RTMP loadVars

Similar Questions

• Pepper Flash Player in Chrome crashes on RTMP NetConnection

  Hello everyone.

  We have already presented the question pepper Flash Player with here the bug database (Bug #4010090 - Flash Player in Chrome on RTMP NetConnection tagline) but just wanted to get a discussion going and possibly get Adobe (smaily :-) to contact Google and let them know that their recent updates have a serious bug.)

  In short we have problems with flash PPAPI in the Chrome browser (pepper).  If you reboot your system with the last Flash of pepper and Chrome and go directly as a site that requires a NetConnection RTMP you will find that no connection is possible on all ports.  We have tested this over and over again and came to the conclusion, that it is a newly introduced bug.  Our lack of traffic too our services confirms him as well, that's why we reached out to Adobe and the developer community.

  A perfect test is:

  http://www.therealtimeweb.com/index.cfm/2004/10/2/FMS-port-tester?CFID=1291705 & CFTOKEN = 484 14760

  You will notice that no port connection cannot be made and if you try to access anywhere else Flash Player will crash.

  Suggestions or recommendations (in part using another browser) would be greatly appreciated.

  Hello.

  We determined that this issue is the result of the Avast Antivirus program. There seems to be a problem with all versions of Avast, even the free version with no firewall.

  We will advise Avast of this issue. We recommend that the community of developers of chrome inform Avast as well.

  Concerning

• RTMP Adaptive switching question

  Hello Mr.

  
  

  I create a flash player using osmf plugin and smilplugin code I found (ofhttps://getstreaming.wordpress.com/2011/05/27/dynamic-streaming-test-drive/#comment-349).

  
  

  I have a problem when playing SMIL file in my "player customized" through wowza media server.

  The question is adaptive switching does not work in my custom player if RTMP protocol, but my custom player play more smil with Adaptive url switching and manual switching (@! http://mediapm.edgesuite.net/OVP/content/Demo/SMIL/elephants_dream.SMIL ).

  and also play HDS and HSL streaming through wowza media server.

  I appreciate your help.

  sample smil file which I use for below Wowza

  <? XML version = "1.0" encoding = "UTF-8"? >

  < smil >

  < head >

  < base meta = "_definst_/vod / rtmp://127.0.0.1:1935" / >

  < / head >

  < body >

  <>switch

  < src = 'mp4:main360.mp4' video system-bitrate = "200000" / >

  < src = 'mp4:main720.mp4' video system-bitrate = ' 480000 "/ >"

  < src = 'mp4:sample.mp4' video system-bitrate = ' 720000 "/ >"

  < / switch >

  < / body >

  < / smil >

  Hello

  Please use following methods to play RTMP SMIL-

  private var dynResource:DynamicStreamingResource;

  private var tracingNetLoader:TracingNetLoader;

  private var element: VideoElement;

  private void createSMILObj(str:String):void {}

  try {}

  var objSMIL:Object = createSMILObj (str); Function give below

  var obj:Object is objSMIL ["files"];.

  var fileLen:Number = getObjLen (obj); function given below

  dynResource = new DynamicStreamingResource (String (objSMIL ["baseURL"]));

  vector = new Vector. (fileLen);

  for (var i: int = 0; i

  Vector [i] = new DynamicStreamingItem(obj[i].src,obj[i].bitrates,obj[i].width);

  }

  dynResource.streamItems = vector;

  tracingNetLoader = new TracingNetLoader();

  element = new VideoElement (dynResource, tracingNetLoader);

  initPlayer (element);  Pass this object OSMF element and play the video.

