> 2010-11-15 13:05:29,390 WARN {IoThreadPool-3} org.mrcp4j.server.SESSION >> EXCEPTION: Eine vorhandene Verbindung wurde vom Remotehost geschlossen >> >> java.io.IOException: Eine vorhandene Verbindung wurde vom Remotehost >> geschlossen >> >> at sun.nio.ch.SocketDispatcher.read0(Native Method) >> at sun.nio.ch.SocketDispatcher.read(SocketDispatcher.java:25) >> at sun.nio.ch.IOUtil.readIntoNativeBuffer(IOUtil.java:233) >> at sun.nio.ch.IOUtil.read(IOUtil.java:200) >> at sun.nio.ch.SocketChannelImpl.read(SocketChannelImpl.java:236) >> at >> org.apache.mina.io.socket.SocketIoProcessor.read(SocketIoProcessor.java:265) >> at >> org.apache.mina.io.socket.SocketIoProcessor.processSessions(SocketIoProcessor.java:238) >> at >> org.apache.mina.io.socket.SocketIoProcessor.access$200(SocketIoProcessor.java:42) >> at >> org.apache.mina.io.socket.SocketIoProcessor$Worker.run(SocketIoProcessor.java:555) >> >> >> Am 15.11.2010 18:51, schrieb spencer lord: >> >> Hi, >> The unhandled SIP 183 message, should not be causing issues with demos. >> It is just an informational message. I do not think that any actions is >> needed in response to this message with MRCP clients. I could be wrong, but >> that is another discussion >> >> What do you see in the transmitter window when you run the synthclient? >> >> Spencer >> >> >> On Mon, Nov 15, 2010 at 8:44 AM, Christian Schulz <chs...@df...>wrote: >> >>> Hi Spencer >>> >>> thanks for your reply, unfortunately your guess did not help. The error >>> message is still the same. I was trying all demos including the speech synth >>> client. There the error message is slightly different maybe it gives you a >>> hint for what reason the demos are not working out. Btw did you check it out >>> if the demos for download are running on your machine? >>> >>> 2010-11-15 17:27:22,158 INFO {main} >>> org.speechforge.cairo.demo.tts.SpeechSynthClient >>> >>> Sending a SIP invitation to the cairo server. >>> 2010-11-15 17:27:22,767 WARN {EventScannerThread} >>> org.speechforge.cairo.sip.SipListenerImpl >>> >>> Received an unhandled SIP response status code (ignoring it): 183 : >>> Session progress >>> 2010-11-15 17:27:22,839 INFO {main} >>> org.speechforge.cairo.demo.tts.SpeechSynthClient >>> Received the SIP Response. >>> 2010-11-15 17:27:36,408 INFO {Thread-1} >>> org.speechforge.cairo.sip.SimpleSipAgent >>> Sent a SIP BYE. >>> C:\Users\chschulz\sources\nlp\cairo\cairo-0.3\demo\bin> >>> >>> Best, >>> >>> Chris >>> >>> >>> Am 15.11.2010 17:06, schrieb spencer lord: >>> >>> Hi Chris, >>> >>> It looks like it may be a problem with jmf. Cairo uses JMF for RTP >>> streaming. Did you install it on the machine you are using? The jmf >>> download link on Cairo installation web page is stale. Sorry, we need to >>> fix that. You can download it here >>> >>> http://www.oracle.com/technetwork/java/javase/download-142937.html >>> >>> If it is already installed, make sue that jmf.jar nd sound.jar are in the >>> lib.ext directory of the jre that you are using. >>> >>> Let me know if that solves the problem. >>> >>> Spencer >>> >>> On Mon, Nov 15, 2010 at 6:40 AM, Christian Schulz <chs...@df...>wrote: >>> >>>> Hi all, >>>> >>>> I am new to the cairo project and I was trying out the demos you can >>>> download here: >>>> http://sourceforge.net/projects/cairo/files/ >>>> >>>> Following the instructions here >>>> http://www.speechforge.org/projects/cairo/intro.html (that is identical >>>> to the instruction in the readme file included in the download) I >>>> however cannot get the demo applications run. In particular the >>>> recognizer will output this error, while trying to connect to the >>>> receiver: >>>> >>>> Sending a SIP invitation to the cairo server. >>>> 2010-11-15 13:05:07,296 WARN {EventScannerThread} >>>> org.speechforge.cairo.sip.SipListenerImpl >>>> Received an unhandled SIP response status code (ignoring it): 183 : >>>> Session progress >>>> 2010-11-15 13:05:07,861 INFO {main} >>>> org.speechforge.cairo.demo.recog.RecognitionClient >>>> Received the SIP Response. >>>> java.lang.NullPointerException >>>> 2010-11-15 13:05:08,397 WARN {Thread-7} >>>> org.speechforge.cairo.rtp.RTPPlayer playSource(): encountered unexpected >>>> exception: >>>> javax.media.NoDataSourceException: Error instantiating class: >>>> com.sun.media.protocol.dsound.DataSource : >>>> java.lang.NullPointerException >>>> at javax.media.Manager.createDataSource(Manager.java:1012) >>>> at >>>> org.speechforge.cairo.rtp.RTPPlayer.playSource(RTPPlayer.java:110) >>>> at >>>> >>>> org.speechforge.cairo.rtp.NativeMediaClient$TransmitThread.run(NativeMediaClient.java:105) >>>> Exception in thread "Thread-7" java.lang.RuntimeException: playSource() >>>> encountered unexpected exception >>>> at >>>> org.speechforge.cairo.rtp.RTPPlayer.playSource(RTPPlayer.java:153) >>>> at >>>> >>>> org.speechforge.cairo.rtp.NativeMediaClient$TransmitThread.run(NativeMediaClient.java:105) >>>> Caused by: javax.media.NoDataSourceException: Error instantiating class: >>>> com.sun.media.protocol.dsound.DataSource : >>>> java.lang.NullPointerException >>>> at javax.media.Manager.createDataSource(Manager.java:1012) >>>> at >>>> org.speechforge.cairo.rtp.RTPPlayer.playSource(RTPPlayer.java:110) >>>> ... 1 more >>>> 2010-11-15 13:05:18,577 WARN {main} >>>> org.speechforge.cairo.demo.recog.RecognitionClient >>>> org.mrcp4j.client.MrcpInvocationException: MRCP response contains an >>>> error code, the request invocation did not complete successfully. >>>> org.mrcp4j.client.MrcpInvocationException: MRCP response contains an >>>> error code, the request invocation did not complete successfully. >>>> at >>>> org.mrcp4j.client.MrcpChannel.sendRequest(MrcpChannel.java:143) >>>> at >>>> >>>> org.speechforge.cairo.demo.recog.RecognitionClient.doRecognize(RecognitionClient.java:164) >>>> at >>>> org.speechforge.cairo.demo.recog.RecognitionClient.main(RecognitionCl >>>> ient.java:358) >>>> 2010-11-15 13:05:27,929 INFO {main} >>>> org.speechforge.cairo.sip.SimpleSipAgent >>>> Sent a SIP BYE. >>>> >>>> Thanks for your help! >>>> >>>> chris >>>> >>>> >>>> ------------------------------------------------------------------------------ >>>> Centralized Desktop Delivery: Dell and VMware Reference Architecture >>>> Simplifying enterprise desktop deployment and management using >>>> Dell EqualLogic storage and VMware View: A highly scalable, end-to-end >>>> client virtualization framework. Read more! >>>> http://p.sf.net/sfu/dell-eql-dev2dev >>>> _______________________________________________ >>>> cairo-user mailing list >>>> cai...@li... >>>> https://lists.sourceforge.net/lists/listinfo/cairo-user >>>> >>> >>> >>> ------------------------------------------------------------------------------ >>> Centralized Desktop Delivery: Dell and VMware Reference Architecture >>> Simplifying enterprise desktop deployment and management using >>> Dell EqualLogic storage and VMware View: A highly scalable, end-to-end >>> client virtualization framework. Read more!http://p.sf.net/sfu/dell-eql-dev2dev >>> >>> >>> _______________________________________________ >>> cairo-user mailing lis...@li...https://lists.sourceforge.net/lists/listinfo/cairo-user >>> >>> >>> -- >>> Christian H. Schulz >>> DFKI GmbH, Campus D3 2 >>> Stuhlsatzenhausweg 3 >>> D-66123 Saarbrücken, Germany >>> +49 (0)681 85775-5371 (tel.) -5021 (fax) >>> mail: chs...@df..., http: www.dfki.de/~chschulz <http://www.dfki.de/%7Echschulz> >>> >>> ------------------------------------------------------------------ >>> Deutsches Forschungszentrum fuer Kuenstliche Intelligenz GmbH >>> Firmensitz: Trippstadter Strasse 122, D-67663 Kaiserslautern >>> Geschaeftsfuehrung: >>> Prof. Dr. Dr. h.c. mult. Wolfgang Wahlster (Vorsitzender) >>> Dr. Walter Olthoff >>> Vorsitzender des Aufsichtsrats: Prof. Dr. h.c. Hans A. Aukes >>> Amtsgericht Kaiserslautern, HRB 2313 >>> >>> >>> >>> ------------------------------------------------------------------------------ >>> Centralized Desktop Delivery: Dell and VMware Reference Architecture >>> Simplifying enterprise desktop deployment and management using >>> Dell EqualLogic storage and VMware View: A highly scalable, end-to-end >>> client virtualization framework. Read more! >>> http://p.sf.net/sfu/dell-eql-dev2dev >>> _______________________________________________ >>> cairo-user mailing list >>> cai...@li... >>> https://lists.sourceforge.net/lists/listinfo/cairo-user >>> >>> >> >> ------------------------------------------------------------------------------ >> Centralized Desktop Delivery: Dell and VMware Reference Architecture >> Simplifying enterprise desktop deployment and management using >> Dell EqualLogic storage and VMware View: A highly scalable, end-to-end >> client virtualization framework. Read more!http://p.sf.net/sfu/dell-eql-dev2dev >> >> >> _______________________________________________ >> cairo-user mailing lis...@li...https://lists.sourceforge.net/lists/listinfo/cairo-user >> >> >> -- >> Christian H. Schulz >> DFKI GmbH, Campus D3 2 >> Stuhlsatzenhausweg 3 >> D-66123 Saarbrücken, Germany >> +49 (0)681 85775-5371 (tel.) -5021 (fax) >> mail: chs...@df..., http: www.dfki.de/~chschulz <http://www.dfki.de/%7Echschulz> >> >> ------------------------------------------------------------------ >> Deutsches Forschungszentrum fuer Kuenstliche Intelligenz GmbH >> Firmensitz: Trippstadter Strasse 122, D-67663 Kaiserslautern >> Geschaeftsfuehrung: >> Prof. Dr. Dr. h.c. mult. Wolfgang Wahlster (Vorsitzender) >> Dr. Walter Olthoff >> Vorsitzender des Aufsichtsrats: Prof. Dr. h.c. Hans A. Aukes >> Amtsgericht Kaiserslautern, HRB 2313 >> >> >> >> ------------------------------------------------------------------------------ >> Centralized Desktop Delivery: Dell and VMware Reference Architecture >> Simplifying enterprise desktop deployment and management using >> Dell EqualLogic storage and VMware View: A highly scalable, end-to-end >> client virtualization framework. Read more! >> http://p.sf.net/sfu/dell-eql-dev2dev >> _______________________________________________ >> cairo-user mailing list >> cai...@li... >> https://lists.sourceforge.net/lists/listinfo/cairo-user >> >> > > ------------------------------------------------------------------------------ > Centralized Desktop Delivery: Dell and VMware Reference Architecture > Simplifying enterprise desktop deployment and management using > Dell EqualLogic storage and VMware View: A highly scalable, end-to-end > client virtualization framework. Read more!http://p.sf.net/sfu/dell-eql-dev2dev > > > _______________________________________________ > cairo-user mailing lis...@li...https://lists.sourceforge.net/lists/listinfo/cairo-user > > > -- > Christian H. Schulz > DFKI GmbH, Campus D3 2 > Stuhlsatzenhausweg 3 > D-66123 Saarbrücken, Germany > +49 (0)681 85775-5371 (tel.) -5021 (fax) > mail: chs...@df..., http: www.dfki.de/~chschulz <http://www.dfki.de/%7Echschulz> > > ------------------------------------------------------------------ > Deutsches Forschungszentrum fuer Kuenstliche Intelligenz GmbH > Firmensitz: Trippstadter Strasse 122, D-67663 Kaiserslautern > Geschaeftsfuehrung: > Prof. Dr. Dr. h.c. mult. Wolfgang Wahlster (Vorsitzender) > Dr. Walter Olthoff > Vorsitzender des Aufsichtsrats: Prof. Dr. h.c. Hans A. Aukes > Amtsgericht Kaiserslautern, HRB 2313 > > |
From: Christian S. <chs...@df...> - 2010-11-18 10:06:30
|
Hi Spencer, so I was trying to start the demo with my home machine. And there I did not get the reported error. I cannot tell why. The only difference between the machines I can think of is that at home I have windows xp running instead of windows 7. So there is also no speech synthesis generated in the demo running on windows 7. By the way, just want to clarify I am only referring to different variants of the operating system to make it easier to talk about the issue. So it is not the case that i am thinking that that difference causes the issue. Nevertheless further hints are welcome! Chris Am 15.11.2010 19:08, schrieb spencer lord: > This exception in the transmitter is ok too. It is low priority > known problem, regarding the way the mrcp channels are shutdown. > MRCP4J complains with this message if things are not closed in certain > way -- but it handles the situation. At the very least you should be > able to run the demos. > > Are you sure you do not hear any audio when you run synthClient? I > think it should be working. Is it possible the volume is turned down? > > On Mon, Nov 15, 2010 at 9:53 AM, Christian Schulz <chs...@df... > <mailto:chs...@df...>> wrote: > > For the transmitter: > > Resource bound and waiting... > Wrote synthesized speech to > C:\temp\cairo\basePromptDir\12c4f7a65f6@speechsynth\1289823631677.au > <mailto:12c4f7a65f6@speechsynth%5C1289823631677.au> > 2010-11-15 13:20:45,349 WARN {IoThreadPool-1} > org.mrcp4j.server.SESSION EXCEPTION: Eine vorhandene Verbindung > wurde vom Remotehost geschlossen > > java.io.IOException: Eine vorhandene Verbindung wurde vom > Remotehost geschlossen > > at sun.nio.ch.SocketDispatcher.read0(Native Method) > at sun.nio.ch.SocketDispatcher.read(SocketDispatcher.java:25) > at sun.nio.ch.IOUtil.readIntoNativeBuffer(IOUtil.java:233) > at sun.nio.ch.IOUtil.read(IOUtil.java:200) > at > sun.nio.ch.SocketChannelImpl.read(SocketChannelImpl.java:236) > at > org.apache.mina.io.socket.SocketIoProcessor.read(SocketIoProcessor.java:265) > at > org.apache.mina.io.socket.SocketIoProcessor.processSessions(SocketIoProcessor.java:238) > at > org.apache.mina.io.socket.SocketIoProcessor.access$200(SocketIoProcessor.java:42) > at > org.apache.mina.io.socket.SocketIoProcessor$Worker.run(SocketIoProcessor.java:555) > > For the receiver: > > Resource bound and waiting... > 2010-11-15 13:05:27,967 WARN {RMI TCP > Connection(2)-134.96.189.199} > org.speechforge.cairo.server.recog.RTPRecogChannel > No recengine to return to pool! > 2010-11-15 13:05:29,390 WARN {IoThreadPool-3} > org.mrcp4j.server.SESSION EXCEPTION: Eine vorhandene Verbindung > wurde vom Remotehost geschlossen > > java.io.IOException: Eine vorhandene Verbindung wurde vom > Remotehost geschlossen > > at sun.nio.ch.SocketDispatcher.read0(Native Method) > at sun.nio.ch.SocketDispatcher.read(SocketDispatcher.java:25) > at sun.nio.ch.IOUtil.readIntoNativeBuffer(IOUtil.java:233) > at sun.nio.ch.IOUtil.read(IOUtil.java:200) > at > sun.nio.ch.SocketChannelImpl.read(SocketChannelImpl.java:236) > at > org.apache.mina.io.socket.SocketIoProcessor.read(SocketIoProcessor.java:265) > at > org.apache.mina.io.socket.SocketIoProcessor.processSessions(SocketIoProcessor.java:238) > at > org.apache.mina.io.socket.SocketIoProcessor.access$200(SocketIoProcessor.java:42) > at > org.apache.mina.io.socket.SocketIoProcessor$Worker.run(SocketIoProcessor.java:555) > > > Am 15.11.2010 18:51, schrieb spencer lord: >> Hi, >> The unhandled SIP 183 message, should not be causing issues with >> demos. It is just an informational message. I do not think >> that any actions is needed in response to this message with MRCP >> clients. I could be wrong, but that is another discussion >> >> What do you see in the transmitter window when you run the >> synthclient? >> >> Spencer >> >> >> On Mon, Nov 15, 2010 at 8:44 AM, Christian Schulz >> <chs...@df... <mailto:chs...@df...>> wrote: >> >> Hi Spencer >> >> thanks for your reply, unfortunately your guess did not help. >> The error message is still the same. I was trying all demos >> including the speech synth client. There the error message is >> slightly different maybe it gives you a hint for what reason >> the demos are not working out. Btw did you check it out if >> the demos for download are running on your machine? >> >> 2010-11-15 17:27:22,158 INFO {main} >> org.speechforge.cairo.demo.tts.SpeechSynthClient >> >> Sending a SIP invitation to the cairo server. >> 2010-11-15 17:27:22,767 WARN {EventScannerThread} >> org.speechforge.cairo.sip.SipListenerImpl >> >> Received an unhandled SIP response status code (ignoring >> it): 183 : Session progress >> 2010-11-15 17:27:22,839 INFO {main} >> org.speechforge.cairo.demo.tts.SpeechSynthClient >> Received the SIP Response. >> 2010-11-15 17:27:36,408 INFO {Thread-1} >> org.speechforge.cairo.sip.SimpleSipAgent >> Sent a SIP BYE. >> C:\Users\chschulz\sources\nlp\cairo\cairo-0.3\demo\bin> >> >> Best, >> >> Chris >> >> >> Am 15.11.2010 17:06, schrieb spencer lord: >>> Hi Chris, >>> >>> It looks like it may be a problem with jmf. Cairo uses JMF >>> for RTP streaming. Did you install it on the machine you >>> are using? The jmf download link on Cairo installation web >>> page is stale. Sorry, we need to fix that. You can >>> download it here >>> >>> http://www.oracle.com/technetwork/java/javase/download-142937.html >>> >>> If it is already installed, make sue that jmf.jar nd >>> sound.jar are in the lib.ext directory of the jre that you >>> are using. >>> >>> Let me know if that solves the problem. >>> >>> Spencer >>> >>> On Mon, Nov 15, 2010 at 6:40 AM, Christian Schulz >>> <chs...@df... <mailto:chs...@df...>> wrote: >>> >>> Hi all, >>> >>> I am new to the cairo project and I was trying out the >>> demos you can >>> download here: >>> http://sourceforge.net/projects/cairo/files/ >>> >>> Following the instructions here >>> http://www.speechforge.org/projects/cairo/intro.html >>> (that is identical >>> to the instruction in the readme file included in the >>> download) I >>> however cannot get the demo applications run. In >>> particular the >>> recognizer will output this error, while trying to >>> connect to the receiver: >>> >>> Sending a SIP invitation to the cairo server. >>> 2010-11-15 13:05:07,296 WARN {EventScannerThread} >>> org.speechforge.cairo.sip.SipListenerImpl >>> Received an unhandled SIP response status code >>> (ignoring it): 183 : >>> Session progress >>> 2010-11-15 13:05:07,861 INFO {main} >>> org.speechforge.cairo.demo.recog.RecognitionClient >>> Received the SIP Response. >>> java.lang.NullPointerException >>> 2010-11-15 13:05:08,397 WARN {Thread-7} >>> org.speechforge.cairo.rtp.RTPPlayer playSource(): >>> encountered unexpected >>> exception: >>> javax.media.NoDataSourceException: Error instantiating >>> class: >>> com.sun.media.protocol.dsound.DataSource : >>> java.lang.NullPointerException >>> at >>> javax.media.Manager.createDataSource(Manager.java:1012) >>> at >>> org.speechforge.cairo.rtp.RTPPlayer.playSource(RTPPlayer.java:110) >>> at >>> org.speechforge.cairo.rtp.NativeMediaClient$TransmitThread.run(NativeMediaClient.java:105) >>> Exception in thread "Thread-7" >>> java.lang.RuntimeException: playSource() >>> encountered unexpected exception >>> at >>> org.speechforge.cairo.rtp.RTPPlayer.playSource(RTPPlayer.java:153) >>> at >>> org.speechforge.cairo.rtp.NativeMediaClient$TransmitThread.run(NativeMediaClient.java:105) >>> Caused by: javax.media.NoDataSourceException: Error >>> instantiating class: >>> com.sun.media.protocol.dsound.DataSource : >>> java.lang.NullPointerException >>> at >>> javax.media.Manager.createDataSource(Manager.java:1012) >>> at >>> org.speechforge.cairo.rtp.RTPPlayer.playSource(RTPPlayer.java:110) >>> ... 1 more >>> 2010-11-15 13:05:18,577 WARN {main} >>> org.speechforge.cairo.demo.recog.RecognitionClient >>> org.mrcp4j.client.MrcpInvocationException: MRCP >>> response contains an >>> error code, the request invocation did not complete >>> successfully. >>> org.mrcp4j.client.MrcpInvocationException: MRCP response >>> contains an >>> error code, the request invocation did not complete >>> successfully. >>> at >>> org.mrcp4j.client.MrcpChannel.sendRequest(MrcpChannel.java:143) >>> at >>> org.speechforge.cairo.demo.recog.RecognitionClient.doRecognize(RecognitionClient.java:164) >>> at >>> org.speechforge.cairo.demo.recog.RecognitionClient.main(RecognitionCl >>> ient.java:358) >>> 2010-11-15 13:05:27,929 INFO {main} >>> org.speechforge.cairo.sip.SimpleSipAgent >>> Sent a SIP BYE. >>> >>> Thanks for your help! >>> >>> chris >>> >>> ------------------------------------------------------------------------------ >>> Centralized Desktop Delivery: Dell and VMware Reference >>> Architecture >>> Simplifying enterprise desktop deployment and management >>> using >>> Dell EqualLogic storage and VMware View: A highly >>> scalable, end-to-end >>> client virtualization framework. Read more! >>> http://p.sf.net/sfu/dell-eql-dev2dev >>> _______________________________________________ >>> cairo-user mailing list >>> cai...@li... >>> <mailto:cai...@li...> >>> https://lists.sourceforge.net/lists/listinfo/cairo-user >>> >>> >>> >>> ------------------------------------------------------------------------------ >>> Centralized Desktop Delivery: Dell and VMware Reference Architecture >>> Simplifying enterprise desktop deployment and management using >>> Dell EqualLogic storage and VMware View: A highly scalable, end-to-end >>> client virtualization framework. Read more! >>> http://p.sf.net/sfu/dell-eql-dev2dev >>> >>> >>> _______________________________________________ >>> cairo-user mailing list >>> cai...@li... <mailto:cai...@li...> >>> https://lists.sourceforge.net/lists/listinfo/cairo-user >> >> -- >> Christian H. Schulz >> DFKI GmbH, Campus D3 2 >> Stuhlsatzenhausweg 3 >> D-66123 Saarbrücken, Germany >> +49 (0)681 85775-5371 (tel.) -5021 (fax) >> mail:chs...@df... <mailto:chs...@df...