Adding my-streaming-amf : Getting Error
I am following the http://www.adobe.com/devnet/livecycle/articles/blazeds_gettingstarted.html
tutorial and when I go and run the app I get the following error:
[MessagingError message='Destination 'project' either does not exist or the destination has no channels defined (and the application does not define any default channels.)']
at mx.messaging.config::ServerConfig$/internalGetChannelSet()[E:\dev\3.0.x\frameworks\projec ts\rpc\src\mx\messaging\config\ServerConfig.as:624]
at mx.messaging.config::ServerConfig$/getChannelSet()[E:\dev\3.0.x\frameworks\projects\rpc\s rc\mx\messaging\config\ServerConfig.as:255]
at mx.messaging::MessageAgent/initChannelSet()[E:\dev\3.0.x\frameworks\projects\rpc\src\mx\m essaging\MessageAgent.as:1083]
at mx.messaging::MessageAgent/internalSend()[E:\dev\3.0.x\frameworks\projects\rpc\src\mx\mes saging\MessageAgent.as:1059]
at mx.messaging::Producer/internalSend()[E:\dev\3.0.x\frameworks\projects\rpc\src\mx\messagi ng\Producer.as:143]
at mx.messaging::AbstractProducer/send()[E:\dev\3.0.x\frameworks\projects\rpc\src\mx\messagi ng\AbstractProducer.as:469]
at views::ChatSystem/send()[C:\workspace\InnerImpulse\src\views\ChatSystem.mxml:11]
at views::ChatSystem/___ChatSystem_Button1_click()[C:\workspace\project\src\views\ChatSystem .mxml:29]
In my messaging-config.xml I have
In my services-config.xml I have created the
true
What am I doing wrong? In the tutorial it states that just specifying is enough, but I was getting the same error without specifying the channel. Even if I specify the channel I still get the same error.
I figured it out. The compiler flag needed to be -compiler.services instead of the default -services. Adobe needs to do a better job at documentation. They are turning out to be like IBM.
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[Webrtc] [mozilla 26] Getting unhanded state type: 1 after adding remote stream
Hi,
I am working on a webRTC application. So far, I am able to make call b/w Chrome to Chrome and mozilla to mozilla. But sometimes mozilla to mozilla and almost everytime mozilla to chrome or vice-versa not working.
I see that there is connection being made but audio/video stream are not being played.
The onaddstream callback get triggered, but just after that "Unhandled state type: 1" warning get appeared.
Please help.
here is my code of onaddstream.
pc.onaddstream = function(event) {
// TODO: Finalize this API
rtc.fire('add remote stream', event.stream, id);
setAppLog('', 'add remote stream : ' + event.stream);
console.log( ' add remote stream : ' + event.stream);
//On console..
20:25:32.895 Unknown property 'overflow-wrap'. Declaration dropped. connect.css:56
20:25:32.895 Unknown property 'spacing'. Declaration dropped. connect.css:76
20:25:32.896 Unknown property 'spacing'. Declaration dropped. connect.css:240
20:25:32.896 Error in parsing value for 'padding'. Declaration dropped. connect.css:250
20:25:32.896 Unknown pseudo-class or pseudo-element '-webkit-slider-thumb'. Ruleset ignored due to bad selector. connect.css:302
20:25:32.896 Unknown property '-moz-box-shadow'. Declaration dropped. connect.css:326
20:25:32.896 Error in parsing value for 'background'. Declaration dropped. connect.css:329
20:25:32.896 Unknown property '-moz-border-radius'. Declaration dropped. connect.css:332
20:25:32.897 SyntaxError: Using //@ to indicate source map URL pragmas is deprecated. Use //# instead jquery.min.js:1
20:25:33.453 Use of getPreventDefault() is deprecated. Use defaultPrevented instead. jquery.min.js:5
20:25:33.529 [object HTMLHeadingElement] webrtcconnection.js:51
20:25:33.532 "This appears to be Firefox" adapter.js:17
20:25:33.829 Unknown property 'box-sizing'. Declaration dropped. video
20:25:36.445 "rtc.createStream : [object Object]" webrtc.io.js:338