  } catch (error: Error) {}

  trace ("detected in the SMIL XML loading error")

  }

  }

  public static void createSMILObj(_str:String):Object {}

  var baseUrl:String;

  var smilXML:XML;

  var videoList:XMLList;

  var mainObj:Object = new Object();

  var fileObj:Object = new Object();

  var str:String;

  var tempArray:Array = _str.split (')");

  var tempStr:String = String (")"+ tempArray [1]);

  Str = String (tempStr) .split (")<><>

  Str = Str.Split (")

  Str = Str.Split ("System-Bitrate"). Join ("systembitrate");

  smilXML = new XML (str);

  baseUrl=smilXML.head.meta.@base;

  trace ("base =" + baseUrl);

  videoList = smilXML.body.switchs.video;

  try {}

  for (var i: int = 0; i

  .@width fileObj[i]={src:videoList[i].@src,bitrates:videoList[i].@systembitrate,width:videoList[i]};

  }

  } catch (error: Error) {}

  trace ("error detected in SMIL XML File Update...")

  }

  mainObj ['baseURL'] = baseUrl;

  mainObj ["files"] is fileObj;.

  Return mainObj;

  }

  public void getObjLength(obj:Object):Number {}

  var number: number = 0;

  for (var id in obj) {}

  counties ++;

  }

  return of the counties;

  }

  Best regards

  Sunil Kumar

• How to determine the RTMP URL to a hosted

  We are trying to set up a capture game card to activate the RTMP streaming protocol. We already have an account hosted in < ourdomain >. adobeconnect.com. but we do not see a RTMP url anywhere in the account settings? How can we find a RTMP url? Or if this question makes no sense, what are we missing?

  Thank you very much

  -M.

  If you want the RTMP path for a specific meeting, open the Help menu in the upper right of the meeting room. Then hold down SHIFT and click to the left on Acrobat Connect Pro. The dialogue window that opens should show the RTMP path to the meeting room.

  Not sure this will work with the game capture card, but I guess it could be interesting to test.

• Problem with rtmp streaming video

  Hello everyone.
  I am pursuing my work e-learning for my client.
  I have a lot videos to be included in the lessons; to do this, I put two dedicated servers () And HertznerAmazon) and I put for streaming rtmp.
  
  With Amazon works, the other not.
  
  If you want to try the two rivers RTMP ( (), VLC, Open address)
  
  Amazon
  RTMP: //sstjnt3ej9a0f.cloudfront.net/cfx/st/mp4: mencalez1a
  
  hertzner
  RTMP: //188.40.83.89:1935 /vod2/mencalez1a.mp4
  
  The two streams in VLC works correctly.
  but once support in Captivate, to amazon works the other does not.
  
  I enclose two screens with le basic options.
  
  Can help me?

  
  

  
  

  
  

  OK, SOLVED!

  Error was outside Captivate.

  Thank you all

• Rtmp streaming flex FlashMediaLiveEncoder

  Hi all

  My English is not very good, please don't bother me.

  I want to use the Protocol rtmp streaming in flex videoplayer

  but I tried several methods still no way to success

  OUT files stored in [Adobe Media Server 5]\webroot\live_recorded\test.flv

  <s:VideoPlayer width="350" height="250" loop="true">   
    <s:source>
         <s:DynamicStreamingVideoSource id="mySVS" host="rtmp://127.0.0.1/live/live_recorded/">
              <s:DynamicStreamingVideoItem id="dreamgirl1000" streamName="test"/>
         </s:DynamicStreamingVideoSource>
    </s:source>
  </s:VideoPlayer>
  

  &

  <s:VideoPlayer   
      source="rtmp://127.0.0.1/live/live_recorded/test"  
      width="350" height="250"  
      loop="true"/>  
  

  Question2:

  Flex has no file streaming player?

  like VLC player streaming

  key in the url [rtmp://localhost/live/livestream]

  After can play streaming

  I thank you very much.

  Hsuang.

  Using NetStream()?

• Question record RTMP using FMS

  We have a flex application to record a single video stream RTMP on FMS to limited of 100 kbps bandwidth. When the network comes down to 40 to 60 kbit/s and

  raises to normal, the recorded video must not freeze and audio/video synchronization on the recorded video should be fine.