>, http:www.dfki.de/~chschulz <http://www.dfki.de/%7Echschulz> >> >> ------------------------------------------------------------------ >> Deutsches Forschungszentrum fuer Kuenstliche Intelligenz GmbH >> Firmensitz: Trippstadter Strasse 122, D-67663 Kaiserslautern >> Geschaeftsfuehrung: >> Prof. Dr. Dr. h.c. mult. Wolfgang Wahlster (Vorsitzender) >> Dr. Walter Olthoff >> Vorsitzender des Aufsichtsrats: Prof. Dr. h.c. Hans A. Aukes >> Amtsgericht Kaiserslautern, HRB 2313 >> >> >> ------------------------------------------------------------------------------ >> Centralized Desktop Delivery: Dell and VMware Reference >> Architecture >> Simplifying enterprise desktop deployment and management using >> Dell EqualLogic storage and VMware View: A highly scalable, >> end-to-end >> client virtualization framework. Read more! >> http://p.sf.net/sfu/dell-eql-dev2dev >> _______________________________________________ >> cairo-user mailing list >> cai...@li... >> <mailto:cai...@li...> >> https://lists.sourceforge.net/lists/listinfo/cairo-user >> >> >> >> ------------------------------------------------------------------------------ >> Centralized Desktop Delivery: Dell and VMware Reference Architecture >> Simplifying enterprise desktop deployment and management using >> Dell EqualLogic storage and VMware View: A highly scalable, end-to-end >> client virtualization framework. Read more! >> http://p.sf.net/sfu/dell-eql-dev2dev >> >> >> _______________________________________________ >> cairo-user mailing list >> cai...@li... <mailto:cai...@li...> >> https://lists.sourceforge.net/lists/listinfo/cairo-user > > -- > Christian H. Schulz > DFKI GmbH, Campus D3 2 > Stuhlsatzenhausweg 3 > D-66123 Saarbrücken, Germany > +49 (0)681 85775-5371 (tel.) -5021 (fax) > mail:chs...@df... <mailto:chs...@df...>, http:www.dfki.de/~chschulz <http://www.dfki.de/%7Echschulz> > > ------------------------------------------------------------------ > Deutsches Forschungszentrum fuer Kuenstliche Intelligenz GmbH > Firmensitz: Trippstadter Strasse 122, D-67663 Kaiserslautern > Geschaeftsfuehrung: > Prof. Dr. Dr. h.c. mult. Wolfgang Wahlster (Vorsitzender) > Dr. Walter Olthoff > Vorsitzender des Aufsichtsrats: Prof. Dr. h.c. Hans A. Aukes > Amtsgericht Kaiserslautern, HRB 2313 > > > ------------------------------------------------------------------------------ > Centralized Desktop Delivery: Dell and VMware Reference Architecture > Simplifying enterprise desktop deployment and management using > Dell EqualLogic storage and VMware View: A highly scalable, end-to-end > client virtualization framework. Read more! > http://p.sf.net/sfu/dell-eql-dev2dev > _______________________________________________ > cairo-user mailing list > cai...@li... > <mailto:cai...@li...> > https://lists.sourceforge.net/lists/listinfo/cairo-user > > > > ------------------------------------------------------------------------------ > Centralized Desktop Delivery: Dell and VMware Reference Architecture > Simplifying enterprise desktop deployment and management using > Dell EqualLogic storage and VMware View: A highly scalable, end-to-end > client virtualization framework. Read more! > http://p.sf.net/sfu/dell-eql-dev2dev > > > _______________________________________________ > cairo-user mailing list > cai...@li... > https://lists.sourceforge.net/lists/listinfo/cairo-user -- Christian H. Schulz DFKI GmbH, Campus D3 2 Stuhlsatzenhausweg 3 D-66123 Saarbrücken, Germany +49 (0)681 85775-5371 (tel.) -5021 (fax) mail: chs...@df..., http: www.dfki.de/~chschulz ------------------------------------------------------------------ Deutsches Forschungszentrum fuer Kuenstliche Intelligenz GmbH Firmensitz: Trippstadter Strasse 122, D-67663 Kaiserslautern Geschaeftsfuehrung: Prof. Dr. Dr. h.c. mult. Wolfgang Wahlster (Vorsitzender) Dr. Walter Olthoff Vorsitzender des Aufsichtsrats: Prof. Dr. h.c. Hans A. Aukes Amtsgericht Kaiserslautern, HRB 2313 |
From: spencer l. <spe...@gm...> - 2010-11-15 18:08:13
|
This exception in the transmitter is ok too. It is low priority known problem, regarding the way the mrcp channels are shutdown. MRCP4J complains with this message if things are not closed in certain way -- but it handles the situation. At the very least you should be able to run the demos. Are you sure you do not hear any audio when you run synthClient? I think it should be working. Is it possible the volume is turned down? On Mon, Nov 15, 2010 at 9:53 AM, Christian Schulz <chs...@df...> wrote: > For the transmitter: > > Resource bound and waiting... > Wrote synthesized speech to C:\temp\cairo\basePromptDir\ > 12c4f7a65f6@speechsynth\1289823631677.au<12c4f7a65f6@speechsynth%5C1289823631677.au> > 2010-11-15 13:20:45,349 WARN {IoThreadPool-1} org.mrcp4j.server.SESSION > EXCEPTION: Eine vorhandene Verbindung wurde vom Remotehost geschlossen > > java.io.IOException: Eine vorhandene Verbindung wurde vom Remotehost > geschlossen > > at sun.nio.ch.SocketDispatcher.read0(Native Method) > at sun.nio.ch.SocketDispatcher.read(SocketDispatcher.java:25) > at sun.nio.ch.IOUtil.readIntoNativeBuffer(IOUtil.java:233) > at sun.nio.ch.IOUtil.read(IOUtil.java:200) > at sun.nio.ch.SocketChannelImpl.read(SocketChannelImpl.java:236) > at > org.apache.mina.io.socket.SocketIoProcessor.read(SocketIoProcessor.java:265) > at > org.apache.mina.io.socket.SocketIoProcessor.processSessions(SocketIoProcessor.java:238) > at > org.apache.mina.io.socket.SocketIoProcessor.access$200(SocketIoProcessor.java:42) > at > org.apache.mina.io.socket.SocketIoProcessor$Worker.run(SocketIoProcessor.java:555) > > For the receiver: > > Resource bound and waiting... > 2010-11-15 13:05:27,967 WARN {RMI TCP Connection(2)-134.96.189.199} > org.speechforge.cairo.server.recog.RTPRecogChannel > No recengine to return to pool! > 2010-11-15 13:05:29,390 WARN {IoThreadPool-3} org.mrcp4j.server.SESSION > EXCEPTION: Eine vorhandene Verbindung wurde vom Remotehost geschlossen > > java.io.IOException: Eine vorhandene Verbindung wurde vom Remotehost > geschlossen > > at sun.nio.ch.SocketDispatcher.read0(Native Method) > at sun.nio.ch.SocketDispatcher.read(SocketDispatcher.java:25) > at sun.nio.ch.IOUtil.readIntoNativeBuffer(IOUtil.java:233) > at sun.nio.ch.IOUtil.read(IOUtil.java:200) > at sun.nio.ch.SocketChannelImpl.read(SocketChannelImpl.java:236) > at > org.apache.mina.io.socket.SocketIoProcessor.read(SocketIoProcessor.java:265) > at > org.apache.mina.io.socket.SocketIoProcessor.processSessions(SocketIoProcessor.java:238) > at > org.apache.mina.io.socket.SocketIoProcessor.access$200(SocketIoProcessor.java:42) > at > org.apache.mina.io.socket.SocketIoProcessor$Worker.run(SocketIoProcessor.java:555) > > > Am 15.11.2010 18:51, schrieb spencer lord: > > Hi, > The unhandled SIP 183 message, should not be causing issues with demos. > It is just an informational message. I do not think that any actions is > needed in response to this message with MRCP clients. I could be wrong, but > that is another discussion > > What do you see in the transmitter window when you run the synthclient? > > Spencer > > > On Mon, Nov 15, 2010 at 8:44 AM, Christian Schulz <chs...@df...>wrote: > >> Hi Spencer >> >> thanks for your reply, unfortunately your guess did not help. The error >> message is still the same. I was trying all demos including the speech synth >> client. There the error message is slightly different maybe it gives you a >> hint for what reason the demos are not working out. Btw did you check it out >> if the demos for download are running on your machine? >> >> 2010-11-15 17:27:22,158 INFO {main} >> org.speechforge.cairo.demo.tts.SpeechSynthClient >> >> Sending a SIP invitation to the cairo server. >> 2010-11-15 17:27:22,767 WARN {EventScannerThread} >> org.speechforge.cairo.sip.SipListenerImpl >> >> Received an unhandled SIP response status code (ignoring it): 183 : >> Session progress >> 2010-11-15 17:27:22,839 INFO {main} >> org.speechforge.cairo.demo.tts.SpeechSynthClient >> Received the SIP Response. >> 2010-11-15 17:27:36,408 INFO {Thread-1} >> org.speechforge.cairo.sip.SimpleSipAgent >> Sent a SIP BYE. >> C:\Users\chschulz\sources\nlp\cairo\cairo-0.3\demo\bin> >> >> Best, >> >> Chris >> >> >> Am 15.11.2010 17:06, schrieb spencer lord: >> >> Hi Chris, >> >> It looks like it may be a problem with jmf. Cairo uses JMF for RTP >> streaming. Did you install it on the machine you are using? The jmf >> download link on Cairo installation web page is stale. Sorry, we need to >> fix that. You can download it here >> >> http://www.oracle.com/technetwork/java/javase/download-142937.html >> >> If it is already installed, make sue that jmf.jar nd sound.jar are in the >> lib.ext directory of the jre that you are using. >> >> Let me know if that solves the problem. >> >> Spencer >> >> On Mon, Nov 15, 2010 at 6:40 AM, Christian Schulz <chs...@df...>wrote: >> >>> Hi all, >>> >>> I am new to the cairo project and I was trying out the demos you can >>> download here: >>> http://sourceforge.net/projects/cairo/files/ >>> >>> Following the instructions here >>> http://www.speechforge.org/projects/cairo/intro.html (that is identical >>> to the instruction in the readme file included in the download) I >>> however cannot get the demo applications run. In particular the >>> recognizer will output this error, while trying to connect to the >>> receiver: >>> >>> Sending a SIP invitation to the cairo server. >>> 2010-11-15 13:05:07,296 WARN {EventScannerThread} >>> org.speechforge.cairo.sip.SipListenerImpl >>> Received an unhandled SIP response status code (ignoring it): 183 : >>> Session progress >>> 2010-11-15 13:05:07,861 INFO {main} >>> org.speechforge.cairo.demo.recog.RecognitionClient >>> Received the SIP Response. >>> java.lang.NullPointerException >>> 2010-11-15 13:05:08,397 WARN {Thread-7} >>> org.speechforge.cairo.rtp.RTPPlayer playSource(): encountered unexpected >>> exception: >>> javax.media.NoDataSourceException: Error instantiating class: >>> com.sun.media.protocol.dsound.DataSource : java.lang.NullPointerException >>> at javax.media.Manager.createDataSource(Manager.java:1012) >>> at >>> org.speechforge.cairo.rtp.RTPPlayer.playSource(RTPPlayer.java:110) >>> at >>> >>> org.speechforge.cairo.rtp.NativeMediaClient$TransmitThread.run(NativeMediaClient.java:105) >>> Exception in thread "Thread-7" java.lang.RuntimeException: playSource() >>> encountered unexpected exception >>> at >>> org.speechforge.cairo.rtp.RTPPlayer.playSource(RTPPlayer.java:153) >>> at >>> >>> org.speechforge.cairo.rtp.NativeMediaClient$TransmitThread.run(NativeMediaClient.java:105) >>> Caused by: javax.media.NoDataSourceException: Error instantiating class: >>> com.sun.media.protocol.dsound.DataSource : java.lang.NullPointerException >>> at javax.media.Manager.createDataSource(Manager.java:1012) >>> at >>> org.speechforge.cairo.rtp.RTPPlayer.playSource(RTPPlayer.java:110) >>> ... 1 more >>> 2010-11-15 13:05:18,577 WARN {main} >>> org.speechforge.cairo.demo.recog.RecognitionClient >>> org.mrcp4j.client.MrcpInvocationException: MRCP response contains an >>> error code, the request invocation did not complete successfully. >>> org.mrcp4j.client.MrcpInvocationException: MRCP response contains an >>> error code, the request invocation did not complete successfully. >>> at >>> org.mrcp4j.client.MrcpChannel.sendRequest(MrcpChannel.java:143) >>> at >>> >>> org.speechforge.cairo.demo.recog.RecognitionClient.doRecognize(RecognitionClient.java:164) >>> at >>> org.speechforge.cairo.demo.recog.RecognitionClient.main(RecognitionCl >>> ient.java:358) >>> 2010-11-15 13:05:27,929 INFO {main} >>> org.speechforge.cairo.sip.SimpleSipAgent >>> Sent a SIP BYE. >>> >>> Thanks for your help! >>> >>> chris >>> >>> >>> ------------------------------------------------------------------------------ >>> Centralized Desktop Delivery: Dell and VMware Reference Architecture >>> Simplifying enterprise desktop deployment and management using >>> Dell EqualLogic storage and VMware View: A highly scalable, end-to-end >>> client virtualization framework. Read more! >>> http://p.sf.net/sfu/dell-eql-dev2dev >>> _______________________________________________ >>> cairo-user mailing list >>> cai...@li... >>> https://lists.sourceforge.net/lists/listinfo/cairo-user >>> >> >> >> ------------------------------------------------------------------------------ >> Centralized Desktop Delivery: Dell and VMware Reference Architecture >> Simplifying enterprise desktop deployment and management using >> Dell EqualLogic storage and VMware View: A highly scalable, end-to-end >> client virtualization framework. Read more!http://p.sf.net/sfu/dell-eql-dev2dev >> >> >> _______________________________________________ >> cairo-user mailing lis...@li...https://lists.sourceforge.net/lists/listinfo/cairo-user >> >> >> -- >> Christian H. Schulz >> DFKI GmbH, Campus D3 2 >> Stuhlsatzenhausweg 3 >> D-66123 Saarbrücken, Germany >> +49 (0)681 85775-5371 (tel.) -5021 (fax) >> mail: chs...@df..., http: www.dfki.de/~chschulz <http://www.dfki.de/%7Echschulz> >> >> ------------------------------------------------------------------ >> Deutsches Forschungszentrum fuer Kuenstliche Intelligenz GmbH >> Firmensitz: Trippstadter Strasse 122, D-67663 Kaiserslautern >> Geschaeftsfuehrung: >> Prof. Dr. Dr. h.c. mult. Wolfgang Wahlster (Vorsitzender) >> Dr. Walter Olthoff >> Vorsitzender des Aufsichtsrats: Prof. Dr. h.c. Hans A. Aukes >> Amtsgericht Kaiserslautern, HRB 2313 >> >> >> >> ------------------------------------------------------------------------------ >> Centralized Desktop Delivery: Dell and VMware Reference Architecture >> Simplifying enterprise desktop deployment and management using >> Dell EqualLogic storage and VMware View: A highly scalable, end-to-end >> client virtualization framework. Read more! >> http://p.sf.net/sfu/dell-eql-dev2dev >> _______________________________________________ >> cairo-user mailing list >> cai...@li... >> https://lists.sourceforge.net/lists/listinfo/cairo-user >> >> > > ------------------------------------------------------------------------------ > Centralized Desktop Delivery: Dell and VMware Reference Architecture > Simplifying enterprise desktop deployment and management using > Dell EqualLogic storage and VMware View: A highly scalable, end-to-end > client virtualization framework. Read more!http://p.sf.net/sfu/dell-eql-dev2dev > > > _______________________________________________ > cairo-user mailing lis...@li...https://lists.sourceforge.net/lists/listinfo/cairo-user > > > -- > Christian H. Schulz > DFKI GmbH, Campus D3 2 > Stuhlsatzenhausweg 3 > D-66123 Saarbrücken, Germany > +49 (0)681 85775-5371 (tel.) -5021 (fax) > mail: chs...@df..., http: www.dfki.de/~chschulz <http://www.dfki.de/%7Echschulz> > > ------------------------------------------------------------------ > Deutsches Forschungszentrum fuer Kuenstliche Intelligenz GmbH > Firmensitz: Trippstadter Strasse 122, D-67663 Kaiserslautern > Geschaeftsfuehrung: > Prof. Dr. Dr. h.c. mult. Wolfgang Wahlster (Vorsitzender) > Dr. Walter Olthoff > Vorsitzender des Aufsichtsrats: Prof. Dr. h.c. Hans A. Aukes > Amtsgericht Kaiserslautern, HRB 2313 > > > > ------------------------------------------------------------------------------ > Centralized Desktop Delivery: Dell and VMware Reference Architecture > Simplifying enterprise desktop deployment and management using > Dell EqualLogic storage and VMware View: A highly scalable, end-to-end > client virtualization framework. Read more! > http://p.sf.net/sfu/dell-eql-dev2dev > _______________________________________________ > cairo-user mailing list > cai...@li... > https://lists.sourceforge.net/lists/listinfo/cairo-user > > |
From: Christian S. <chs...@df...> - 2010-11-15 17:54:03
|
For the transmitter: Resource bound and waiting... Wrote synthesized speech to C:\temp\cairo\basePromptDir\12c4f7a65f6@speechsynth\1289823631677.au 2010-11-15 13:20:45,349 WARN {IoThreadPool-1} org.mrcp4j.server.SESSION EXCEPTION: Eine vorhandene Verbindung wurde vom Remotehost geschlossen java.io.IOException: Eine vorhandene Verbindung wurde vom Remotehost geschlossen at sun.nio.ch.SocketDispatcher.read0(Native Method) at sun.nio.ch.SocketDispatcher.read(SocketDispatcher.java:25) at sun.nio.ch.IOUtil.readIntoNativeBuffer(IOUtil.java:233) at sun.nio.ch.IOUtil.read(IOUtil.java:200) at sun.nio.ch.SocketChannelImpl.read(SocketChannelImpl.java:236) at org.apache.mina.io.socket.SocketIoProcessor.read(SocketIoProcessor.java:265) at org.apache.mina.io.socket.SocketIoProcessor.processSessions(SocketIoProcessor.java:238) at org.apache.mina.io.socket.SocketIoProcessor.access$200(SocketIoProcessor.java:42) at org.apache.mina.io.socket.SocketIoProcessor$Worker.run(SocketIoProcessor.java:555) For the receiver: Resource bound and waiting... 2010-11-15 13:05:27,967 WARN {RMI TCP Connection(2)-134.96.189.199} org.speechforge.cairo.server.recog.RTPRecogChannel No recengine to return to pool! 2010-11-15 13:05:29,390 WARN {IoThreadPool-3} org.mrcp4j.server.SESSION EXCEPTION: Eine vorhandene Verbindung wurde vom Remotehost geschlossen java.io.IOException: Eine vorhandene Verbindung wurde vom Remotehost geschlossen at sun.nio.ch.SocketDispatcher.read0(Native Method) at sun.nio.ch.SocketDispatcher.read(SocketDispatcher.java:25) at sun.nio.ch.IOUtil.readIntoNativeBuffer(IOUtil.java:233) at sun.nio.ch.IOUtil.read(IOUtil.java:200) at sun.nio.ch.SocketChannelImpl.read(SocketChannelImpl.java:236) at org.apache.mina.io.socket.SocketIoProcessor.read(SocketIoProcessor.java:265) at org.apache.mina.io.socket.SocketIoProcessor.processSessions(SocketIoProcessor.java:238) at org.apache.mina.io.socket.SocketIoProcessor.access$200(SocketIoProcessor.java:42) at org.apache.mina.io.socket.SocketIoProcessor$Worker.run(SocketIoProcessor.java:555) Am 15.11.2010 18:51, schrieb spencer lord: > Hi, > The unhandled SIP 183 message, should not be causing issues with > demos. It is just an informational message. I do not think that any > actions is needed in response to this message with MRCP clients. I > could be wrong, but that is another discussion > > What do you see in the transmitter window when you run the synthclient? > > Spencer > > > On Mon, Nov 15, 2010 at 8:44 AM, Christian Schulz <chs...@df... > <mailto:chs...@df...>> wrote: > > Hi Spencer > > thanks for your reply, unfortunately your guess did not help. The > error message is still the same. I was trying all demos including > the speech synth client. There the error message is slightly > different maybe it gives you a hint for what reason the demos are > not working out. Btw did you check it out if the demos for > download are running on your machine? > > 2010-11-15 17:27:22,158 INFO {main} > org.speechforge.cairo.demo.tts.SpeechSynthClient > > Sending a SIP invitation to the cairo server. > 2010-11-15 17:27:22,767 WARN {EventScannerThread} > org.speechforge.cairo.sip.SipListenerImpl > > Received an unhandled SIP response status code (ignoring it): 183 > : Session progress > 2010-11-15 17:27:22,839 INFO {main} > org.speechforge.cairo.demo.tts.SpeechSynthClient > Received the SIP Response. > 2010-11-15 17:27:36,408 INFO {Thread-1} > org.speechforge.cairo.sip.SimpleSipAgent > Sent a SIP BYE. > C:\Users\chschulz\sources\nlp\cairo\cairo-0.3\demo\bin> > > Best, > > Chris > > > Am 15.11.2010 17:06, schrieb spencer lord: >> Hi Chris, >> >> It looks like it may be a problem with jmf. Cairo uses JMF for >> RTP streaming. Did you install it on the machine you are >> using? The jmf download link on Cairo installation web page is >> stale. Sorry, we need to fix that. You can download it here >> >> http://www.oracle.com/technetwork/java/javase/download-142937.html >> >> If it is already installed, make sue that jmf.jar nd sound.jar >> are in the lib.ext directory of the jre that you are using. >> >> Let me know if that solves the problem. >> >> Spencer >> >> On Mon, Nov 15, 2010 at 6:40 AM, Christian Schulz >> <chs...@df... <mailto:chs...@df...>> wrote: >> >> Hi all, >> >> I am new to the cairo project and I was trying out the demos >> you can >> download here: >> http://sourceforge.net/projects/cairo/files/ >> >> Following the instructions here >> http://www.speechforge.org/projects/cairo/intro.html (that is >> identical >> to the instruction in the readme file included in the download) I >> however cannot get the demo applications run. In particular the >> recognizer will output this error, while trying to connect to >> the receiver: >> >> Sending a SIP invitation to the cairo server. >> 2010-11-15 13:05:07,296 WARN {EventScannerThread} >> org.speechforge.cairo.sip.SipListenerImpl >> Received an unhandled SIP response status code (ignoring >> it): 183 : >> Session progress >> 2010-11-15 13:05:07,861 INFO {main} >> org.speechforge.cairo.demo.recog.RecognitionClient >> Received the SIP Response. >> java.lang.NullPointerException >> 2010-11-15 13:05:08,397 WARN {Thread-7} >> org.speechforge.cairo.rtp.RTPPlayer playSource(): encountered >> unexpected >> exception: >> javax.media.NoDataSourceException: Error instantiating class: >> com.sun.media.protocol.dsound.DataSource : >> java.lang.NullPointerException >> at >> javax.media.Manager.createDataSource(Manager.java:1012) >> at >> org.speechforge.cairo.rtp.RTPPlayer.playSource(RTPPlayer.java:110) >> at >> org.speechforge.cairo.rtp.NativeMediaClient$TransmitThread.run(NativeMediaClient.java:105) >> Exception in thread "Thread-7" java.lang.RuntimeException: >> playSource() >> encountered unexpected exception >> at >> org.speechforge.cairo.rtp.RTPPlayer.playSource(RTPPlayer.java:153) >> at >> org.speechforge.cairo.rtp.NativeMediaClient$TransmitThread.run(NativeMediaClient.java:105) >> Caused by: javax.media.NoDataSourceException: Error >> instantiating class: >> com.sun.media.protocol.dsound.DataSource : >> java.lang.NullPointerException >> at >> javax.media.Manager.createDataSource(Manager.java:1012) >> at >> org.speechforge.cairo.rtp.RTPPlayer.playSource(RTPPlayer.java:110) >> ... 