20:25:37.530 "rtc.on get_peers : [object Object]" webrtc.io.js:172
20:25:39.995 "rtc on ready " webrtc.io.js:476
20:25:40.221 "rtc.createPeerConnection : 71b80816-3832-2b33-9301-b04db3a561fb" webrtc.io.js:246
20:25:40.222 "rtc.addStreams : [object LocalMediaStream]" webrtc.io.js:389
20:25:40.223 "createDataChannel 71b80816-3832-2b33-9301-b04db3a561fb" webrtc.io.js:434
20:25:40.224 "rtc.sendOffer : 71b80816-3832-2b33-9301-b04db3a561fb" webrtc.io.js:287
20:25:40.225 "sendOffer : 71b80816-3832-2b33-9301-b04db3a561fb" webrtc.io.js:229
20:25:41.180 "rtc.on receive_answer : [object Object]" webrtc.io.js:217
20:25:41.181 "rtc.receiveAnswer : 71b80816-3832-2b33-9301-b04db3a561fb" webrtc.io.js:332
20:25:41.580 " ADDING REMOTE STREAM... + 71b80816-3832-2b33-9301-b04db3a561fb : [object MediaStream]" webrtcconnection.js:573
20:25:41.586 [object HTMLVideoElement] webrtcconnection.js:576
20:25:41.601 " Remote stream added" webrtcconnection.js:671
20:25:41.603 " add remote stream : [object MediaStream]" webrtc.io.js:276
20:25:41.620 Unhandled state type: 1
Thanks
R.Hi R,
Note: If the answer is not found here, I would certainly recommend going to stackoverflow.com and posting your question. [http://stackoverflow.com/questions/tagged/webrtc]
*What version of the WebRTC API are you using, check the streaming api to see if its up to date
*The spacing error, maybe try the [http://jigsaw.w3.org/css-validator/ W3C Validator] will help find why 'spacing' css value is not recognized
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Revision: 14601
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Author: [email protected]
Date: 2010-03-05 09:47:28 -0800 (Fri, 05 Mar 2010)
Log Message:
Fix bug where parsing of stream that gets passed to NetStream.play doesn't take urlIncludesFMSApplicationInstance into account, resulting in stream not found error.
Modified Paths:
osmf/trunk/framework/OSMF/org/osmf/net/NetStreamPlayTrait.as
osmf/trunk/framework/OSMF/org/osmf/net/NetStreamUtils.as
osmf/trunk/framework/OSMF/org/osmf/net/dvr/DVRCastNetConnectionFactory.as
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m_tskAnalogIP1.Control(TaskAction.Verify);
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m_tskRunningAnalogIP1.Start();
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private void AnalogRead(IAsyncResult ar)
try
if (m_tskRunningAnalogIP1 != null && m_tskRunningAnalogIP1 == ar.AsyncState)
// Read the data
m_arrdAnalog = m_rdrAnalog1.EndReadMultiSample(ar);
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catch (Exception ex)
Regards,
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Solved!
Go to Solution.Got the solution by adding timeout.
m_tskAnalogIP1 = new Task("AnalogIP1");
m_tskAnalogIP1.AIChannels.CreateVoltageChannel("dev6/ai0", "", AITerminalConfiguration.Rse, AI_MIN_RANGE, AI_MAX_RANGE, AIVoltageUnits.Volts);
m_tskAnalogIP1.Timing.ConfigureSampleClock("", 1000.0, SampleClockActiveEdge.Rising, SampleQuantityMode.FiniteSamples, 20000);
m_tskAnalogIP1.Triggers.ReferenceTrigger.ConfigureDigitalEdgeTrigger("/Dev6/PFI0",DigitalEdgeReferenceTriggerEdge.Rising, 800);
m_tskAnalogIP1.Control(TaskAction.Verify);
m_tskRunningAnalogIP1 = m_tskAnalogIP1;
m_tskRunningAnalogIP1.Start();
m_rdrAnalog1 = new AnalogMultiChannelReader(m_tskAnalogIP1.Stream);
m_tskAnalogIP1.Stream.Timeout = 30000;
m_tskAnalogIP1.SynchronizeCallbacks = true;
m_callbkAnalogIP1 = new AsyncCallback(AnalogRead);
m_rdrAnalog1.BeginReadMultiSample(v_iNoOfSamples, m_callbkAnalogIP1, m_tskAnalogIP1);
Delay(2000);
m_wrtrDigital1.BeginWriteWaveform(true, m_waveformDO, m_callbkDigitalOP1, m_tskDigitalOP1); // forced trigger
private void AnalogRead(IAsyncResult ar)
try
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m_boolAnalogSamplingFinished = true;
catch (Exception ex)
Regards,
Shrinivas
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