  The problem is that we are facing, when the recorded video is played back in a player (VLC), we noticed that the video is still frozen where the

  network of dips to 40 to 60 Kbps for a few seconds and only audio playback. Also noticed the video is normally when the network is normal.

  Don't we have any solution to fix the freeze on the recorded video. Can we play with Buffertime to solve this problem? If so, how?

  Note: We have no functionality of teleconference of multi here and it's only a record only RTMP stream.

  Or, do we not have any RTMP that stores packages in the client browser and it sends to the server to store the video?

  We solved the problem by using the netstream.bufferLength. First of all, we have the buffertime to about 60 sec. When the registration is finished remove us the camera and the microphone of the netstream and seeking the bufferlength become null to a timer and after that we are closing the stream. In this way, the video is completely saved.

• RTMP to HLS

  Dear Adobe team,

  I tried to achieve long loooooooooooooong

  Here's the scenario

  There is a list of reading... who plays alongside server.

  CSA code example

  application.cleintStream = Stream.get ("somestream");

  application.cleintStream.play('playlist-video',0,-1);

  It works very well, flash player can connect to the server and read the stream.

  But the problem is the HLS out of it.

  Now the same stream is also necessary for TLS.

  How can I generate/convert this 'somestream' to 'HLS-stream'... ?

  Can any help/guide me how proceed.

  It would be really good help...

  Thank you

  Dear Charles

  Thanks for your comments, but that is not always the exact solution.

  Maybe I thought about it, here are the details of the solution

  Could save some time

  1. doing server side playlist and play the serverside

  2. once the video play at the level of the server, publish this game on livepkgr flow

  3. once the stream is published on livepkgr, FMS automatic creats HLS as LIVE stream

  4. This is how you can publish RTMP on HLS

  Code: -.

  First selection side Server brand

  application.cleintStream = Stream.get ("streamName");

  application.cleintStream.play ("mp4:folder / video1.mp4", 0, -1, true);

  application.cleintStream.play ("mp4:folder / video2.mp4", 0, -1, false);

  application.cleintStream.play ("mp4:folder / video3.mp4", 0, -1, false);

  now connect to LIVEPKGR that attached n publish this feed at LIVEPKGR

  application. NC = new NetConnection();

  application. NC. Connect ("rtmp://localhost:1935 / livepkgr");

  application.nc.onStatus = {function (info)}

  trace (info.code + "livepkgr");

  If (info.code == "NetConnection.Connect.Success") {}

  application.ns = new NetStream (application.nc);

  application.ns.setBufferTime (3);

  application.ns.Attach (application.cleintStream);

  application.ns.Publish (application.cleintStream.Name, "Live");

  }

  }

  Once you are done with this, its quite easy and simple to do output (m3u8) HLS

  Some points to note key: -.

  modify [Event.xml] under fms installed folder\applications\livepkgr\events\_definst_\streamName

  streamName

  Validate IF the flow is getting published

  streamName folder should be created under fms installed folder\applications\livepkgr\streams\_definst_\

  Then final set flow HLS should look like this [ http://localhost/hls-live/livepkgr/_definst_/streamName/streamName.m3u8 ]

  Hopefully this will save some other time.

  Note that I did with AMS-5

  See you soon

• RTMPS is affected by the bug of software?

  Or any component to use RTMPS? For example: Adobe Media Server, Flash Player (on Windows, Mac, etc.)?

  Please let me know.

  Thank you.

  Yes

• RTMPS connection delayed due to the direct connection attempt

  Hello world

  I have a big problem with the way the RTMPS connections are initiated in flash player.

  One of my customer network infrastructure does not allow direct connections to my server via port 443, but allows connections proxified (using the CONNECT method).

  The problem is that flash player expects the direct connection attempt to fail before trying the proxified... connection and on this direct infrastructure network connection timeout is 20 s (I can't change that).

  Users must wait for this time to connect to the server...

  Is there a way to tell the flash player do not attempt direct connection, or do not wait for this connection Timeout?