1 more >> 2010-11-15 13:05:18,577 WARN {main} >> org.speechforge.cairo.demo.recog.RecognitionClient >> org.mrcp4j.client.MrcpInvocationException: MRCP response >> contains an >> error code, the request invocation did not complete successfully. >> org.mrcp4j.client.MrcpInvocationException: MRCP response >> contains an >> error code, the request invocation did not complete successfully. >> at >> org.mrcp4j.client.MrcpChannel.sendRequest(MrcpChannel.java:143) >> at >> org.speechforge.cairo.demo.recog.RecognitionClient.doRecognize(RecognitionClient.java:164) >> at >> org.speechforge.cairo.demo.recog.RecognitionClient.main(RecognitionCl >> ient.java:358) >> 2010-11-15 13:05:27,929 INFO {main} >> org.speechforge.cairo.sip.SimpleSipAgent >> Sent a SIP BYE. >> >> Thanks for your help! >> >> chris >> >> ------------------------------------------------------------------------------ >> Centralized Desktop Delivery: Dell and VMware Reference >> Architecture >> Simplifying enterprise desktop deployment and management using >> Dell EqualLogic storage and VMware View: A highly scalable, >> end-to-end >> client virtualization framework. Read more! >> http://p.sf.net/sfu/dell-eql-dev2dev >> _______________________________________________ >> cairo-user mailing list >> cai...@li... >> <mailto:cai...@li...> >> https://lists.sourceforge.net/lists/listinfo/cairo-user >> >> >> >> ------------------------------------------------------------------------------ >> Centralized Desktop Delivery: Dell and VMware Reference Architecture >> Simplifying enterprise desktop deployment and management using >> Dell EqualLogic storage and VMware View: A highly scalable, end-to-end >> client virtualization framework. Read more! >> http://p.sf.net/sfu/dell-eql-dev2dev >> >> >> _______________________________________________ >> cairo-user mailing list >> cai...@li... <mailto:cai...@li...> >> https://lists.sourceforge.net/lists/listinfo/cairo-user > > -- > Christian H. Schulz > DFKI GmbH, Campus D3 2 > Stuhlsatzenhausweg 3 > D-66123 Saarbrücken, Germany > +49 (0)681 85775-5371 (tel.) -5021 (fax) > mail:chs...@df... <mailto:chs...@df...>, http:www.dfki.de/~chschulz <http://www.dfki.de/%7Echschulz> > > ------------------------------------------------------------------ > Deutsches Forschungszentrum fuer Kuenstliche Intelligenz GmbH > Firmensitz: Trippstadter Strasse 122, D-67663 Kaiserslautern > Geschaeftsfuehrung: > Prof. Dr. Dr. h.c. mult. Wolfgang Wahlster (Vorsitzender) > Dr. Walter Olthoff > Vorsitzender des Aufsichtsrats: Prof. Dr. h.c. Hans A. Aukes > Amtsgericht Kaiserslautern, HRB 2313 > > > ------------------------------------------------------------------------------ > Centralized Desktop Delivery: Dell and VMware Reference Architecture > Simplifying enterprise desktop deployment and management using > Dell EqualLogic storage and VMware View: A highly scalable, end-to-end > client virtualization framework. Read more! > http://p.sf.net/sfu/dell-eql-dev2dev > _______________________________________________ > cairo-user mailing list > cai...@li... > <mailto:cai...@li...> > https://lists.sourceforge.net/lists/listinfo/cairo-user > > > > ------------------------------------------------------------------------------ > Centralized Desktop Delivery: Dell and VMware Reference Architecture > Simplifying enterprise desktop deployment and management using > Dell EqualLogic storage and VMware View: A highly scalable, end-to-end > client virtualization framework. Read more! > http://p.sf.net/sfu/dell-eql-dev2dev > > > _______________________________________________ > cairo-user mailing list > cai...@li... > https://lists.sourceforge.net/lists/listinfo/cairo-user -- Christian H. Schulz DFKI GmbH, Campus D3 2 Stuhlsatzenhausweg 3 D-66123 Saarbrücken, Germany +49 (0)681 85775-5371 (tel.) -5021 (fax) mail: chs...@df..., http: www.dfki.de/~chschulz ------------------------------------------------------------------ Deutsches Forschungszentrum fuer Kuenstliche Intelligenz GmbH Firmensitz: Trippstadter Strasse 122, D-67663 Kaiserslautern Geschaeftsfuehrung: Prof. Dr. Dr. h.c. mult. Wolfgang Wahlster (Vorsitzender) Dr. Walter Olthoff Vorsitzender des Aufsichtsrats: Prof. Dr. h.c. Hans A. Aukes Amtsgericht Kaiserslautern, HRB 2313 |
From: spencer l. <spe...@gm...> - 2010-11-15 17:52:01
|
Hi, The unhandled SIP 183 message, should not be causing issues with demos. It is just an informational message. I do not think that any actions is needed in response to this message with MRCP clients. I could be wrong, but that is another discussion What do you see in the transmitter window when you run the synthclient? Spencer On Mon, Nov 15, 2010 at 8:44 AM, Christian Schulz <chs...@df...> wrote: > Hi Spencer > > thanks for your reply, unfortunately your guess did not help. The error > message is still the same. I was trying all demos including the speech synth > client. There the error message is slightly different maybe it gives you a > hint for what reason the demos are not working out. Btw did you check it out > if the demos for download are running on your machine? > > 2010-11-15 17:27:22,158 INFO {main} > org.speechforge.cairo.demo.tts.SpeechSynthClient > > Sending a SIP invitation to the cairo server. > 2010-11-15 17:27:22,767 WARN {EventScannerThread} > org.speechforge.cairo.sip.SipListenerImpl > > Received an unhandled SIP response status code (ignoring it): 183 : > Session progress > 2010-11-15 17:27:22,839 INFO {main} > org.speechforge.cairo.demo.tts.SpeechSynthClient > Received the SIP Response. > 2010-11-15 17:27:36,408 INFO {Thread-1} > org.speechforge.cairo.sip.SimpleSipAgent > Sent a SIP BYE. > C:\Users\chschulz\sources\nlp\cairo\cairo-0.3\demo\bin> > > Best, > > Chris > > > Am 15.11.2010 17:06, schrieb spencer lord: > > Hi Chris, > > It looks like it may be a problem with jmf. Cairo uses JMF for RTP > streaming. Did you install it on the machine you are using? The jmf > download link on Cairo installation web page is stale. Sorry, we need to > fix that. You can download it here > > http://www.oracle.com/technetwork/java/javase/download-142937.html > > If it is already installed, make sue that jmf.jar nd sound.jar are in the > lib.ext directory of the jre that you are using. > > Let me know if that solves the problem. > > Spencer > > On Mon, Nov 15, 2010 at 6:40 AM, Christian Schulz <chs...@df...>wrote: > >> Hi all, >> >> I am new to the cairo project and I was trying out the demos you can >> download here: >> http://sourceforge.net/projects/cairo/files/ >> >> Following the instructions here >> http://www.speechforge.org/projects/cairo/intro.html (that is identical >> to the instruction in the readme file included in the download) I >> however cannot get the demo applications run. In particular the >> recognizer will output this error, while trying to connect to the >> receiver: >> >> Sending a SIP invitation to the cairo server. >> 2010-11-15 13:05:07,296 WARN {EventScannerThread} >> org.speechforge.cairo.sip.SipListenerImpl >> Received an unhandled SIP response status code (ignoring it): 183 : >> Session progress >> 2010-11-15 13:05:07,861 INFO {main} >> org.speechforge.cairo.demo.recog.RecognitionClient >> Received the SIP Response. >> java.lang.NullPointerException >> 2010-11-15 13:05:08,397 WARN {Thread-7} >> org.speechforge.cairo.rtp.RTPPlayer playSource(): encountered unexpected >> exception: >> javax.media.NoDataSourceException: Error instantiating class: >> com.sun.media.protocol.dsound.DataSource : java.lang.NullPointerException >> at javax.media.Manager.createDataSource(Manager.java:1012) >> at >> org.speechforge.cairo.rtp.RTPPlayer.playSource(RTPPlayer.java:110) >> at >> >> org.speechforge.cairo.rtp.NativeMediaClient$TransmitThread.run(NativeMediaClient.java:105) >> Exception in thread "Thread-7" java.lang.RuntimeException: playSource() >> encountered unexpected exception >> at >> org.speechforge.cairo.rtp.RTPPlayer.playSource(RTPPlayer.java:153) >> at >> >> org.speechforge.cairo.rtp.NativeMediaClient$TransmitThread.run(NativeMediaClient.java:105) >> Caused by: javax.media.NoDataSourceException: Error instantiating class: >> com.sun.media.protocol.dsound.DataSource : java.lang.NullPointerException >> at javax.media.Manager.createDataSource(Manager.java:1012) >> at >> org.speechforge.cairo.rtp.RTPPlayer.playSource(RTPPlayer.java:110) >> ... 1 more >> 2010-11-15 13:05:18,577 WARN {main} >> org.speechforge.cairo.demo.recog.RecognitionClient >> org.mrcp4j.client.MrcpInvocationException: MRCP response contains an >> error code, the request invocation did not complete successfully. >> org.mrcp4j.client.MrcpInvocationException: MRCP response contains an >> error code, the request invocation did not complete successfully. >> at org.mrcp4j.client.MrcpChannel.sendRequest(MrcpChannel.java:143) >> at >> >> org.speechforge.cairo.demo.recog.RecognitionClient.doRecognize(RecognitionClient.java:164) >> at >> org.speechforge.cairo.demo.recog.RecognitionClient.main(RecognitionCl >> ient.java:358) >> 2010-11-15 13:05:27,929 INFO {main} >> org.speechforge.cairo.sip.SimpleSipAgent >> Sent a SIP BYE. >> >> Thanks for your help! >> >> chris >> >> >> ------------------------------------------------------------------------------ >> Centralized Desktop Delivery: Dell and VMware Reference Architecture >> Simplifying enterprise desktop deployment and management using >> Dell EqualLogic storage and VMware View: A highly scalable, end-to-end >> client virtualization framework. Read more! >> http://p.sf.net/sfu/dell-eql-dev2dev >> _______________________________________________ >> cairo-user mailing list >> cai...@li... >> https://lists.sourceforge.net/lists/listinfo/cairo-user >> > > > ------------------------------------------------------------------------------ > Centralized Desktop Delivery: Dell and VMware Reference Architecture > Simplifying enterprise desktop deployment and management using > Dell EqualLogic storage and VMware View: A highly scalable, end-to-end > client virtualization framework. Read more!http://p.sf.net/sfu/dell-eql-dev2dev > > > _______________________________________________ > cairo-user mailing lis...@li...https://lists.sourceforge.net/lists/listinfo/cairo-user > > > -- > Christian H. Schulz > DFKI GmbH, Campus D3 2 > Stuhlsatzenhausweg 3 > D-66123 Saarbrücken, Germany > +49 (0)681 85775-5371 (tel.) -5021 (fax) > mail: chs...@df..., http: www.dfki.de/~chschulz <http://www.dfki.de/%7Echschulz> > > ------------------------------------------------------------------ > Deutsches Forschungszentrum fuer Kuenstliche Intelligenz GmbH > Firmensitz: Trippstadter Strasse 122, D-67663 Kaiserslautern > Geschaeftsfuehrung: > Prof. Dr. Dr. h.c. mult. Wolfgang Wahlster (Vorsitzender) > Dr. Walter Olthoff > Vorsitzender des Aufsichtsrats: Prof. Dr. h.c. Hans A. Aukes > Amtsgericht Kaiserslautern, HRB 2313 > > > > ------------------------------------------------------------------------------ > Centralized Desktop Delivery: Dell and VMware Reference Architecture > Simplifying enterprise desktop deployment and management using > Dell EqualLogic storage and VMware View: A highly scalable, end-to-end > client virtualization framework. Read more! > http://p.sf.net/sfu/dell-eql-dev2dev > _______________________________________________ > cairo-user mailing list > cai...@li... > https://lists.sourceforge.net/lists/listinfo/cairo-user > > |
From: Christian S. <chs...@df...> - 2010-11-15 16:45:45
|
Hi Spencer thanks for your reply, unfortunately your guess did not help. The error message is still the same. I was trying all demos including the speech synth client. There the error message is slightly different maybe it gives you a hint for what reason the demos are not working out. Btw did you check it out if the demos for download are running on your machine? 2010-11-15 17:27:22,158 INFO {main} org.speechforge.cairo.demo.tts.SpeechSynthClient Sending a SIP invitation to the cairo server. 2010-11-15 17:27:22,767 WARN {EventScannerThread} org.speechforge.cairo.sip.SipListenerImpl Received an unhandled SIP response status code (ignoring it): 183 : Session progress 2010-11-15 17:27:22,839 INFO {main} org.speechforge.cairo.demo.tts.SpeechSynthClient Received the SIP Response. 2010-11-15 17:27:36,408 INFO {Thread-1} org.speechforge.cairo.sip.SimpleSipAgent Sent a SIP BYE. C:\Users\chschulz\sources\nlp\cairo\cairo-0.3\demo\bin> Best, Chris Am 15.11.2010 17:06, schrieb spencer lord: > Hi Chris, > > It looks like it may be a problem with jmf. Cairo uses JMF for RTP > streaming. Did you install it on the machine you are using? The jmf > download link on Cairo installation web page is stale. Sorry, we need > to fix that. You can download it here > > http://www.oracle.com/technetwork/java/javase/download-142937.html > > If it is already installed, make sue that jmf.jar nd sound.jar are in > the lib.ext directory of the jre that you are using. > > Let me know if that solves the problem. > > Spencer > > On Mon, Nov 15, 2010 at 6:40 AM, Christian Schulz <chs...@df... > <mailto:chs...@df...>> wrote: > > Hi all, > > I am new to the cairo project and I was trying out the demos you can > download here: > http://sourceforge.net/projects/cairo/files/ > > Following the instructions here > http://www.speechforge.org/projects/cairo/intro.html (that is > identical > to the instruction in the readme file included in the download) I > however cannot get the demo applications run. In particular the > recognizer will output this error, while trying to connect to the > receiver: > > Sending a SIP invitation to the cairo server. > 2010-11-15 13:05:07,296 WARN {EventScannerThread} > org.speechforge.cairo.sip.SipListenerImpl > Received an unhandled SIP response status code (ignoring it): 183 : > Session progress > 2010-11-15 13:05:07,861 INFO {main} > org.speechforge.cairo.demo.recog.RecognitionClient > Received the SIP Response. > java.lang.NullPointerException > 2010-11-15 13:05:08,397 WARN {Thread-7} > org.speechforge.cairo.rtp.RTPPlayer playSource(): encountered > unexpected > exception: > javax.media.NoDataSourceException: Error instantiating class: > com.sun.media.protocol.dsound.DataSource : > java.lang.NullPointerException > at javax.media.Manager.createDataSource(Manager.java:1012) > at > org.speechforge.cairo.rtp.RTPPlayer.playSource(RTPPlayer.java:110) > at > org.speechforge.cairo.rtp.NativeMediaClient$TransmitThread.run(NativeMediaClient.java:105) > Exception in thread "Thread-7" java.lang.RuntimeException: > playSource() > encountered unexpected exception > at > org.speechforge.cairo.rtp.RTPPlayer.playSource(RTPPlayer.java:153) > at > org.speechforge.cairo.rtp.NativeMediaClient$TransmitThread.run(NativeMediaClient.java:105) > Caused by: javax.media.NoDataSourceException: Error instantiating > class: > com.sun.media.protocol.dsound.DataSource : > java.lang.NullPointerException > at javax.media.Manager.createDataSource(Manager.java:1012) > at > org.speechforge.cairo.rtp.RTPPlayer.playSource(RTPPlayer.java:110) > ... 1 more > 2010-11-15 13:05:18,577 WARN {main} > org.speechforge.cairo.demo.recog.RecognitionClient > org.mrcp4j.client.MrcpInvocationException: MRCP response contains an > error code, the request invocation did not complete successfully. > org.mrcp4j.client.MrcpInvocationException: MRCP response contains an > error code, the request invocation did not complete successfully. > at > org.mrcp4j.client.MrcpChannel.sendRequest(MrcpChannel.java:143) > at > org.speechforge.cairo.demo.recog.RecognitionClient.doRecognize(RecognitionClient.java:164) > at > org.speechforge.cairo.demo.recog.RecognitionClient.main(RecognitionCl > ient.java:358) > 2010-11-15 13:05:27,929 INFO {main} > org.speechforge.cairo.sip.SimpleSipAgent > Sent a SIP BYE. > > Thanks for your help! > > chris > > ------------------------------------------------------------------------------ > Centralized Desktop Delivery: Dell and VMware Reference Architecture > Simplifying enterprise desktop deployment and management using > Dell EqualLogic storage and VMware View: A highly scalable, end-to-end > client virtualization framework. Read more! > http://p.sf.net/sfu/dell-eql-dev2dev > _______________________________________________ > cairo-user mailing list > cai...@li... > <mailto:cai...@li...> > https://lists.sourceforge.net/lists/listinfo/cairo-user > > > > ------------------------------------------------------------------------------ > Centralized Desktop Delivery: Dell and VMware Reference Architecture > Simplifying enterprise desktop deployment and management using > Dell EqualLogic storage and VMware View: A highly scalable, end-to-end > client virtualization framework. Read more! > http://p.sf.net/sfu/dell-eql-dev2dev > > > _______________________________________________ > cairo-user mailing list > cai...@li... > https://lists.sourceforge.net/lists/listinfo/cairo-user -- Christian H. Schulz DFKI GmbH, Campus D3 2 Stuhlsatzenhausweg 3 D-66123 Saarbrücken, Germany +49 (0)681 85775-5371 (tel.) -5021 (fax) mail: chs...@df..., http: www.dfki.de/~chschulz ------------------------------------------------------------------ Deutsches Forschungszentrum fuer Kuenstliche Intelligenz GmbH Firmensitz: Trippstadter Strasse 122, D-67663 Kaiserslautern Geschaeftsfuehrung: Prof. Dr. Dr. h.c. mult. Wolfgang Wahlster (Vorsitzender) Dr. Walter Olthoff Vorsitzender des Aufsichtsrats: Prof. Dr. h.c. Hans A. Aukes Amtsgericht Kaiserslautern, HRB 2313 |
From: spencer l. <spe...@gm...> - 2010-11-15 16:06:17
|
Hi Chris, It looks like it may be a problem with jmf. Cairo uses JMF for RTP streaming. Did you install it on the machine you are using? The jmf download link on Cairo installation web page is stale. Sorry, we need to fix that. You can download it here http://www.oracle.com/technetwork/java/javase/download-142937.html If it is already installed, make sue that jmf.jar nd sound.jar are in the lib.ext directory of the jre that you are using. Let me know if that solves the problem. Spencer On Mon, Nov 15, 2010 at 6:40 AM, Christian Schulz <chs...@df...> wrote: > Hi all, > > I am new to the cairo project and I was trying out the demos you can > download here: > http://sourceforge.net/projects/cairo/files/ > > Following the instructions here > http://www.speechforge.org/projects/cairo/intro.html (that is identical > to the instruction in the readme file included in the download) I > however cannot get the demo applications run. In particular the > recognizer will output this error, while trying to connect to the receiver: > > Sending a SIP invitation to the cairo server. > 2010-11-15 13:05:07,296 WARN {EventScannerThread} > org.speechforge.cairo.sip.SipListenerImpl > Received an unhandled SIP response status code (ignoring it): 183 : > Session progress > 2010-11-15 13:05:07,861 INFO {main} > org.speechforge.cairo.demo.recog.RecognitionClient > Received the SIP Response. > java.lang.NullPointerException > 2010-11-15 13:05:08,397 WARN {Thread-7} > org.speechforge.cairo.rtp.RTPPlayer playSource(): encountered unexpected > exception: > javax.media.NoDataSourceException: Error instantiating class: > com.sun.media.protocol.dsound.DataSource : java.lang.NullPointerException > at javax.media.Manager.createDataSource(Manager.java:1012) > at > org.speechforge.cairo.rtp.RTPPlayer.playSource(RTPPlayer.java:110) > at > > org.speechforge.cairo.rtp.NativeMediaClient$TransmitThread.run(NativeMediaClient.java:105) > Exception in thread "Thread-7" java.lang.RuntimeException: playSource() > encountered unexpected exception > at > org.speechforge.cairo.rtp.RTPPlayer.playSource(RTPPlayer.java:153) > at > > org.speechforge.cairo.rtp.NativeMediaClient$TransmitThread.run(NativeMediaClient.java:105) > Caused by: javax.media.NoDataSourceException: Error instantiating class: > com.sun.media.protocol.dsound.DataSource : java.lang.NullPointerException > at javax.media.Manager.createDataSource(Manager.java:1012) > at > org.speechforge.cairo.rtp.RTPPlayer.playSource(RTPPlayer.java:110) > ... 1 more > 2010-11-15 13:05:18,577 WARN {main} > org.speechforge.cairo.demo.recog.RecognitionClient > org.mrcp4j.client.MrcpInvocationException: MRCP response contains an > error code, the request invocation did not complete successfully. > org.mrcp4j.client.MrcpInvocationException: MRCP response contains an > error code, the request invocation did not complete successfully. > at org.mrcp4j.client.MrcpChannel.sendRequest(MrcpChannel.java:143) > at > > org.speechforge.cairo.demo.recog.RecognitionClient.doRecognize(RecognitionClient.java:164) > at > org.speechforge.cairo.demo.recog.RecognitionClient.main(RecognitionCl > ient.java:358) > 2010-11-15 13:05:27,929 INFO {main} > org.speechforge.cairo.sip.SimpleSipAgent > Sent a SIP BYE. > > Thanks for your help! > > chris > > > ------------------------------------------------------------------------------ > Centralized Desktop Delivery: Dell and VMware Reference Architecture > Simplifying enterprise desktop deployment and management using > Dell EqualLogic storage and VMware View: A highly scalable, end-to-end > client virtualization framework. Read more! > http://p.sf.net/sfu/dell-eql-dev2dev > _______________________________________________ > cairo-user mailing list > cai...@li... > https://lists.sourceforge.net/lists/listinfo/cairo-user > |
From: Christian S. <chs...@df...> - 2010-11-15 14:41:04
|
Hi all, I am new to the cairo project and I was trying out the demos you can download here: http://sourceforge.net/projects/cairo/files/ Following the instructions here http://www.speechforge.org/projects/cairo/intro.html (that is identical to the instruction in the readme file included in the download) I however cannot get the demo applications run. In particular the recognizer will output this error, while trying to connect to the receiver: Sending a SIP invitation to the cairo server. 2010-11-15 13:05:07,296 WARN {EventScannerThread} org.speechforge.cairo.sip.SipListenerImpl Received an unhandled SIP response status code (ignoring it): 183 : Session progress 2010-11-15 13:05:07,861 INFO {main} org.speechforge.cairo.demo.recog.RecognitionClient Received the SIP Response. java.lang.NullPointerException 2010-11-15 13:05:08,397 WARN {Thread-7} org.speechforge.cairo.rtp.RTPPlayer playSource(): encountered unexpected exception: javax.media.NoDataSourceException: Error instantiating class: com.sun.media.protocol.dsound.DataSource : java.lang.NullPointerException at javax.media.Manager.createDataSource(Manager.java:1012) at org.speechforge.cairo.rtp.RTPPlayer.playSource(RTPPlayer.java:110) at org.speechforge.cairo.rtp.NativeMediaClient$TransmitThread.run(NativeMediaClient.java:105) Exception in thread "Thread-7" java.lang.RuntimeException: playSource() encountered unexpected exception at org.speechforge.cairo.rtp.RTPPlayer.playSource(RTPPlayer.java:153) at org.speechforge.cairo.rtp.NativeMediaClient$TransmitThread.run(NativeMediaClient.java:105) Caused by: javax.media.NoDataSourceException: Error instantiating class: com.sun.media.protocol.dsound.DataSource : java.lang.NullPointerException at javax.media.Manager.createDataSource(Manager.java:1012) at org.speechforge.cairo.rtp.RTPPlayer.playSource(RTPPlayer.java:110) ... 1 more 2010-11-15 13:05:18,577 WARN {main} org.speechforge.cairo.demo.recog.RecognitionClient org.mrcp4j.client.MrcpInvocationException: MRCP response contains an error code, the request invocation did not complete successfully. org.mrcp4j.client.MrcpInvocationException: MRCP response contains an error code, the request invocation did not complete successfully. at org.mrcp4j.client.MrcpChannel.sendRequest(MrcpChannel.java:143) at org.speechforge.cairo.demo.recog.RecognitionClient.doRecognize(RecognitionClient.java:164) at org.speechforge.cairo.demo.recog.RecognitionClient.main(RecognitionCl ient.java:358) 2010-11-15 13:05:27,929 INFO {main} org.speechforge.cairo.sip.SimpleSipAgent Sent a SIP BYE. Thanks for your help! chris |
From: spencer l. <spe...@gm...> - 2010-02-08 18:48:30
|
Hi, Sorry for mixing up the emails and missing this one. I can see the configuration and the docs explaining it need to be improved and simplified. I think you are close. Let me try to explain the steps you should take . To start, you will need to run 4 scripts. (later we can talk about consolidating it into a single script) 1. Run Cairo (the MRCPv2 Server) SCRIPT1: rserver.sh (run this first) SCRIPT2: transmitter1.sh SCRIPT3: receiver1.sh (this script uses cairoconfig.xml, which is where you can specify your own sphinx-config.xml file) 2. Run Zanzibar in pbx mode SCRIPT4: asteriskconnector.sh (this script uses pbxconfig.xml) Modify pbxConfig Change cairoSipHostName to the ip address of the machine running cairo mysipaddress does not need to be changed. In the dialplan change x-application:basic|org.speechforge.apps.demos.Parrot to x-application:beanid|Parrot to Twinkle Looks ok. I have had some trouble using localhost and 127.0.0.1 in my sip addresses. If 1001at127.0.0.1 does not work try switching to the ip address of the ip address of the machine running asterisk. I hope this helps, Spencer On Tue, Jan 19, 2010 at 3:29 AM, johny jj2 <joh...@gm...> wrote: > Thank you for your answer! > > > > 1. There are three config files with the same things, e.g. all > withcairo.xml, democonfig.xml and pbxconfig.xml contain > <value>192.168.0.103</value>. I decided to make changes only in > pbxconfig.xml. (I also had to copy the whole config directory as > explained in one of previous mails, because one of scripts couldn't be > executed). > > > > 2. I changed in pbxconfig.xml mySipAddress from > sip:cai...@sp... <sip%3Ac...@sp...> to > sip:mainaccount@localhost (this is > what I see in my linux terminal) and IP from 192.168.0.100 to > 192.168.1.101 (which is shown by ifconfig for my wifi card). What > should be the value of mySipAddress and cairoSipAddress? > > > > I call from Twinkle to 1001at127.0.0.1 (of course instead of 'at' I > write the symbol @). > > > > Asterisk in verbose mode shows: > > > > -- Executing [1001@default:1] > SIPAddHeader("SIP/localhost-00000000", > "x-channel:SIP/localhost-00000000") in new stack > > -- Executing [1001@default:2] > SIPAddHeader("SIP/localhost-00000000", > "x-application:basic|org.speechforge.apps.demos.Parrot") in new stack > > -- Executing [1001@default:3] Dial("SIP/localhost-00000000", > "SIP/Zanzibar") in new stack > > -- Called Zanzibar > > -- SIP/Zanzibar-00000001 is ringing > > localhost*CLI> > > == Spawn extension (default, 1001, 3) exited non-zero on > 'SIP/localhost-00000000' > > localhost*CLI> > > > > And fourth script (asteriskConnector.sh): > > > > ***: 127.0.0.1 > > Connecting to 192.168.0.103:5038 > > Got an invite request > > Got a dialog terminated even > > > > In this case Twinkle rings all the time and cannot connect. > > > > 3. Instead of 192.168.1.101 I write 127.0.0.1. > > > > Asterisk in verbose mode: > > > > localhost*CLI> > > -- Executing [1001@default:1] > SIPAddHeader("SIP/localhost-00000004", > "x-channel:SIP/localhost-00000004") in new stack > > -- Executing [1001@default:2] > SIPAddHeader("SIP/localhost-00000004", > "x-application:basic|org.speechforge.apps.demos.Parrot") in new stack > > -- Executing [1001@default:3] Dial("SIP/localhost-00000004", > "SIP/Zanzibar") in new stack > > -- Called Zanzibar > > -- SIP/Zanzibar-00000005 is ringing > > -- SIP/Zanzibar-00000005 answered SIP/localhost-00000004 > > localhost*CLI> > > == Spawn extension (default, 1001, 3) exited non-zero on > 'SIP/localhost-00000004' > > > > Fourth script (asteriskConnector.sh): > > > > *: 127.0.0.1 > > Connecting to 192.168.0.103:5038 > > Got an invite request > > Received an unhandled SIP response status code (ignoring it): 183 : > Session progress > > java.lang.Exception: Application Type basic not supported > I apologize for this bug/problem with docs. I recommend using the beanid approach (see above) and configuring the bean with Spring in pbxconfig. , but if you want to specify the classname, use the keyword "className" rather than "basic" (case sensitive, another bug). Make sure you have a mrcp server (cairo) running too > > > > In this case Asterisk receives the call from Twinkle immediately but > then nothing happens. The session is established, the timer in Twinkle > shows how much time ago but nothing else happens. > > > > --- > > > > Ports are correctly set. For Twinkle, SIP listens on 5061, for > Asterisk 5060, openIVR 5090 and Cairo 5050. (Asterisk is informed in > sip.conf that Zanzibar is on port 5090). I checked "netstat -l -p" > after running all four scripts and Asterisk. It looks like Asterisk > listens on all IP numbers (*) and port 5060 and Java listens only on > localhost (typically 127.0.0.1). > > > > In extensions.conf in [demo] section I've got: > > exten => 1001,1,SIPAddHeader(x-channel:${CHANNEL}) > > exten => > 1001,n,SIPAddHeader(x-application:basic|org.speechforge.apps.demos.Parrot) > > exten => 1001,n,Dial(SIP/Zanzibar) > > The exten which you specified in docs didn't work because of ',n' in > first line instead of ',1'. > > > ====================================== > > Summing up: I'm worried about this "java.lang.Exception: Application > Type basic not supported" because it looks like Zanzibar/Cairo doesn't > understand type which is dedicated for it. I'm also not sure what to > do with 192.168.0.103:5038. And I don't quite get why there are the > same things in three config files. What should I set for mySipAddress > and cairoSipAddress in pbxconfig.xml? > ====================================== > > > > --- > > > > To answer your earlier question about getting results with plain old java > -- without tags and grammars. You dont need to use tags -- you could > analyze the raw string in java when you get the results. But at least with > sphinx4, you need to either use a grammar or a language model to define > what can be said. > > > > I am thinking about adding a simple grammar feature, where you just > supply a list of words. Would that be helpful? > > > > I think it would. I just would like to have everything in java file > without grammar and vxml, in similar way as it is in HelloDigits in > Sphinx4, where the grammar is the simplest possible. (So that I would > have value of string type in java code. Every time the code would be > checking if this string is empty or not. If it is not, I would enter > the given loop which would check what word it is. According to what > word it is, it would be able to follow some kind of action, like > calculating the sum or saving results to text file. Then the string > would be empty again and I would be waiting for the other time when > something is recognized and the string in nonempty again. This > approach wouldn't require any vxml and it requires very simple grammar > with only list of words). So that's the question about how to analyze > those raw results. And about language model, I needed to create it > with lmtoolkit available on-line and include it in acoustic model but > it is used only for big dictionaries I guess. > At present you will need to specify a simple jsgf grammar. Evn HelloDigits uses a simple jsgf grammar digits.gram I will add a word list grammar to the enhancement list for cairo. > > > I also thought about the way of creating the application and > recompiling the whole Cairo/Zanzibar. I need to create my app and add > my acoustic model. (I hope there wouldn't be any problem with loaders > of acoustic model, perhaps I need to use loaders of Sphinx3 for my > model). I think the only what I need is to make changes to content of > zanzibar-0.1-bin.src.bz2 and I don't need content of those other > archives like rtp-0.2 and client-0.2. (Again, there are the same files > in src/voicexml and demo/voicexml, which do I need? I think creating > my application would be just creating MyApp.java in > src/java/org/speechforge/apps/demos, creating MyApp.gram in > src/resources/grammar, adding my wav files to src/resources/prompts, > editing sphinx-config.xml in src/resources/config to replace WSJ with > my acoustic model, adding the same grammar to src/voicexml, adding > empty [can it be empty file? if not, what should be here?] MyApp.vxml > to src/voicexml if it is possible not to use any vxml. But if I don't > use vxml, how to use wav files from /src/resources/prompts in > MyApp.java from /src/java/org/speechforge/apps/demos? I mean what code > would be responsible for that? Do I miss anything in the process of > developing the application?). Then I would need to recompile it > because I've got new apps and other acoustic model. In Sphinx4 the > recompiling is very simple task, it is simply running 'ant' in its > main directory, which recompiles only those files which need to be > recompiled due to some changes. Is there similar simple way to do it > with Zanzibar? > You should not have to recompile or modify zanziabr or cairo. 1. Use cairo-config to specify own your sphinx-config.xml file (with your own acoustic models) 2. create your own jar file with your own app and add to the lib directory of zanzibar 3. Configure your app using spring (look at examples in pbxconfig, notice the prompts and grammars are specified here. They need not be in prompts or grammars directory. They can be anywhere you specify.) One issue is that we have bundled an older version of sphinx4 in cairo. We should do a new release with the latest version with the newer acoustic model jar code) > > > But first and most important thing is about running this parrot > example as explained earlier in this mail. > > > > Thanks again! > > Regards! > > > ------------------------------------------------------------------------------ > Throughout its 18-year history, RSA Conference consistently attracts the > world's best and brightest in the field, creating opportunities for > Conference > attendees to learn about information security's most important issues > through > interactions with peers, luminaries and emerging and established companies. > http://p.sf.net/sfu/rsaconf-dev2dev > _______________________________________________ > cairo-user mailing list > cai...@li... > https://lists.sourceforge.net/lists/listinfo/cairo-user > |
From: johny j. <joh...@gm...> - 2010-02-08 01:14:17
|
Hello! May I wish that you would answer to my post from Jan 19? Previously you answered twice to one of my old posts, not to the newest post. You can check it here: http://old.nabble.com/using-other-acoustic-models-in-Cairo-Zanzibar-to26879547.html Thanks again! 2010/1/19 johny jj2 <joh...@gm...>: > Thank you for your answer! > > > > 1. There are three config files with the same things, e.g. all > withcairo.xml, democonfig.xml and pbxconfig.xml contain > <value>192.168.0.103</value>. I decided to make changes only in > pbxconfig.xml. (I also had to copy the whole config directory as > explained in one of previous mails, because one of scripts couldn't be > executed). > > > > 2. I changed in pbxconfig.xml mySipAddress from > sip:cai...@sp... to sip:mainaccount@localhost (this is > what I see in my linux terminal) and IP from 192.168.0.100 to > 192.168.1.101 (which is shown by ifconfig for my wifi card). What > should be the value of mySipAddress and cairoSipAddress? > > > > I call from Twinkle to 1001at127.0.0.1 (of course instead of 'at' I > write the symbol @). > > > > Asterisk in verbose mode shows: > > > > -- Executing [1001@default:1] > SIPAddHeader("SIP/localhost-00000000", > "x-channel:SIP/localhost-00000000") in new stack > > -- Executing [1001@default:2] > SIPAddHeader("SIP/localhost-00000000", > "x-application:basic|org.speechforge.apps.demos.Parrot") in new stack > > -- Executing [1001@default:3] Dial("SIP/localhost-00000000", > "SIP/Zanzibar") in new stack > > -- Called Zanzibar > > -- SIP/Zanzibar-00000001 is ringing > > localhost*CLI> > > == Spawn extension (default, 1001, 3) exited non-zero on > 'SIP/localhost-00000000' > > localhost*CLI> > > > > And fourth script (asteriskConnector.sh): > > > > ***: 127.0.0.1 > > Connecting to 192.168.0.103:5038 > > Got an invite request > > Got a dialog terminated even > > > > In this case Twinkle rings all the time and cannot connect. > > > > 3. Instead of 192.168.1.101 I write 127.0.0.1. > > > > Asterisk in verbose mode: > > > > localhost*CLI> > > -- Executing [1001@default:1] > SIPAddHeader("SIP/localhost-00000004", > "x-channel:SIP/localhost-00000004") in new stack > > -- Executing [1001@default:2] > SIPAddHeader("SIP/localhost-00000004", > "x-application:basic|org.speechforge.apps.demos.Parrot") in new stack > > -- Executing [1001@default:3] Dial("SIP/localhost-00000004", > "SIP/Zanzibar") in new stack > > -- Called Zanzibar > > -- SIP/Zanzibar-00000005 is ringing > > -- SIP/Zanzibar-00000005 answered SIP/localhost-00000004 > > localhost*CLI> > > == Spawn extension (default, 1001, 3) exited non-zero on > 'SIP/localhost-00000004' > > > > Fourth script (asteriskConnector.sh): > > > > *: 127.0.0.1 > > Connecting to 192.168.0.103:5038 > > Got an invite request > > Received an unhandled SIP response status code (ignoring it): 183 : > Session progress > > java.lang.Exception: Application Type basic not supported > > > > In this case Asterisk receives the call from Twinkle immediately but > then nothing happens. The session is established, the timer in Twinkle > shows how much time ago but nothing else happens. > > > > --- > > > > Ports are correctly set. For Twinkle, SIP listens on 5061, for > Asterisk 5060, openIVR 5090 and Cairo 5050. (Asterisk is informed in > sip.conf that Zanzibar is on port 5090). I checked "netstat -l -p" > after running all four scripts and Asterisk. It looks like Asterisk > listens on all IP numbers (*) and port 5060 and Java listens only on > localhost (typically 127.0.0.1). > > > > In extensions.conf in [demo] section I've got: > > exten => 1001,1,SIPAddHeader(x-channel:${CHANNEL}) > > exten => 1001,n,SIPAddHeader(x-application:basic|org.speechforge.apps.demos.Parrot) > > exten => 1001,n,Dial(SIP/Zanzibar) > > The exten which you specified in docs didn't work because of ',n' in > first line instead of ',1'. > > > ====================================== > > Summing up: I'm worried about this "java.lang.Exception: Application > Type basic not supported" because it looks like Zanzibar/Cairo doesn't > understand type which is dedicated for it. I'm also not sure what to > do with 192.168.0.103:5038. And I don't quite get why there are the > same things in three config files. What should I set for mySipAddress > and cairoSipAddress in pbxconfig.xml? > ====================================== > > > > --- > > >> To answer your earlier question about getting results with plain old java -- without tags and grammars. You dont need to use tags -- you could analyze the raw string in java when you get the results. But at least with sphinx4, you need to either use a grammar or a language model to define what can be said. > > >> I am thinking about adding a simple grammar feature, where you just supply a list of words. Would that be helpful? > > > > I think it would. I just would like to have everything in java file > without grammar and vxml, in similar way as it is in HelloDigits in > Sphinx4, where the grammar is the simplest possible. (So that I would > have value of string type in java code. Every time the code would be > checking if this string is empty or not. If it is not, I would enter > the given loop which would check what word it is. According to what > word it is, it would be able to follow some kind of action, like > calculating the sum or saving results to text file. Then the string > would be empty again and I would be waiting for the other time when > something is recognized and the string in nonempty again. This > approach wouldn't require any vxml and it requires very simple grammar > with only list of words). So that's the question about how to analyze > those raw results. And about language model, I needed to create it > with lmtoolkit available on-line and include it in acoustic model but > it is used only for big dictionaries I guess. > > > > I also thought about the way of creating the application and > recompiling the whole Cairo/Zanzibar. I need to create my app and add > my acoustic model. (I hope there wouldn't be any problem with loaders > of acoustic model, perhaps I need to use loaders of Sphinx3 for my > model). I think the only what I need is to make changes to content of > zanzibar-0.1-bin.src.bz2 and I don't need content of those other > archives like rtp-0.2 and client-0.2. (Again, there are the same files > in src/voicexml and demo/voicexml, which do I need? I think creating > my application would be just creating MyApp.java in > src/java/org/speechforge/apps/demos, creating MyApp.gram in > src/resources/grammar, adding my wav files to src/resources/prompts, > editing sphinx-config.xml in src/resources/config to replace WSJ with > my acoustic model, adding the same grammar to src/voicexml, adding > empty [can it be empty file? if not, what should be here?] MyApp.vxml > to src/voicexml if it is possible not to use any vxml. But if I don't > use vxml, how to use wav files from /src/resources/prompts in > MyApp.java from /src/java/org/speechforge/apps/demos? I mean what code > would be responsible for that? Do I miss anything in the process of > developing the application?). Then I would need to recompile it > because I've got new apps and other acoustic model. In Sphinx4 the > recompiling is very simple task, it is simply running 'ant' in its > main directory, which recompiles only those files which need to be > recompiled due to some changes. Is there similar simple way to do it > with Zanzibar? > > > > But first and most important thing is about running this parrot > example as explained earlier in this mail. > > > > Thanks again! > > Regards! > |
From: johny j. <joh...@gm...> - 2010-01-23 22:18:33
|
Thanks for your answer! However - it looks like you answered twice to my previous post. Look here: http://old.nabble.com/using-other-acoustic-models-in-Cairo-Zanzibar-to26879547.html . There are two answers to my post from Jan 09 (they are Jan 22 and Jan 10) but there is no answer to my newest post (Jan 19). Regards! |
From: spencer l. <spe...@gm...> - 2010-01-22 20:15:59
|