  Perhaps a directive in the automatic proxy configuration (.pac) file?

  I already tried every NetConnection proxyType values of parameter without success.

  Any help would be appreciated!

  You mentioned the 'CONNECT' method Have you tried "CONNECTOnly" that requires to initially use the default CONNECTION method and it is not always obey?

  Documentation on NetConnection.proxyType:

  Set the proxyType property before calling the NetConnection.connect() method.

  ...

  If the property is set to "CONNECTOnly", the player first attempts to use the CONNECT method. If this fails, there is no relief.

• RTMP video Fund works great – but Android gives me NetStream.Seek.Failed

  I have an application built using Adobe AIR. The application uses StageVideo and play a live encoded H264 RTMP stream using NetStream.play. The application works as expected in the ADLS and when it is installed as an application on the desktop.

  But if use exactly the same code, and install the application on the Android, I get the NetStatusEvent 'NetStream.Seek.Failed '. Because of this event, the therefore I am unable to play the video stream on the Android.

  So my question: If you are building an app for Android, can you not resuse the exact same NetStream code that you used on the desktop? Is there a bug on the Android runtime which prevents the RTMP to seeking properly?

  FYI: The startvalue netstream.play is-2 (so it will search recorded flow first, then a live stream.) In my case, it's a live stream)

  I use Adobe AIR 3.7.0.2090 and Samsung Galaxy Tab 3, 4.2.2 Android

  The same problem or an error occurs when you use the AIR SDK 3.8.0.930

  It is like a problem. I am now using Adobe AIR SDK 3.8.0.1040 and have done some recent updates to the code. I'm not sure what fixed the issue, but I can say that Yes, Android does not work as expected, and I reuse the same code from the desktop application for the Android.

• VOD works with HTTP but not RTMP

  I have a few FLV videos that I can't work with RTMP streaming using VOD app and player of strobe.  Locally, they play fine in VLC media player.  It will also play properly in the flash drive so accessible with an HTTP request.  When I try to listen to using the RTMP protocol, reading seems to be normal and I can get to anywhere in the video.  However, playback will stop randomly at different points in the middle of the video.  If the player to debug, I see that NetStreamTimeTrait becomes NetStreamCodes.NETSTREAM_PLAY_COMPLETE.

  Can someone tell me how can I solve the FLV file, or what steps I can take to understand why playback will stop at the beginning?

  These videos were captured using the Wowza DVR app, and I used a HDS capture utility to recompose the video in a FLV.  The resulting FLV file was missing the duration metadata, so I used "yamdi" to inject the missing metadata.  I also tried "flvfix" utility without effect.

  Extension .mp4 as renaming does not work... You will have a tool to convert your FLV to mp4 videos.

• RTMP with F4M created by F4F Packer

  I'm packing a multi bitrate stream using the Manager links F4F and protected with Flash Access 2.0.

  Is is possible to 'neutralize' the resulting F4M using the RTMP protocol?

  If so, please share a reference to the sample to the F4M.

  Thank you

  Ken Florian

  Hi Ken,

  Eric is having some issues of access forum, so he asked me to this post.

  RTMP is not returned for HDS (f4m) fragments.  Here are separate protocols, with one on RTMP (Real Time Media Protocol) and the other goes over HTTP.

  RTMP is present natively by the Flash Runtime, while HDS content should be consumed via a video player created using OSMF.

  See you soon,.

  / Eric Ha

• RTMP can be used to provide protected FA flow

  Following up on my question here

  Can I use RTMP to deliver multi bit rate, flash access 2.0 multimedia protected?

  Hi Ken,

  Yes, you can use RTMP to provide content protected by FA number debits.

  Kind regards

  James

• Audio IOS RTMP live stream

  Hi guys I hope that some of you will be able to help me.

  I'm trying to stream audio from a live RTMP stream using the NetConnection and NetStream classes. I managed to get my app works without problem on Android, but I have some major difficulty make him play the audio back on iPad. Interestingly, it works in the device emulators when you are debugging, but I guess this isn't really an accurate representation. I tried streaming RTMP in AAC and MP3, but without success to be. I check through debugging that there haven't connected to the stream, however I just get no audio playback.