Hi On Sat, Jan 9, 2010 at 1:37 PM, johny jj2 <joh...@gm...> wrote: > Thank you for your answer! > > I've got two major difficulties, one with adding action to recognizing > words which match the grammar and the other with running Parrot from > softphone. Please, answer at least first of these two. > > ---------------------------------------------------- > > May you tell me, please, how to add action to Parrot if the {WEATHER} > is recognized? I'd like the application to respond with wav file and > to follow some java code (e.g. write to file; in my case it would be > calculating sum and playing either 1.wav or 2.wav). > > I guess there must be change to both Parrot.java and parrot.vxml. > > For example to parrot.vxml (perhaps it is not good way) - to answer > the action with wav file: > > <filled> > > <prompt> <value expr="main"/> </prompt> > <if cond="main=='weather'"> //NEW CODE BEGINS HERE > <block> > <audio src="weather_wav_file_URL"/> > </block> > </if> //NEW CODE ENDS HERE > > <if cond="main=='quit'"> > > <exit/> > > <else/> > > <clear namelist="main"/> > > <reprompt/> > > </if> > > > > </filled> > > This looks basically correct, but I would use an "else if" in the existing "if" block rather than a new "if" block. I am not an expert in voicexml, so take my answer with a grain of salt :) > And Parrot.java - to add some action with java code, e.g. writing to file: > > if ((rule.getTag().equals("QUIT")) && (rule.getRule().equals("main"))) > { > > //... > > } > if ((rule.getTag().equals("WEATHER")) && > (rule.getRule().equals("main"))) { > > //here e.g. I perform calculations - according to their > result I'd like the application to say either "control sum correct" or > "control sum incorrect" - wav answer is DEPENDENT ON CALCULATIONS, not > only on recognized word (which is the case of using vxml file)!!! how > to specify playing either 1.wav or 2.wav here ?!?!? > > } > This looks good. I would consider using "else if" also note the first parameter to playAndrecognize is TRUE indicating that the next parameter is a uri to an audio file. If it is false the next parameter is assumed to be text to be synthesized. Then in the else if block, set the audioUri to a the audio file of your choice. String audioUri; boolean PromptIsUri = true; RecognitionResult r = sClient.playAndRecognizeBlocking(PromptIsUri ,audioUri,grammar, false); if ((r != null) && (!r.isOutOfGrammar())) { for (RuleMatch rule : r.getRuleMatches()) { if ((rule.getTag().equals("QUIT")) && (rule.getRule().equals("main"))) { stopFlag = true; this.getContext().dialogCompleted(); else if ((rule.getTag().equals("WEATHER")) && (rule.getRule().equals("main"))) { //logic to figur out which audio file to use in the the next playAndRec command audioUri = file:///prompts/a.wav } } > Some additional, minor, questions: > > 1. I don't get why in parrot.vxml there is "Would you like to hear the > weather, get sports news or hear a stock quote?" if the parrot > application is about repeating what somebody says. > It is not recognizing anything. It only recognizes what is in the grammar. The prompt is trying to indicate what is in the grammar. It is a dumb parrot that only knows a few words :) > 2. Why is there > ~/cairo/zanzibar-0.1-src/src/resources/prompts/parrot.wav if this wav > is never used? > Sorry, I meant to add an option to the demo to show how you can use a pre-recorded prompt rather than TTS. I will add it to the next release. > > ---------------------------------------------------- > > May you also help me to run this Parrot from softphone, please? > > 1. I extracted zanzibar-0.1-bin.tar.bz2 to > /home/mainaccount/cairo/zanzibar-0.1-bin. Later I did all the stuff > connected with global variables for Java, extracting JSAPI etc. > 2. I run /home/mainaccount/cairo/zanzibar-0.1-bin/bin/rserver.sh from > Terminal. (There was error in sh file - I had to change from \ to / > because it couldn't find launch.sh). > 3. I copy config directory from ~/cairo/zanzibar-0.1-bin to > ~/cairo/zanzibar-0.1-bin/bin because one of those three sh couldn't > find config directory. > 4. OK, all three sh files from ~/cairo/zanzibar-0.1-bin/bin/cairo and > also ~/cairo/zanzibar-0.1-bin/bin/zanzibar/asteriskConnector.sh are > running. > 5. I add to sip.conf: > > [Zanzibar] > type=peer > host=localhost //I wasn't sure what to write here > port=5090 > dtmfmode=info > canreinvite=no > localhost is probably ok if zanzibar and asterisk are on the same machine. You can put the real host name or ip address of the zanzibar machine to be safe. > > 6. I add to extensions.conf at the end of [mainmenu] section (is it > proper place?): > > exten => 1,n,SIPAddHeader(x-channel:${CHANNEL}) > exten => > 1,n,SIPAddHeader(x-application:basic|org.speechforge.apps.demos.Parrot) > exten => 1,n,Dial(SIP/Zanzibar) > Not sure if it should be at the end of mainmenu, can you send the main menu dialplan? I am very sorry, but the doc has a mistake and there is a bug in the code. This section is partially correct type|applicationName where: type is either "beanId" "className" or "vxml" So if you want to use the classname option instead of basic, use "classname" The bug is that you must use "classname" in all lower case. exten => 1,n,SIPAddHeader(x-application:classname|org.speechforge.apps.demos.Parrot) if you still have the bean "Parrot" defined in your context.xml file, you can also use this instead (again case is important) exten => 1,n,SIPAddHeader(x-application:beanId|Parrot) <bean id="Parrot" class="org.speechforge.apps.demos.Parrot" singleton="false"> <property name="prompt"> <value>You can start speaking any time. Would you like to hear the weather, get sports news or hear a stock quote? Say goodbye to exit.</value> </property> <property name="grammar"> <value>file:../../demo/grammar/example-loop.gram</value> </property> </bean> > 7. I add to /etc/asterisk/manager.conf at the end of file: > > [twinkle] > secret=password > permit=0.0.0.0/0.0.0.0 > read=system,call,log,verbose,agent,command,user > write=system,call,log,verbose,agent,command,user<http://0.0.0.0/0.0.0.0%0Aread=system,call,log,verbose,agent,command,user%0Awrite=system,call,log,verbose,agent,command,user> > This is not really used unless you are transfering calls. I do not believe you are doing this, so no worries about this config. > > 8. I cannot find context.xml file to do what is written here > http://www.spokentech.org/openivr/aik.html > Sorry, but another issue with the docs. This is also called the configuration file. It is basically a spring config file that is passed in on te command line to the main zanzibar program. So look at the shell script that you are using to start zanzibar (probably asteriskConnector.sh) and check this line. In this case the context or config file is pbxconfig.xml CONFIG=file:../config/pbxconfig.xml > > 9. I run 'asterisk' in Terminal > > 10. I run 'twinkle'. I choos profile: 'twinkle', SIP service provider: > 'None (direct IP to IP calls), username: twinkle, domain: localhost, > system settings -> network -> SIP port: 5061. I write in call: > '1000@127.0.0.1'. It connects me to "Congratulations. You have > successfully installed and executed the Asterisk open source". So I > can connect to Asterisk but cannot connect to Parrot demo. I guess the > call is improper but there also may be something wrong with > configurations above. > I think you are very close. The issue is with your dialplan. Probably your placement of the three statements you describe in section 6. above. > > Regards! > > PS > > Are your acoustic models in jar files? > Yes, it contains two directories (etc and model_parameters) without > loaders (it needs sphinx3 loader to be specified in xml). > > > ------------------------------------------------------------------------------ > This SF.Net email is sponsored by the Verizon Developer Community > Take advantage of Verizon's best-in-class app development support > A streamlined, 14 day to market process makes app distribution fast and > easy > Join now and get one step closer to millions of Verizon customers > http://p.sf.net/sfu/verizon-dev2dev > _______________________________________________ > cairo-user mailing list > cai...@li... > https://lists.sourceforge.net/lists/listinfo/cairo-user > |
From: johny j. <joh...@gm...> - 2010-01-19 11:29:45
|
Thank you for your answer! 1. There are three config files with the same things, e.g. all withcairo.xml, democonfig.xml and pbxconfig.xml contain <value>192.168.0.103</value>. I decided to make changes only in pbxconfig.xml. (I also had to copy the whole config directory as explained in one of previous mails, because one of scripts couldn't be executed). 2. I changed in pbxconfig.xml mySipAddress from sip:cai...@sp... to sip:mainaccount@localhost (this is what I see in my linux terminal) and IP from 192.168.0.100 to 192.168.1.101 (which is shown by ifconfig for my wifi card). What should be the value of mySipAddress and cairoSipAddress? I call from Twinkle to 1001at127.0.0.1 (of course instead of 'at' I write the symbol @). Asterisk in verbose mode shows: -- Executing [1001@default:1] SIPAddHeader("SIP/localhost-00000000", "x-channel:SIP/localhost-00000000") in new stack -- Executing [1001@default:2] SIPAddHeader("SIP/localhost-00000000", "x-application:basic|org.speechforge.apps.demos.Parrot") in new stack -- Executing [1001@default:3] Dial("SIP/localhost-00000000", "SIP/Zanzibar") in new stack -- Called Zanzibar -- SIP/Zanzibar-00000001 is ringing localhost*CLI> == Spawn extension (default, 1001, 3) exited non-zero on 'SIP/localhost-00000000' localhost*CLI> And fourth script (asteriskConnector.sh): ***: 127.0.0.1 Connecting to 192.168.0.103:5038 Got an invite request Got a dialog terminated even In this case Twinkle rings all the time and cannot connect. 3. Instead of 192.168.1.101 I write 127.0.0.1. Asterisk in verbose mode: localhost*CLI> -- Executing [1001@default:1] SIPAddHeader("SIP/localhost-00000004", "x-channel:SIP/localhost-00000004") in new stack -- Executing [1001@default:2] SIPAddHeader("SIP/localhost-00000004", "x-application:basic|org.speechforge.apps.demos.Parrot") in new stack -- Executing [1001@default:3] Dial("SIP/localhost-00000004", "SIP/Zanzibar") in new stack -- Called Zanzibar -- SIP/Zanzibar-00000005 is ringing -- SIP/Zanzibar-00000005 answered SIP/localhost-00000004 localhost*CLI> == Spawn extension (default, 1001, 3) exited non-zero on 'SIP/localhost-00000004' Fourth script (asteriskConnector.sh): *: 127.0.0.1 Connecting to 192.168.0.103:5038 Got an invite request Received an unhandled SIP response status code (ignoring it): 183 : Session progress java.lang.Exception: Application Type basic not supported In this case Asterisk receives the call from Twinkle immediately but then nothing happens. The session is established, the timer in Twinkle shows how much time ago but nothing else happens. --- Ports are correctly set. For Twinkle, SIP listens on 5061, for Asterisk 5060, openIVR 5090 and Cairo 5050. (Asterisk is informed in sip.conf that Zanzibar is on port 5090). I checked "netstat -l -p" after running all four scripts and Asterisk. It looks like Asterisk listens on all IP numbers (*) and port 5060 and Java listens only on localhost (typically 127.0.0.1). In extensions.conf in [demo] section I've got: exten => 1001,1,SIPAddHeader(x-channel:${CHANNEL}) exten => 1001,n,SIPAddHeader(x-application:basic|org.speechforge.apps.demos.Parrot) exten => 1001,n,Dial(SIP/Zanzibar) The exten which you specified in docs didn't work because of ',n' in first line instead of ',1'. ====================================== Summing up: I'm worried about this "java.lang.Exception: Application Type basic not supported" because it looks like Zanzibar/Cairo doesn't understand type which is dedicated for it. I'm also not sure what to do with 192.168.0.103:5038. And I don't quite get why there are the same things in three config files. What should I set for mySipAddress and cairoSipAddress in pbxconfig.xml? ====================================== --- > To answer your earlier question about getting results with plain old java -- without tags and grammars. You dont need to use tags -- you could analyze the raw string in java when you get the results. But at least with sphinx4, you need to either use a grammar or a language model to define what can be said. > I am thinking about adding a simple grammar feature, where you just supply a list of words. Would that be helpful? I think it would. I just would like to have everything in java file without grammar and vxml, in similar way as it is in HelloDigits in Sphinx4, where the grammar is the simplest possible. (So that I would have value of string type in java code. Every time the code would be checking if this string is empty or not. If it is not, I would enter the given loop which would check what word it is. According to what word it is, it would be able to follow some kind of action, like calculating the sum or saving results to text file. Then the string would be empty again and I would be waiting for the other time when something is recognized and the string in nonempty again. This approach wouldn't require any vxml and it requires very simple grammar with only list of words). So that's the question about how to analyze those raw results. And about language model, I needed to create it with lmtoolkit available on-line and include it in acoustic model but it is used only for big dictionaries I guess. I also thought about the way of creating the application and recompiling the whole Cairo/Zanzibar. I need to create my app and add my acoustic model. (I hope there wouldn't be any problem with loaders of acoustic model, perhaps I need to use loaders of Sphinx3 for my model). I think the only what I need is to make changes to content of zanzibar-0.1-bin.src.bz2 and I don't need content of those other archives like rtp-0.2 and client-0.2. (Again, there are the same files in src/voicexml and demo/voicexml, which do I need? I think creating my application would be just creating MyApp.java in src/java/org/speechforge/apps/demos, creating MyApp.gram in src/resources/grammar, adding my wav files to src/resources/prompts, editing sphinx-config.xml in src/resources/config to replace WSJ with my acoustic model, adding the same grammar to src/voicexml, adding empty [can it be empty file? if not, what should be here?] MyApp.vxml to src/voicexml if it is possible not to use any vxml. But if I don't use vxml, how to use wav files from /src/resources/prompts in MyApp.java from /src/java/org/speechforge/apps/demos? I mean what code would be responsible for that? Do I miss anything in the process of developing the application?). Then I would need to recompile it because I've got new apps and other acoustic model. In Sphinx4 the recompiling is very simple task, it is simply running 'ant' in its main directory, which recompiles only those files which need to be recompiled due to some changes. Is there similar simple way to do it with Zanzibar? But first and most important thing is about running this parrot example as explained earlier in this mail. Thanks again! Regards! |
From: spencer l. <spe...@gm...> - 2010-01-14 18:45:07
|
Hi, Can you start by checking the zanzibar configuration file that your are using? Specifically the sipservice section. Make sure that the cairoSipAddress is set to the same machine that is running cairo (more specifically rserver.sh). If you can send the file to me I will take a look too. <bean id="sipService" class="org.speechforge.zanzibar.sip.SipServer" init-method="startup" destroy-method="shutdown"> ... </bean On Thu, Jan 14, 2010 at 10:19 AM, johny jj2 <joh...@gm...> wrote: > Hello! > > It looks like there is PROBLEM WITH ASTERISKCONNECTOR.SH, which I > cannot fix (I show only parts indicating errors or warnings): > > Unable to locate MessageSource with name 'messageSource': using > default [org.springframework.context.support.DelegatingMessageSource@b8deef > ] > Unable to locate ApplicationEventMulticaster with name > 'applicationEventMulticaster': using default > [org.springframework.context.event.SimpleApplicationEventMulticaster@4a6cbf > ] > > ... > > Starting up the main Server... > ***: 127.0.0.1 > Connecting to 192.168.0.103:5038 > IO Excepton while loging in to asterisk manager interface. Call > control services is disabled. > > ... > > THE PROBLEM IS HERE - I OBTAIN THIS ERROR IN TERMINAL. The above was > shown after running asteriskConnector (of course I run those three > other sh files before connector), the below appears when I call > Zanzibar from Twinkle. It looks like everything is all right from > Asterisk point of view, however it is not all right for Zanzibar part. > I think so because in verbose mode of Asterisk I see that Twinkle > calls Zanzibar properly but Zanzibar cannot receive the call. So it is > ringing and cannot connect. > > Got an invite request > javax.sip.SipException: IO Error sending request > at > gov.nist.javax.sip.stack.SIPClientTransaction.sendRequest(SIPClientTransaction.java:940) > at > org.speechforge.cairo.sip.SipAgent.sendInviteWithoutProxy(SipAgent.java:443) > at > org.speechforge.zanzibar.sip.SipServer.processInviteRequest(SipServer.java:372) > at > org.speechforge.cairo.sip.SipListenerImpl.processInvite(SipListenerImpl.java:446) > at > org.speechforge.cairo.sip.SipListenerImpl.processRequest(SipListenerImpl.java:123) > at gov.nist.javax.sip.EventScanner.deliverEvent(EventScanner.java:223) > at gov.nist.javax.sip.EventScanner.run(EventScanner.java:492) > at java.lang.Thread.run(Thread.java:619) > Caused by: java.io.IOException: Invalid argument > at java.net.PlainDatagramSocketImpl.send(Native Method) > at java.net.DatagramSocket.send(DatagramSocket.java:612) > at > gov.nist.javax.sip.stack.UDPMessageChannel.sendMessage(UDPMessageChannel.java:641) > at > gov.nist.javax.sip.stack.MessageChannel.sendMessage(MessageChannel.java:183) > at > gov.nist.javax.sip.stack.SIPTransaction.sendMessage(SIPTransaction.java:734) > at > gov.nist.javax.sip.stack.SIPClientTransaction.sendMessage(SIPClientTransaction.java:480) > at > gov.nist.javax.sip.stack.SIPClientTransaction.sendRequest(SIPClientTransaction.java:936) > ... 7 more > Got a dialog terminated event > > And my second question - is there any way to create very simple > grammar like <words> = word_one | ... | word_last, very simple vxml > file (how should it look like?) and move the whole logic of the talk > from grammar and vxml to .java source code? I mean - in similar way to > how it is handled in Sphinx4, which doesn't require vxml. I would be > grateful for little example. > <words> = word_one|word_two|word_three; will let you say just one of the three words <words> = (word_one|word_two|word_three)+; will let you say each word, in any order, 1 or more times <words> = (word_one|word_two|word_three)*; will let you say each word, in any order, 0 or more times? <words> = (word_one word_two word_three); will let yous say word_one word_two word_three, once, in the order written Hope this helps. Let me know if you have any more questions. Also you can look here for more details. http://java.sun.com/products/java-media/speech/forDevelopers/JSGF/ I am thinking about adding a simple grammar feature, where you just supply a list of words. Would that be helpful? To answer your earlier question about getting results with plain old java -- without tags and grammars. You dont need to use tags -- you could analyze the raw string in java when you get the results. But at least with sphinx4, you need to either use a grammar or a language model to define what can be said. > For more details (answers to your questions, content of .conf files > etc) you can have a look at my previous message. > > Thank you for your time! > Regards! > > > ------------------------------------------------------------------------------ > Throughout its 18-year history, RSA Conference consistently attracts the > world's best and brightest in the field, creating opportunities for > Conference > attendees to learn about information security's most important issues > through > interactions with peers, luminaries and emerging and established companies. > http://p.sf.net/sfu/rsaconf-dev2dev > _______________________________________________ > cairo-user mailing list > cai...@li... > https://lists.sourceforge.net/lists/listinfo/cairo-user > |
From: johny j. <joh...@gm...> - 2010-01-14 18:19:44
|
Hello! It looks like there is PROBLEM WITH ASTERISKCONNECTOR.SH, which I cannot fix (I show only parts indicating errors or warnings): Unable to locate MessageSource with name 'messageSource': using default [org.springframework.context.support.DelegatingMessageSource@b8deef] Unable to locate ApplicationEventMulticaster with name 'applicationEventMulticaster': using default [org.springframework.context.event.SimpleApplicationEventMulticaster@4a6cbf] ... Starting up the main Server... ***: 127.0.0.1 Connecting to 192.168.0.103:5038 IO Excepton while loging in to asterisk manager interface. Call control services is disabled. ... THE PROBLEM IS HERE - I OBTAIN THIS ERROR IN TERMINAL. The above was shown after running asteriskConnector (of course I run those three other sh files before connector), the below appears when I call Zanzibar from Twinkle. It looks like everything is all right from Asterisk point of view, however it is not all right for Zanzibar part. I think so because in verbose mode of Asterisk I see that Twinkle calls Zanzibar properly but Zanzibar cannot receive the call. So it is ringing and cannot connect. Got an invite request javax.sip.SipException: IO Error sending request at gov.nist.javax.sip.stack.SIPClientTransaction.sendRequest(SIPClientTransaction.java:940) at org.speechforge.cairo.sip.SipAgent.sendInviteWithoutProxy(SipAgent.java:443) at org.speechforge.zanzibar.sip.SipServer.processInviteRequest(SipServer.java:372) at org.speechforge.cairo.sip.SipListenerImpl.processInvite(SipListenerImpl.java:446) at org.speechforge.cairo.sip.SipListenerImpl.processRequest(SipListenerImpl.java:123) at gov.nist.javax.sip.EventScanner.deliverEvent(EventScanner.java:223) at gov.nist.javax.sip.EventScanner.run(EventScanner.java:492) at java.lang.Thread.run(Thread.java:619) Caused by: java.io.IOException: Invalid argument at java.net.PlainDatagramSocketImpl.send(Native Method) at java.net.DatagramSocket.send(DatagramSocket.java:612) at gov.nist.javax.sip.stack.UDPMessageChannel.sendMessage(UDPMessageChannel.java:641) at gov.nist.javax.sip.stack.MessageChannel.sendMessage(MessageChannel.java:183) at gov.nist.javax.sip.stack.SIPTransaction.sendMessage(SIPTransaction.java:734) at gov.nist.javax.sip.stack.SIPClientTransaction.sendMessage(SIPClientTransaction.java:480) at gov.nist.javax.sip.stack.SIPClientTransaction.sendRequest(SIPClientTransaction.java:936) ... 7 more Got a dialog terminated event And my second question - is there any way to create very simple grammar like <words> = word_one | ... | word_last, very simple vxml file (how should it look like?) and move the whole logic of the talk from grammar and vxml to .java source code? I mean - in similar way to how it is handled in Sphinx4, which doesn't require vxml. I would be grateful for little example. For more details (answers to your questions, content of .conf files etc) you can have a look at my previous message. Thank you for your time! Regards! |
From: johny j. <joh...@gm...> - 2010-01-10 23:50:05
|
Thanks for your answer! Summing up most crucial things from what I wrote below, is there any way to move the whole logic from vxml and gram files to .java file? If it is, may you give some examplary code showing this approach, please? May you also write something more about "analyze the raw results" and "select the name of a wav file" parts? ------------------------ You say you are not vxml expert. Unfortunately I'm neither. I tried to find some kind of internet forum or mailing list for vxml users but I couldn't. One of those "forums" didn't contain internet forum, the other was forum for 'Java VoiceXML Interpreter' application. Do you know any vxml forum? I found this great example about vxml http://www.w3.org/TR/voicexml20/#dml2.1.4 (second code in the section - credit card information). It uses digits.grxml (http://mail-archives.apache.org/mod_mbox/jakarta-taglibs-dev/200506.mbox/%3C14269939D726BB43BAD3D11BBE5B8B0C6FA361@ukflumail01.FLUENCYVOICE.LOCAL%3E). >From this example it looks like I can ask the user to speak twelve digits. But the above doesn't help with adding control sum calculation to the application. So I thought that it may be good idea to use some kind of inline script in vxml. I found this http://www.w3.org/TR/voicexml20/#dml1.5.3 which uses value returned by acct_info.vxml#basic which I found here http://msdn.microsoft.com/en-us/library/bb857574.aspx . I also found this http://www.w3.org/TR/voicexml20/#dml5.3.12 about using inline scripts. (Calculating control sum is not the only one thing, which is not so easy from vxml side - I also need to save recognized digits to database or text file). By the way, can I use ECMAscripts in vxml file for Cairo/Zanzibar? I also found http://www.w3.org/TR/voicexml20/#dml2.3.4 (part which says "Sorry, your credit card") which checks information about credit card. It is somehow similar to my application which also needs to check something (control sum) based on sequence of digits. I just wonder how to use it in my case. The above still doesn't help too much. The life would be much easier for me if I can simply specify grammar which has <list_of_words> = first_word | ... | last_word, then to have in source code string variable which contains actually recognized word and do everything in java source code, without any vxml or grammar files. It wouldn't be elegant solution and the only difficulty would be again with playing wav files as answers. Of course I also would need to have the loop which checks if 'string r != ""' and allow me to read the string and proceed with it if it is not null. That would involve using some variables to store actual position. Is there any way to move the whole logic to .java file? Doing it - together with allowing to play wav files from inside it - would save much time and effort for trying to do the same things with vxml. It looks like the easiest way, if it is possible. If it is, may you give some examplary code showing this approach, please? Anyway, let me show what I've got now: <?xml version="1.0" encoding="UTF-8"?> <vxml version="2.0" xmlns="http://www.w3.org/2001/vxml" xmlns:xsi="http://www.w3.org/2001/XMLSchema-instance" xsi:schemaLocation="http://www.w3.org/2001/vxml http://www.w3.org/TR/voicexml20/vxml.xsd"> <form id="get_sequences_of_digits"> <block>Welcome to the application. You can speak twelve digits, then say next. If you mistake, you can simply say the word mistake /that's other difficulty to add this feature/. After saying twelve digits, the control sum will be checked. If it is all right, you will be informed about it and allowed to say next twelve digits. If it won't be correct, you will be asked if you'd like to repeat digits again or accept digits with improper code. If you will say less or more than twelve digits and then say next, you also will be asked to accept or reject number. You can exit the application by saying exit.</block> <field name="twelve_digits"> <grammar type="application/srgs+xml" src="/grammars/digits.grxml"/> <prompt bargein="false"><audio src="/audio/speak_twelve_digits.wav"/></prompt> <!-- from examplary talk it looks like 'if' which I erased from here wasn't used --> <filled> <!-- the code below is useless, i can't specify it properly --> <if cond="(control_sum_is_ok != true")> Control sum improper. Would you like to accept it anyway? <clear namelist="card_num"/> <throw event="nomatch"/> <else/> Control sum is OK. You can proceed with new set of digits. <clear namelist="card_num"/> <throw event="nomatch"/> </if> </filled> </field> </form> </vxml> And the code of my version of digits.grxml: <?xml version="1.0"?> <grammar xmlns="http://www.w3.org/2001/06/grammar" xmlns:nuance="http://voicexml.nuance.com/grammar" version="1.0" mode="voice" xml:lang="en-US" tag-format="Nuance" root="numbers"> <!-- http://www.apache.org/licenses/LICENSE-2.0 --> <rule id="digit"> <one-of> <item> zero <tag> return("0") </tag></item> <item> jeden <tag> return("1") </tag></item> <item> dwa <tag> return("2") </tag></item> <item> trzy <tag> return("3") </tag></item> <item> cztery <tag> return("4") </tag></item> <item> piec <tag> return("5") </tag></item> <item> szesc <tag> return("6") </tag></item> <item> siedem <tag> return("7") </tag></item> <item> osiem <tag> return("8") </tag></item> <item> dziewiec <tag> return("9") </tag></item> </one-of> </rule> </grammar> -------------------------- I still wonder why you suggested me to use rserver, transmitter1, receiver1 and asteriskConnector rather than launch.sh in your first mail, if this launch should run all four things. > You can always try to parse and analyize the raw results > In any case, the code that analizes the results (either the raw results or the tags) can select the name of a wav file that will be passed into the next call to playAndRecognize(..) May I ask for some further tips how to implement it, please? > Given a number like 1234, will your grammar allow for "one two three four" or "one thousand two hundred and thirty four" or "twelve thirtyfour" I would make it possible only to use ten digits. You are fortunate because there is already existing English acoustic models of good quality on VoxForge. I'm not this fortunate (I don't use English) and the only what I can do is to create speaker-dependent model. I don't want to force users to follow too much training which would be required for more sophisticated acoustic models. > Sorry about the bug, can you let me know what you fixed, so I can check it in of others. Maybe send the fixed file? I changed only one character in one, short file (/zanzibar-0.1/bin/cairo/rserver.sh) from: sh ..\launch.sh $CLASS -sipPort 5050 -sipTransport udp to this: sh ../launch.sh $CLASS -sipPort 5050 -sipTransport udp In the first case it told me that it cannot find ..launch.sh. > Also do you remember which script had trouble? There should be no need to copy things around like that. As far as I remember it was /zanzibar-0.1/bin/cairo/transmitter1.sh. > if you can send the dialplan, I can take a look too? It is just original dialplan of newly-installed Asterisk. After all the only changes which I have done are as follows: sip.conf (http://www.spokentech.org/openivr/aik.html) [Zanzibar] type=peer host=localhost //I wasn't sure what to write here port=5090 dtmfmode=info canreinvite=no extensions.conf (Is [demo] the proper place to put it? Would I need to change it somehow in the future, when I will be using SIP provider account with PSTN numer instead of Twinkle softphone?): exten => 1001,1,SIPAddHeader(x-channel:${CHANNEL}) exten => 1001,n,SIPAddHeader(x-application:basic|org.speechforge.apps.demos.Parrot) exten => 1001,n,Dial(SIP/Zanzibar) And now I connect from Twinkle to 1001@127.0.0.1. This time I hear "connecting" but it never reaches "established" state. In the future I will use number from the SIP provider which gave this examplary configuration: http://forum.ipfon.pl/index.php?topic=64 > I am interested in building models too. Have you had much success yet? I created all the files required to build acoustic model. Those are list of phonemes for my language, list of words with their transcription, language model, wav files with their transcription for both training and testing and so on. I also configured SphinxTrain, run all of those perl scripts (there were some difficulties with doing that), created model from those files, tested it with decode.pl script and test package (my wav files with their transcriptions), obtained good results for speaker-dependent model, packed to jar file. Now I'd like to test those in Cairo/Zanzibar instead of using WSJ model. If you've got any questions about building models, just ask. If I will know, I'll answer for sure. Let me also say about process of compiling. Would it be just like: 1. I extract cairo-rtp-0.2-src.tar.bz2, 2. I create my application in src (java/.../demos -> myapp.java, resources -> grammar and wav files, voicexml -> vxml file), 3. edit configuration in src/resources/config/sphinx-config.xml, 4. type "ant" command in zanzibar-0.1 directory from terminal. I hope these changes in xml would be enough: //linguist part <property name="acousticModel" value="pl1"/> or maybe I should leave default value <property name="acousticModel" value="acousticModel"/> //dictionary part value="/home/mainaccount/acoustic/pl1/etc/pl1.dic"/> <property name="fillerPath" value="/home/mainaccount/acoustic/pl1/etc/pl1.filler"/> //acoustic model part - it is because of those model loaders I cannot use model loaders from WSJ. They suggested me to use those from sphinx3 as shown in 'transcriber' demo application. I hope it will work like this. <!-- I took the whole section from transcribe config.xml and chanded tidigits to pl1 --> <component name="tidigits" type="edu.cmu.sphinx.model.acoustic.pl1.Model"> <property name="loader" value="sphinx3Loader"/> <property name="unitManager" value="unitManager"/> </component> <component name="sphinx3Loader" type="edu.cmu.sphinx.model.acoustic.pl1.ModelLoader"> <property name="logMath" value="logMath"/> <property name="unitManager" value="unitManager"/> </component> Thanks for sharing your precious time :-) Regards! |
From: spencer l. <spe...@gm...> - 2010-01-10 19:51:55
|
On Sat, Jan 9, 2010 at 1:37 PM, johny jj2 <joh...@gm...> wrote: > Thank you for your answer! > > I've got two major difficulties, one with adding action to recognizing > words which match the grammar and the other with running Parrot from > softphone. Please, answer at least first of these two. > > ---------------------------------------------------- > > May you tell me, please, how to add action to Parrot if the {WEATHER} > is recognized? I'd like the application to respond with wav file and > to follow some java code (e.g. write to file; in my case it would be > calculating sum and playing either 1.wav or 2.wav). > > I guess there must be change to both Parrot.java and parrot.vxml. > > For example to parrot.vxml (perhaps it is not good way) - to answer > the action with wav file: > > <filled> > > <prompt> <value expr="main"/> </prompt> > <if cond="main=='weather'"> //NEW CODE BEGINS HERE > <block> > <audio src="weather_wav_file_URL"/> > </block> > </if> //NEW CODE ENDS HERE > > <if cond="main=='quit'"> > > <exit/> > > <else/> > > <clear namelist="main"/> > > <reprompt/> > > </if> > > > > </filled> > Yes, i am not a jvoicexml expert either, but you can add conditional statements like you suggest. You can also have more than one tag. Another way to make things dymamic is create the vxml dynamically -- something like a servelet. > And Parrot.java - to add some action with java code, e.g. writing to file: > > if ((rule.getTag().equals("QUIT")) && (rule.getRule().equals("main"))) > { > > //... > > } > if ((rule.getTag().equals("WEATHER")) && > (rule.getRule().equals("main"))) { > > //here e.g. I perform calculations - according to their > result I'd like the application to say either "control sum correct" or > "control sum incorrect" - wav answer is DEPENDENT ON CALCULATIONS, not > only on recognized word (which is the case of using vxml file)!!! how > to specify playing either 1.wav or 2.wav here ?!?!? > > } > Yes. something like this should do what you want. You need to construct a grammar to recognize numbers. You can always try to parse and analyize the raw results, Or you can construct the grammar so that it contains rules/tags to indicate which numbers were said. If my memory serves me, a rule should be able to conatin an array of tags, which may be helpful in your case. Given a number like 1234, will your grammar allow for "one two three four" or "one thousand two hundred and thirty four" or "twelve thirtyfour" TIn any case, the code that analizes the results (either the raw results or the tags) can select the name of a wav file that will be passed into the next call to playAndRecognize(..) > > Some additional, minor, questions: > > 1. I don't get why in parrot.vxml there is "Would you like to hear the > weather, get sports news or hear a stock quote?" if the parrot > application is about repeating what somebody says. > It only repeats, it if it is in the grammar. Take a look at the grammar, it only is set up for sports news, weather and stock quotes. Everything else is out of grammar. > 2. Why is there > ~/cairo/zanzibar-0.1-src/src/resources/prompts/parrot.wav if this wav > is never used? > It should be used. I think I added fo testing and meant to include it in a second version of the parrot demo. It should be quite easy actually. Just change line 68 in parrot.java to this. String filename = "file:///<yourpath>/zanzibar-SNAPSHOT/demo/prompts/parrot.wav"; RecognitionResult r = sClient.playAndRecognizeBlocking(true, filename,grammar, false); > ---------------------------------------------------- > > May you also help me to run this Parrot from softphone, please? > > 1. I extracted zanzibar-0.1-bin.tar.bz2 to > /home/mainaccount/cairo/zanzibar-0.1-bin. Later I did all the stuff > connected with global variables for Java, extracting JSAPI etc. > 2. I run /home/mainaccount/cairo/zanzibar-0.1-bin/bin/rserver.sh from > Terminal. (There was error in sh file - I had to change from \ to / > because it couldn't find launch.sh). > Sorry about the bug, can you let me know what you fixed, so I can check it in of others. maybe send the fixed file? > 3. I copy config directory from ~/cairo/zanzibar-0.1-bin to > ~/cairo/zanzibar-0.1-bin/bin because one of those three sh couldn't > find config directory. > 4. OK, all three sh files from ~/cairo/zanzibar-0.1-bin/bin/cairo and > also ~/cairo/zanzibar-0.1-bin/bin/zanzibar/asteriskConnector.sh are > running. > Also do you remember which script had trouble? There should be no need to copy things around like that. I would like to fix it, so other s dont run into that same problem. > 5. I add to sip.conf: > > [Zanzibar] > type=peer > host=localhost //I wasn't sure what to write here > port=5090 > dtmfmode=info > canreinvite=no > > 6. I add to extensions.conf at the end of [mainmenu] section (is it > proper place?): > > exten => 1,n,SIPAddHeader(x-channel:${CHANNEL}) > exten => > 1,n,SIPAddHeader(x-application:basic|org.speechforge.apps.demos.Parrot) > exten => 1,n,Dial(SIP/Zanzibar) > > 7. I add to /etc/asterisk/manager.conf at the end of file: > > [twinkle] > secret=password > permit=0.0.0.0/0.0.0.0 > read=system,call,log,verbose,agent,command,user > write=system,call,log,verbose,agent,command,user<http://0.0.0.0/0.0.0.0%0Aread=system,call,log,verbose,agent,command,user%0Awrite=system,call,log,verbose,agent,command,user> > > 8. I cannot find context.xml file to do what is written here > http://www.spokentech.org/openivr/aik.html > I will improv the docs. I mean the config.xml file. it used to be named context.xml. The trick is you need to change the one you are using -- an there are moe than one. It is the one being used in zanzibar.sh file. But you are not going to need this (now) -- it is only needed if you want to transfer calls out -- back through asterisk. > 9. I run 'asterisk' in Terminal > > 10. I run 'twinkle'. I choos profile: 'twinkle', SIP service provider: > 'None (direct IP to IP calls), username: twinkle, domain: localhost, > system settings -> network -> SIP port: 5061. I write in call: > '1000@127.0.0.1'. It connects me to "Congratulations. You have > successfully installed and executed the Asterisk open source". So I > can connect to Asterisk but cannot connect to Parrot demo. I guess the > call is improper but there also may be something wrong with > configurations above. > It is good that you have twinkle calling asterisk. The issue is probably now with step 6. The asterisk dialplan (in extensions.conf). It is playing the weclcome message rather than transferring the call to zanzibar. Take a closer look at the dialplan, I bet all calls play the message, for a test change that to have all calls directed to zanzibar. if you can send the dialplan , I can take a look too? > > Regards! > > PS > > Are your acoustic models in jar files? > Yes, it contains two directories (etc and model_parameters) without > loaders (it needs sphinx3 loader to be specified in xml). > I am interested in building models too. Have you had much success yet? > > > ------------------------------------------------------------------------------ > This SF.Net email is sponsored by the Verizon Developer Community > Take advantage of Verizon's best-in-class app development support > A streamlined, 14 day to market process makes app distribution fast and > easy > Join now and get one step closer to millions of Verizon customers > http://p.sf.net/sfu/verizon-dev2dev > _______________________________________________ > cairo-user mailing list > cai...@li... > https://lists.sourceforge.net/lists/listinfo/cairo-user > |
From: johny j. <joh...@gm...> - 2010-01-09 21:37:43
|
Thank you for your answer! I've got two major difficulties, one with adding action to recognizing words which match the grammar and the other with running Parrot from softphone. Please, answer at least first of these two. ---------------------------------------------------- May you tell me, please, how to add action to Parrot if the {WEATHER} is recognized? I'd like the application to respond with wav file and to follow some java code (e.g. write to file; in my case it would be calculating sum and playing either 1.wav or 2.wav). I guess there must be change to both Parrot.java and parrot.vxml. For example to parrot.vxml (perhaps it is not good way) - to answer the action with wav file: <filled> <prompt> <value expr="main"/> </prompt> <if cond="main=='weather'"> //NEW CODE BEGINS HERE <block> <audio src="weather_wav_file_URL"/> </block> </if> //NEW CODE ENDS HERE <if cond="main=='quit'"> <exit/> <else/> <clear namelist="main"/> <reprompt/> </if> </filled> And Parrot.java - to add some action with java code, e.g. writing to file: if ((rule.getTag().equals("QUIT")) && (rule.getRule().equals("main"))) { //... } if ((rule.getTag().equals("WEATHER")) && (rule.getRule().equals("main"))) { //here e.g. I perform calculations - according to their result I'd like the application to say either "control sum correct" or "control sum incorrect" - wav answer is DEPENDENT ON CALCULATIONS, not only on recognized word (which is the case of using vxml file)!!! how to specify playing either 1.wav or 2.wav here ?!?!? } Some additional, minor, questions: 1. I don't get why in parrot.vxml there is "Would you like to hear the weather, get sports news or hear a stock quote?" if the parrot application is about repeating what somebody says. 2. Why is there ~/cairo/zanzibar-0.1-src/src/resources/prompts/parrot.wav if this wav is never used? ---------------------------------------------------- May you also help me to run this Parrot from softphone, please? 1. I extracted zanzibar-0.1-bin.tar.bz2 to /home/mainaccount/cairo/zanzibar-0.1-bin. Later I did all the stuff connected with global variables for Java, extracting JSAPI etc. 2. I run /home/mainaccount/cairo/zanzibar-0.1-bin/bin/rserver.sh from Terminal. (There was error in sh file - I had to change from \ to / because it couldn't find launch.sh). 3. I copy config directory from ~/cairo/zanzibar-0.1-bin to ~/cairo/zanzibar-0.1-bin/bin because one of those three sh couldn't find config directory. 4. OK, all three sh files from ~/cairo/zanzibar-0.1-bin/bin/cairo and also ~/cairo/zanzibar-0.1-bin/bin/zanzibar/asteriskConnector.sh are running. 5. I add to sip.conf: [Zanzibar] type=peer host=localhost //I wasn't sure what to write here port=5090 dtmfmode=info canreinvite=no 6. I add to extensions.conf at the end of [mainmenu] section (is it proper place?): exten => 1,n,SIPAddHeader(x-channel:${CHANNEL}) exten => 1,n,SIPAddHeader(x-application:basic|org.speechforge.apps.demos.Parrot) exten => 1,n,Dial(SIP/Zanzibar) 7. I add to /etc/asterisk/manager.conf at the end of file: [twinkle] secret=password permit=0.0.0.0/0.0.0.0 read=system,call,log,verbose,agent,command,user write=system,call,log,verbose,agent,command,user 8. I cannot find context.xml file to do what is written here http://www.spokentech.org/openivr/aik.html 9. I run 'asterisk' in Terminal 10. I run 'twinkle'. I choos profile: 'twinkle', SIP service provider: 'None (direct IP to IP calls), username: twinkle, domain: localhost, system settings -> network -> SIP port: 5061. I write in call: '1000@127.0.0.1'. It connects me to "Congratulations. You have successfully installed and executed the Asterisk open source". So I can connect to Asterisk but cannot connect to Parrot demo. I guess the call is improper but there also may be something wrong with configurations above. Regards! PS > Are your acoustic models in jar files? Yes, it contains two directories (etc and model_parameters) without loaders (it needs sphinx3 loader to be specified in xml). |
From: spencer l. <spe...@gm...> - 2010-01-06 17:59:37
|
On Sat, Jan 2, 2010 at 6:04 PM, johny jj2 <joh...@gm...> wrote: > SpencerLord, thank you for your previous answers once more :-). > > I got rid of those difficulties which I explained in my previous post. > So now let me come to questions related only to Cairo/Zanzibar. If > those are only Cairo/Zanzibar-connected (and those are) I see no other > place at all to get the answer to these questions than this mailing > list (especially first two questions but also the third one). > > 1. First and most important question. I see things which are told by > application are written in the file parrot.vxml. So I guess it uses > Text To Speech to generate those. I found only one wav file, i.e. > parrot.wav but it doesn't include e.g. "I could not hear you" or "Your > response was out of grammar". I also cannot see where this file > parrot.wav is used in Parrot.java or parrot.vxml or any other file. In > other words it looks like this file is recorder but not used at all. > Am I right that those things written in parrot.vxml are spoken by Text > To Speech? > Where is the usage of parrot.wav specified? And the crucial > thing for my project - I cannot use any Text To Speech because this > TTS requires well-trained acoustic base for my language. (I worked > only with ASR, not with TTS but I guess there must be really much > training for TTS). Those acoustic models are freely available and of > good quality for English language but for my language there are no > acoustic models available for free. So finally coming to this most > important question - where can I specify wav files which I'd like the > application to speak? I think the most comfortable place would be > Parrot.java (or MyApp.java, created similarly as Parrot.java). My > application would ask the user to speak some digits, then it will > calculate control sum based on those digits and say to the user "The > control sum is correct" or, in the other case, "The control sum is > incorrect. Do you want to accept incorrect sum?". So it looks like it > cannot be specified in vxml file because it is not a standard thing > for voice control (by standard thing I mean something like "out of > grammar" or "I could not hear you"). And if it is not a standard > thing, I think it can be specified only in Parrot.java. > Yes the Parrot demo does use tts and not pre-recoded audio. I should add an option to use pre-recored audio. Take a look at the Jukebox demo, it plays pre-recorded audio. For example in this method dylan variable is a url to an audio clip. sClient.playAndRecognizeBlocking(true, dylan, playGrammar, true); Also note that the SpeechClient has a queuePrompt and playBlocking method which have two parameters. IF the urlPrompt flag is true, then the second parameter is a url to an audio file. if false, it is text to be synthesized. As far as vxml playing audio, I am not a vxml expert but I think that this will play a pre-recorded audio file <block> <audio src="wav_file_URL"/> </block> This will do tts: <block> <audio>Hello, World</audio> </block> I think you are talking about getting the semantic meaning from what was spoken. That is usually done by tags in the grammars. The version of jvoicexml that I used in this release did not have support for extracting the semantic meaning. They may have added that since, in whihc case I should release a new version. But note that even as is you can do some conditional logic <if cond="main=='quit'"> <exit/> <else/> <clear namelist="main"/> <reprompt/> </if> > > 2. Other, also important thing. I don't see in sphinx-config.xml any > line which indicates where Sphinx4 is installed. So from this it looks > like Sphinx4 doesn't have to be installed if the directory where it is > installed is not specified in the configuration file. But on the other > hand Cairo/Zanzibar is responsible only for connecting Asterisk with > Sphinx4 so it looks like Sphinx4 has to be installed. What do I lack > in my understanding of the issue? Where to specify the directory for > Sphinx4? Do I need to have this Sphinx4 installed after all? (I guess > I need). > Cairo uses an internal sphinx_config file. It is inside the cairo jar. You can specifiy your own by setting the path in cairo-config. The sphinx jars are alos included -- so no need to install your own version of sphinx. > > 3. Do I have all the required files in my acoustic model jar archive? > I've got two directories (etc and model_parameters). In etc there are > files which were input for SphinxTrain (dictionary, filler, lm, > transcriptions). In model_parameters there is only one directory > pl1.ci_cont. First of all I don't have any ModelLoader.class > (http://images45.fotosik.pl/242/a07bc3c15d943928.jpg) in my pl1.jar > file because it wasn't created by SphinxTrain. More information (if > needed) about my acoustic model given here > (http://www.speedyshare.com/files/20015137/foto.rar) in files 48-53 > (WSJ model), 54-64 (my way of creating the model), 65-69 (final > version of jar file with model). > I can not help you too much here. I am not yet up to speed with building acoustic models. We are using the existing WSJ 8khz models that are packaged in jar files. You can specify your own models, by setting up your own sphinx-config file but make sure that teh jar files are in the path (just put them in the lib directory) Are your acoustic models in jar files? > > 4. In Parrot.java I see [CODE]private String grammar; // = > "file:.../demo/grammar/example-loop.gram";[/CODE]. First thing, why > can't I see any place in Parrot.java which indicates that this grammar > is used? (Similarly the [CODE]private String prompt[/CODE] is > specified only in parrot.vxml). And second thing, why in some places > it is example-loop.gram and in the other parrot.gram? > The grammar is actually used by sphinx. Parrot.java does pass the grammar url in the recognize methods. Similar answer for parrot.vxml, the grammar does evntually get to the sphinx engine and is used for recognition. > > 5. And minor question. May you explain, please, to me this kind of > syntax > value="resource:/org.speechforge.cairo.server.recog.sphinx.SphinxRecEngine!/grammar" > ? I saw similar things in Sphinx4 files and I don't know why some part > of it is before exclamation mark and some is after. You can have a > look here > http://forum.idg.pl/programowanie-f119-zmiana_linka_w_pliku_xml-t193808.html > . The only what matters is the code which I included in the post. > First is what I had, second what I changed it to. Similarly third is > what I had, fourth what I changed it to. > > Thanks for answers in advance! > Regards! > > > ------------------------------------------------------------------------------ > This SF.Net email is sponsored by the Verizon Developer Community > Take advantage of Verizon's best-in-class app development support > A streamlined, 14 day to market process makes app distribution fast and > easy > Join now and get one step closer to millions of Verizon customers > http://p.sf.net/sfu/verizon-dev2dev > _______________________________________________ > cairo-user mailing list > cai...@li... > https://lists.sourceforge.net/lists/listinfo/cairo-user > |
From: johny j. <joh...@gm...> - 2010-01-03 02:04:10
|
SpencerLord, thank you for your previous answers once more :-). I got rid of those difficulties which I explained in my previous post. So now let me come to questions related only to Cairo/Zanzibar. If those are only Cairo/Zanzibar-connected (and those are) I see no other place at all to get the answer to these questions than this mailing list (especially first two questions but also the third one). 1. First and most important question. I see things which are told by application are written in the file parrot.vxml. So I guess it uses Text To Speech to generate those. I found only one wav file, i.e. parrot.wav but it doesn't include e.g. "I could not hear you" or "Your response was out of grammar". I also cannot see where this file parrot.wav is used in Parrot.java or parrot.vxml or any other file. In other words it looks like this file is recorder but not used at all. Am I right that those things written in parrot.vxml are spoken by Text To Speech? Where is the usage of parrot.wav specified? And the crucial thing for my project - I cannot use any Text To Speech because this TTS requires well-trained acoustic base for my language. (I worked only with ASR, not with TTS but I guess there must be really much training for TTS). Those acoustic models are freely available and of good quality for English language but for my language there are no acoustic models available for free. So finally coming to this most important question - where can I specify wav files which I'd like the application to speak? I think the most comfortable place would be Parrot.java (or MyApp.java, created similarly as Parrot.java). My application would ask the user to speak some digits, then it will calculate control sum based on those digits and say to the user "The control sum is correct" or, in the other case, "The control sum is incorrect. Do you want to accept incorrect sum?". So it looks like it cannot be specified in vxml file because it is not a standard thing for voice control (by standard thing I mean something like "out of grammar" or "I could not hear you"). And if it is not a standard thing, I think it can be specified only in Parrot.java. 2. Other, also important thing. I don't see in sphinx-config.xml any line which indicates where Sphinx4 is installed. So from this it looks like Sphinx4 doesn't have to be installed if the directory where it is installed is not specified in the configuration file. But on the other hand Cairo/Zanzibar is responsible only for connecting Asterisk with Sphinx4 so it looks like Sphinx4 has to be installed. What do I lack in my understanding of the issue? Where to specify the directory for Sphinx4? Do I need to have this Sphinx4 installed after all? (I guess I need). 3. Do I have all the required files in my acoustic model jar archive? I've got two directories (etc and model_parameters). In etc there are files which were input for SphinxTrain (dictionary, filler, lm, transcriptions). In model_parameters there is only one directory pl1.ci_cont. First of all I don't have any ModelLoader.class (http://images45.fotosik.pl/242/a07bc3c15d943928.jpg) in my pl1.jar file because it wasn't created by SphinxTrain. More information (if needed) about my acoustic model given here (http://www.speedyshare.com/files/20015137/foto.rar) in files 48-53 (WSJ model), 54-64 (my way of creating the model), 65-69 (final version of jar file with model). 4. In Parrot.java I see [CODE]private String grammar; // = "file:.../demo/grammar/example-loop.gram";[/CODE]. First thing, why can't I see any place in Parrot.java which indicates that this grammar is used? (Similarly the [CODE]private String prompt[/CODE] is specified only in parrot.vxml). And second thing, why in some places it is example-loop.gram and in the other parrot.gram? 5. And minor question. May you explain, please, to me this kind of syntax value="resource:/org.speechforge.cairo.server.recog.sphinx.SphinxRecEngine!/grammar" ? I saw similar things in Sphinx4 files and I don't know why some part of it is before exclamation mark and some is after. You can have a look here http://forum.idg.pl/programowanie-f119-zmiana_linka_w_pliku_xml-t193808.html . The only what matters is the code which I included in the post. First is what I had, second what I changed it to. Similarly third is what I had, fourth what I changed it to. Thanks for answers in advance! Regards! |
From: johny j. <joh...@gm...> - 2009-12-28 00:37:07
|
Hello! Thank you for your valuable answer, SpencerLord. I try my best to follow all the things which I mentioned and to take into account your advices. May I ask you for a favor, please? Let's have a look here: ================== http://johnyjj2.page.tl/ ================== I know your time is precious so in order not to waste it, I created something like little tutorial screen by screen about what I have already configured. It should take much less time than reading long post. It contains short explanations and some of my doubts. I hope you can answer those questions. Mainly they are still about including my own acoustic model in Cairo/Zanzibar. I know it can be simply done by adding the proper information to sphinx-config.xml. But if you use WSJ model by default, I need to create my model very similar in structure to WSJ model. The second of main issues is the place for specifying the algorithm. It looks like doing it in similar way to Parrot Speech Application (http://www.spokentech.org/openivr/writing-speechlets.html) would be best choice. However my doubts would be more clear after you will have a look at the site with pictures (http://johnyjj2.page.tl/). There are also some minor difficulties, they are explained at the top of the site and in the pictures with red color. Thanks for help, regards! |
From: spencer l. <spe...@gm...> - 2009-12-22 18:13:33
|
On Mon, Dec 21, 2009 at 12:27 PM, johnyjj2 <joh...@gm...> wrote: > > Hello! > > =================== GENERAL INFORMATION ABOUT WHAT I'D LIKE TO DO > ============================= > > I'd like to create such an application that: > 1. User calls special number from mobile phone. > 2. It connects him/her to server. (The user chooses if he/she wants to use > DTMF or ASR but now let's focus only on ASR). > 3. He/she speaks twelve digits, then say "details", speak information about > the details, says "next" and repeats the same, until he/she says "finish". > >From time to time server may say something, not based on TTS, but based on > prerecorded mp3/wav files. (The server has to check control sum of those > twelve digits every time according to some kind of simple mathematical > algorithm and inform the user if the number was correct or not)! > 4. The server saves results of recognition in the database. > > =================== WHAT I ALREADY HAVE DONE > ================================================== > > I created acoustic model for my language in SphinxTrain. My language is not > supported in VoxForge (which has got about eight languages, including > English, Spanish, Russian, French and so on). So I have to use my acoustic > model for speech recognition. > > I also have got .java source code file with . > > =================== WHAT I'D LIKE YOU TO HELP ME WITH > ========================================= > > I'd like you to tell me where and how I should specify my algorithm. > > The question "where" is about ninth step of "Steps which I'd like to > follow, > connected with Zanzibar and Cairo" section. > > The question "how" is also connected with formal grammars because I think > it > may be much easier to do everything based on very simple grammar, where > there is only one group of words, which simply contains all the words and > have the main algorithm in source code rather than in formal grammar. > > I also would be greatful for help with integrating Sphinx with Asterisk. My > doubts about integrating are shown in "Steps which I'd like to follow, > connected with Zanzibar and Cairo" section. > > What I'd like to is not the answer to those minor questions but help with > following the fifth step. I indicated important issues with bold font. I > gave those minor question to show you what is or is not clear for me after > reading on-line documentation of Cairo and Zanzibar. > > =================== GENERALS STEPS WHICH I'D LIKE TO FOLLOW > =================================== > > (It is just general idea, I'd like you to help me with fifth point). I > thought about doing the following: > 1. Install Twinkle in order to emulate calling from mobile phone to server. > 2. On the same computer I install Asterisk. > 3. I configure SIP trunk for Asterisk so that it would be able to receive > calls from Twinkle. How should this configure file look like for Twinkle? > I have not used twinkle, but I have used xlite to do something simular. This should be straightforward SIP configuration in Twinkle and Asterisk. > 4. I install Sphinx4 (I already have got Sphinx4 on my computer). I finish > creating application for Sphinx4, which uses my acoustic model, my grammar > and my algorithm. > 5. I use Zanzibar and Cairo to integrate Sphinx4 with Asterisk. (Look at > the > next section). This step looks like the most difficult for me. Especially > because I'm not quite sure where and how to specify my algorithm. > Sphinx4 is bundled in Cairo. So that part of the integration is done. If you have your own modified version of Sphinx4, you will have to replace the sphinx jars in the cairo lib directory. Depending on what you customized in sphinx4, you may have to change cairo and rebuild. Is it just the acoustic models? Zanzibar handles the inegration with Asterisk as well as selecting the application/algorithm to run once a call is connected. Your algorithm can be specified in voicexml or in java inside zanzibar. Or if you use just dtmf, you could probably put your algorithm in the asterisk dialplan. > 6. I create additional dialplan so that: a) the system would ask the user > with the use of DTMF if he/she wants to use ASR or DTMF in main session. If > he/she chooses ASR, the further communication would be with Sphinx4. If > he/she chooses DTMF, everything would be based on DTMF. How can I do such a > thing? Previously I thought that those would be two different, independent > systems (midlet on mobile phone, using httpconnection, post method and > Tomcat on server). Now I guess perhaps using DTMF in Asterisk may be better > solution. > You have options. You can use asterisk dialplan for dtmf. Or you can the java application in zanziabr to do dtmf too. Voicexml should also handle dtmf -- but is not fully tested in zanzibar. Zanzibar only supports DTMF over SIP info messages -- good enough for most needs. Midlet solution is probably possible too work too. > 7. I test my application to ensure myself that the whole system works fine. > 8. I buy account from SIP provider with PSTN number. I install Asterisk and > Sphinx4 on server (previously it was tested on my computer). Possible > problems: too slow internet connection. Solution: I will have to use Digium > card or VoIP instead of SIP, or rent server in Data Centre. Other possible > problem: Windows instead of Linux. I guess using Linux for Asterisk is > better idea, however I hope there shouldn't be big difference between > followin all of mentioned steps in Linux and doing the same in Windows. > I have only run asterisk on linux. SIP/RTP over internet has worked pretty well for me. But I have not done high volume calls. Slow upload speeds with many internet plans could be a problem. Running in server room shuld solve that. > 9. I record samples of destination users' voices. There is no good (several > or hundreads hours of recordings) acoustic model for free, available for > Polish language, the only what is left is speaker-dependent system. I > create > new acoustic model, taking into account new speech samples. > Cairo does not have a recorder resource yet. I started one, but it is not complete. You can record on asterisk as a workaround. 10. System starts working. > 11. I create some kind of feedback to obtain the knowledge how to improve > the system. > 12. I create web application to enable access to obtained data, through > system of information available everywhere in the internet. > > =================== STEPS WHICH I'D LIKE TO FOLLOW, CONNECTED WITH ZANZIBAR > AND CAIRO ======================== > > 1. Install Cairo: http://www.speechforge.org/projects/cairo/install.html > 2. Install Zanzibar: http://www.spokentech.org/openivr/install.html > 3. Start the Cairo server: > http://www.speechforge.org/projects/cairo/intro.html . Should I use > bin/lanuch.sh (if there is sh, not only bat)? Or rather rserver.bat/sh, > transmitter1.bat/sh, receiver1.bat/sh? I guess I need bin/launch.sh. > use rserver, transmitter1 and receiver1 > 4. Start the openIVR Server (Zanzibar server): > http://www.spokentech.org/openivr/intro.html . Should I use allinone.bat > (or > rather sh for Linux, if there is any sh)? Or rather rserver.bat, > transmitter1.bat, receiver1.bat? > if you use allinone, you dont need to run rserver, transmitter and receiver alternatively if you ran the 3 cairo processes you should run asteriskConnector.sh too > 5. Configure Zanzibar server: http://www.spokentech.org/openivr/intro.html. > You say "no changes should be required if you are running zanzibar and > cairo > on the same machine" so I guess no changes are needed. However shouldn't I > change mySipAddress or port? Don't I need configure the Dialog Service by > using ApplicationBySipHeaderService? > mySipAddress and port could be changed or stay the same, just need to be the same as the port and address you use in asterisk when you transfer a call. Yes, you probably want to change from applicationbynumber service to applicationbysipheaderservice > 7. Softphone: http://www.speechforge.org/projects/cairo/intro.html("Running > the examples"). Temporarily configure Zanzibar for Twinkle (instead of > Xlite > which I don't know). Later I guess I will need to erase changes in > configuration, after having access to normal SIP provider account, rather > than Thinkle. > No chanegs should be needed in zanzibar. The config I show for xlite will connect directly to zanzibar from the sip phone for easy testing without requireing asterisk. You could also configure the sip sphone to call asterisk first. > 8. Follow http://www.spokentech.org/openivr/intro.html "Running in > Asterisk > mode - Dialplan integration" section. > 9. Write my own speech application. The main question is here. Should I use > vxml or Cairo client API for writing my application? Or sepcify it in > dialplan of Asterisk? Or maybe in runApplication method? Or in java source > code of Sphinx4 applicaton? Or should I write logic of my app in code based > on this Parrot Speech Application code from > http://www.spokentech.org/openivr/writing-speechlets.html ? In > http://www.spokentech.org/openivr/aik.html I found that I may write logic > of > my app either in voicexml or in java apps (mrcp4j api or ciairo-client > api). > <u>In other words - where and how to specify my algorithm?</u> > All of those options are available to you. depends on the algorithm you want to run. If an asterisk dialplan will work for you, then you dont need zanziabr and cairo. If you are familiar with voicexml, maybe that is a good choice -- it is a standard. The Parrot application approach is good if you are most comfortable writing applications in java. > 10. Follow http://www.spokentech.org/openivr/aik.html . > > =================== MINOR QUESTIONS CONNECTED WITH ZANZIBAR AND CAIRO > ============================================ > > 1. http://www.speechforge.org/projects/cairo/intro.html -> Running the > Demo > MRCPv2 Clients -> Available Clients -> "Each client can be started by > running the appropriate batch script located in the demo/bin directory of > your Cairo installation" -> after all: I need to run batch to start demo. > Should I run batch directly? > 2. http://www.speechforge.org/projects/cairo/intro.html -> Running the > Cairo-client Demo -> Can I use only bargein and parrot to write my code? > Why > only demo-bargein and demo-parrot are available in the cairo-client? At > first I though demo-standalone would be more proper starting point for my > application but it looks like it is not available in the cairo-client. > 3. http://www.speechforge.org/projects/cairo/dependencies.html -> Project > Dependencies -> runtime -> WSJ_8gau_13dCep_16k_40mel_130Hz_6800Hz -> Can't > I > use my own acoustic model?!?! > Yes. You will need to update the sphinx-config.xml > 4. http://www.speechforge.org/projects/cairo/dependencies.html -> Project > Transitive Dependencies -> Do I need to install all of these dependencies > by > "sudo get-app install" or are they included in Cairo? > They are all included. No additional downloads required other than java and jmf (and run jsapi.sh) > 5. http://www.speechforge.org/projects/cairo/dependencies.html -> > Dependency > Listings -> Commons Configuration -> "Tools to assist in the reading of > configuration/preferences files in various formats" -> Would I benefit from > using it? > Probably not. Its a java lib used internally. for the config.xml files. > 6. http://www.speechforge.org/projects/cairo/dependencies.html -> > Dependency > Listings -> Codec -> "phonetic encoding utilities" -> I already created my > list of phonemes for SphinxTrain. Is it somehow connected with it? > No connection. > 7. http://www.speechforge.org/projects/cairo/dependencies.html -> Project > Dependency Graph -> Dependency Listings -> Unnamed - > sphinx:WSJ_8gau_13dCep_8kHz_31mel_200Hz_3500Hz:jar:1.0beta -> Why both > TIdigits and WSJ are used? > For junit tests only. > 8. http://www.spokentech.org/openivr/intro.html -> Running the Examples > (in > Demo Mode) -> Prerequisites -> "You will need a sip softphone like Xlite to > access the demos" -> How to use Twinkle instead? Is the configuration the > same? > Should be no problem. Config should be logically the same. > 9. http://www.spokentech.org/openivr/intro.html -> Running the Examples > (in > Demo Mode) -> Prerequisites -> "you will require (preferably high quality) > microphone" -> Why does it require good quality microphone? I thought in > reality there wouldn't be good quality, but rather poor 8kHz telephone > speech, not 16kHz which is possible for microphones? > yeah, you are right. I should change that. > 10. http://www.spokentech.org/openivr/intro.html -> Running the Examples > (in > Demo Mode) -> Prerequisites -> What's the difference between parrot and > vxml-parrot? By the way, I'm more or less familiar with JSGF grammars. But > wouldn't it be better for my application not to use grammars extensively, > but rather to create algorithm in source code, and use very simple grammar > which only contains list of words, like e.g. <utterance>=<list_of_words>? > Is there a away to use Sphinx4 with such a simple vocabulary? Such a simple "grammar" would be nice for most applications. > 11. http://www.spokentech.org/openivr/intro.html -> Running in Asterisk > Mode > -> Dialplan Integration -> three lines of code for x-channel and > x-application -> It is to integrate Zanzibar with Asterisk dialplan. Do I > need to add it after all? > Not sure I understand the question. > 12. http://www.spokentech.org/openivr/intro.html -> Running in Asterisk > Mode > -> Dialplan Integration -> a) type=beanId -> I don't get it, b) > type=className -> Does it mean I can use MyApplication.java source code of > Sphinx4 here?, c) type=vxml -> Can I accomplish what I need to do with > vxml? > a) it will run the java program specified in the xml file by the beanid (example Jukebox) b) Not sure what you mean by "source code of Sphinx4" c) Maybe. Not sure what you application is exactly. I > 13. http://www.spokentech.org/openivr/intro.html -> Running in Asterisk > Mode > -> Asterisk Manager Interface (AMI) Configuration -> I think I don't need > it. Am I right? > I think you are right. > 14. http://www.spokentech.org/openivr/architecture.html -> I will have > account from SIP provider with PSTN numer. I see SIP and RTP here. What do > I > need to do with this RTP? > In VOIP, RTP is used for audio streamining, SIP for signaling > 15. http://www.spokentech.org/openivr/writing-speechlets.html -> Parrot > Speech Application -> Why are those three last functions empty? > I am using blocking calls, so no need to do anything with callbacks. > 16. http://www.spokentech.org/openivr/aik.html -> Asterisk Integration Kit > -> Dialplan Integration -> What do I need to specify in my dialplan? I > guess > only speech redirection from Asterisk to Zanzibar. I also guess main part > of > my logic would be somewhere else then in dialplan, but I don't know exactly > where. > Yes, if you choose to use Zanziabr/Cairo. You may get by with just asterisk and have all logic in the dialplan. Main question I think you should answer is "Do you need speech recognition for this app?" If you are just collecting audio for model building, maybe not. > > Thanks very much for your help in advance! > Regards! > -- > View this message in context: > http://old.nabble.com/using-other-acoustic-models-in-Cairo-Zanzibar-tp26879547p26879547.html > Sent from the cairo-user mailing list archive at Nabble.com. > > > > ------------------------------------------------------------------------------ > This SF.Net email is sponsored by the Verizon Developer Community > Take advantage of Verizon's best-in-class app development support > A streamlined, 14 day to market process makes app distribution fast and > easy > Join now and get one step closer to millions of Verizon customers > http://p.sf.net/sfu/verizon-dev2dev > _______________________________________________ > cairo-user mailing list > cai...@li... > https://lists.sourceforge.net/lists/listinfo/cairo-user > |
From: johnyjj2 <joh...@gm...> - 2009-12-21 20:28:04
|
Hello! =================== GENERAL INFORMATION ABOUT WHAT I'D LIKE TO DO ============================= I'd like to create such an application that: 1. User calls special number from mobile phone. 2. It connects him/her to server. (The user chooses if he/she wants to use DTMF or ASR but now let's focus only on ASR). 3. He/she speaks twelve digits, then say "details", speak information about the details, says "next" and repeats the same, until he/she says "finish". >From time to time server may say something, not based on TTS, but based on prerecorded mp3/wav files. (The server has to check control sum of those twelve digits every time according to some kind of simple mathematical algorithm and inform the user if the number was correct or not)! 4. The server saves results of recognition in the database. =================== WHAT I ALREADY HAVE DONE ================================================== I created acoustic model for my language in SphinxTrain. My language is not supported in VoxForge (which has got about eight languages, including English, Spanish, Russian, French and so on). So I have to use my acoustic model for speech recognition. I also have got .java source code file with . =================== WHAT I'D LIKE YOU TO HELP ME WITH ========================================= I'd like you to tell me where and how I should specify my algorithm. The question "where" is about ninth step of "Steps which I'd like to follow, connected with Zanzibar and Cairo" section. The question "how" is also connected with formal grammars because I think it may be much easier to do everything based on very simple grammar, where there is only one group of words, which simply contains all the words and have the main algorithm in source code rather than in formal grammar. I also would be greatful for help with integrating Sphinx with Asterisk. My doubts about integrating are shown in "Steps which I'd like to follow, connected with Zanzibar and Cairo" section. What I'd like to is not the answer to those minor questions but help with following the fifth step. I indicated important issues with bold font. I gave those minor question to show you what is or is not clear for me after reading on-line documentation of Cairo and Zanzibar. =================== GENERALS STEPS WHICH I'D LIKE TO FOLLOW =================================== (It is just general idea, I'd like you to help me with fifth point). I thought about doing the following: 1. Install Twinkle in order to emulate calling from mobile phone to server. 2. On the same computer I install Asterisk. 3. I configure SIP trunk for Asterisk so that it would be able to receive calls from Twinkle. How should this configure file look like for Twinkle? 4. I install Sphinx4 (I already have got Sphinx4 on my computer). I finish creating application for Sphinx4, which uses my acoustic model, my grammar and my algorithm. 5. I use Zanzibar and Cairo to integrate Sphinx4 with Asterisk. (Look at the next section). This step looks like the most difficult for me. Especially because I'm not quite sure where and how to specify my algorithm. 6. I create additional dialplan so that: a) the system would ask the user with the use of DTMF if he/she wants to use ASR or DTMF in main session. If he/she chooses ASR, the further communication would be with Sphinx4. If he/she chooses DTMF, everything would be based on DTMF. How can I do such a thing? Previously I thought that those would be two different, independent systems (midlet on mobile phone, using httpconnection, post method and Tomcat on server). Now I guess perhaps using DTMF in Asterisk may be better solution. 7. I test my application to ensure myself that the whole system works fine. 8. I buy account from SIP provider with PSTN number. I install Asterisk and Sphinx4 on server (previously it was tested on my computer). Possible problems: too slow internet connection. Solution: I will have to use Digium card or VoIP instead of SIP, or rent server in Data Centre. Other possible problem: Windows instead of Linux. I guess using Linux for Asterisk is better idea, however I hope there shouldn't be big difference between followin all of mentioned steps in Linux and doing the same in Windows. 9. I record samples of destination users' voices. There is no good (several or hundreads hours of recordings) acoustic model for free, available for Polish language, the only what is left is speaker-dependent system. I create new acoustic model, taking into account new speech samples. 10. System starts working. 11. I create some kind of feedback to obtain the knowledge how to improve the system. 12. I create web application to enable access to obtained data, through system of information available everywhere in the internet. =================== STEPS WHICH I'D LIKE TO FOLLOW, CONNECTED WITH ZANZIBAR AND CAIRO ======================== 1. Install Cairo: http://www.speechforge.org/projects/cairo/install.html 2. Install Zanzibar: http://www.spokentech.org/openivr/install.html 3. Start the Cairo server: http://www.speechforge.org/projects/cairo/intro.html . Should I use bin/lanuch.sh (if there is sh, not only bat)? Or rather rserver.bat/sh, transmitter1.bat/sh, receiver1.bat/sh? I guess I need bin/launch.sh. 4. Start the openIVR Server (Zanzibar server): http://www.spokentech.org/openivr/intro.html . Should I use allinone.bat (or rather sh for Linux, if there is any sh)? Or rather rserver.bat, transmitter1.bat, receiver1.bat? 5. Configure Zanzibar server: http://www.spokentech.org/openivr/intro.html . You say "no changes should be required if you are running zanzibar and cairo on the same machine" so I guess no changes are needed. However shouldn't I change mySipAddress or port? Don't I need configure the Dialog Service by using ApplicationBySipHeaderService? 7. Softphone: http://www.speechforge.org/projects/cairo/intro.html ("Running the examples"). Temporarily configure Zanzibar for Twinkle (instead of Xlite which I don't know). Later I guess I will need to erase changes in configuration, after having access to normal SIP provider account, rather than Thinkle. 8. Follow http://www.spokentech.org/openivr/intro.html "Running in Asterisk mode - Dialplan integration" section. 9. Write my own speech application. The main question is here. Should I use vxml or Cairo client API for writing my application? Or sepcify it in dialplan of Asterisk? Or maybe in runApplication method? Or in java source code of Sphinx4 applicaton? Or should I write logic of my app in code based on this Parrot Speech Application code from http://www.spokentech.org/openivr/writing-speechlets.html ? In http://www.spokentech.org/openivr/aik.html I found that I may write logic of my app either in voicexml or in java apps (mrcp4j api or ciairo-client api). <u>In other words - where and how to specify my algorithm?</u> 10. Follow http://www.spokentech.org/openivr/aik.html . =================== MINOR QUESTIONS CONNECTED WITH ZANZIBAR AND CAIRO ============================================ 1. http://www.speechforge.org/projects/cairo/intro.html -> Running the Demo MRCPv2 Clients -> Available Clients -> "Each client can be started by running the appropriate batch script located in the demo/bin directory of your Cairo installation" -> after all: I need to run batch to start demo. Should I run batch directly? 2. http://www.speechforge.org/projects/cairo/intro.html -> Running the Cairo-client Demo -> Can I use only bargein and parrot to write my code? Why only demo-bargein and demo-parrot are available in the cairo-client? At first I though demo-standalone would be more proper starting point for my application but it looks like it is not available in the cairo-client. 3. http://www.speechforge.org/projects/cairo/dependencies.html -> Project Dependencies -> runtime -> WSJ_8gau_13dCep_16k_40mel_130Hz_6800Hz -> Can't I use my own acoustic model?!?! 4. http://www.speechforge.org/projects/cairo/dependencies.html -> Project Transitive Dependencies -> Do I need to install all of these dependencies by "sudo get-app install" or are they included in Cairo? 5. http://www.speechforge.org/projects/cairo/dependencies.html -> Dependency Listings -> Commons Configuration -> "Tools to assist in the reading of configuration/preferences files in various formats" -> Would I benefit from using it? 6. http://www.speechforge.org/projects/cairo/dependencies.html -> Dependency Listings -> Codec -> "phonetic encoding utilities" -> I already created my list of phonemes for SphinxTrain. Is it somehow connected with it? 7. http://www.speechforge.org/projects/cairo/dependencies.html -> Project Dependency Graph -> Dependency Listings -> Unnamed - sphinx:WSJ_8gau_13dCep_8kHz_31mel_200Hz_3500Hz:jar:1.0beta -> Why both TIdigits and WSJ are used? 8. http://www.spokentech.org/openivr/intro.html -> Running the Examples (in Demo Mode) -> Prerequisites -> "You will need a sip softphone like Xlite to access the demos" -> How to use Twinkle instead? Is the configuration the same? 9. http://www.spokentech.org/openivr/intro.html -> Running the Examples (in Demo Mode) -> Prerequisites -> "you will require (preferably high quality) microphone" -> Why does it require good quality microphone? I thought in reality there wouldn't be good quality, but rather poor 8kHz telephone speech, not 16kHz which is possible for microphones? 10. http://www.spokentech.org/openivr/intro.html -> Running the Examples (in Demo Mode) -> Prerequisites -> What's the difference between parrot and vxml-parrot? By the way, I'm more or less familiar with JSGF grammars. But wouldn't it be better for my application not to use grammars extensively, but rather to create algorithm in source code, and use very simple grammar which only contains list of words, like e.g. <utterance>=<list_of_words>? 11. http://www.spokentech.org/openivr/intro.html -> Running in Asterisk Mode -> Dialplan Integration -> three lines of code for x-channel and x-application -> It is to integrate Zanzibar with Asterisk dialplan. Do I need to add it after all? 12. http://www.spokentech.org/openivr/intro.html -> Running in Asterisk Mode -> Dialplan Integration -> a) type=beanId -> I don't get it, b) type=className -> Does it mean I can use MyApplication.java source code of Sphinx4 here?, c) type=vxml -> Can I accomplish what I need to do with vxml? 13. http://www.spokentech.org/openivr/intro.html -> Running in Asterisk Mode -> Asterisk Manager Interface (AMI) Configuration -> I think I don't need it. Am I right? 14. http://www.spokentech.org/openivr/architecture.html -> I will have account from SIP provider with PSTN numer. I see SIP and RTP here. What do I need to do with this RTP? 15. http://www.spokentech.org/openivr/writing-speechlets.html -> Parrot Speech Application -> Why are those three last functions empty? 16. http://www.spokentech.org/openivr/aik.html -> Asterisk Integration Kit -> Dialplan Integration -> What do I need to specify in my dialplan? I guess only speech redirection from Asterisk to Zanzibar. I also guess main part of my logic would be somewhere else then in dialplan, but I don't know exactly where. Thanks very much for your help in advance! Regards! -- View this message in context: http://old.nabble.com/using-other-acoustic-models-in-Cairo-Zanzibar-tp26879547p26879547.html Sent from the cairo-user mailing list archive at Nabble.com. |