  Everything I read on door to believe that this is possible on IOS that I'm only interested by audio and video not. Can anyone help?

  Code example below (it's quick and dirty! ).

  Thanks in advance!

  <? XML version = "1.0" encoding = "utf-8"? >

  " < = xmlns:fx s:View ' http://ns.Adobe.com/MXML/2009 "

  xmlns:s = "library://ns.adobe.com/flex/spark" title = "Audio" creationComplete = "init ()" >

  < s:layout >

  < s:VerticalLayout paddingLeft = "10" paddingRight = "10".

  paddingTop = "10" paddingBottom = "10" / >

  < / s:layout >

  < fx:Script >

  <! [CDATA]

  import flash.media.Video;

  import flash.net.NetConnection;

  to import flash.net.NetStream;

  Import mx.core.UIComponent;

  private var vid:Video;

  private var videoHolder:UIComponent;

  private var nc:NetConnection;

  private var defaultURL:String = '[STREAM]. "

  private var streamName:String = '[STREAMNAME]. "

  private var ns:NetStream;

  private var msg:Boolean;

  private var intervalMonitorBufferLengthEverySecond:uint;

  private function init (): void

  {

  vid = new Video();

  VID. Width = 864;

  VID. Height = 576;

  VID. Smoothing = true;

  Join the video of the scene

  videoHolder = new UIComponent();

  videoHolder.addChild (vid);

  addEventListener (SecurityErrorEvent.SECURITY_ERROR, onSecurityError);

  grpVideo.addElement (videoHolder);

  Connect();

  }

  public void onSecurityError(e:SecurityError):void

  {

  trace ("error in security :");

  }

  public void connect (): void

  {

  NC = new NetConnection();

  NC.client = this;

  nc.addEventListener (NetStatusEvent.NET_STATUS, netStatusHandler);

  nc.objectEncoding = flash.net.ObjectEncoding.AMF0;

  NC. Connect (defaultURL);

  }

  public void netStatusHandler(e:NetStatusEvent):void

  {

  Switch (e.info.code) {}

  case "NetConnection.Connect.Success":

  trace ("audio - connected successfully");

  createNS();

  break;

  case "NetConnection.Connect.Closed":

  trace ("audio - Connection closed");

  Connect();

  break;

  case "NetConnection.Connect.Failed":

  trace ("audio - connection failed");

  break;

  case "NetConnection.Connect.Rejected":

  trace ("audio - connection rejected");

  break;

  case 'NetConnection.Connect.AppShutdown ':

  trace ("audio - application shutdown");

  break;

  case 'NetConnection.Connect.InvalidApp ':

  trace ("audio - app connection not valid");

  break;

  by default:

  trace ("audio-" + e.info.code + "-" + e.info.description);

  break;

  }

  }

  public function createNS (): void

  {

  trace ("NetStream creation");

  NS = new NetStream (nc);

  NC. Call ("FCSubscribe", null, "live_production"); Do not use this option if your CA requires it

  ns.addEventListener (NetStatusEvent.NET_STATUS, netStreamStatusHandler);

  vid.attachNetStream (ns);

  OnMetaData and onCuePoint //Handle event reminders: solution to http://tinyurl.com/mkadas

  //See another solution to http://www.Adobe.com/devnet/flash/QuickStart/metadata_cue_points/

  var infoClient:Object = new Object();

  infoClient.onMetaData = function MDGs (): void {};

  infoClient.onCuePoint = function oCP (): void {};

  NS.client = clientside;

  ns.bufferTime = 0;

  NS. Play (StreamName);

  ns.addEventListener (AsyncErrorEvent.ASYNC_ERROR, asyncErrorHandler);

  function asyncErrorHandler(event:AsyncErrorEvent):void {}

  trace (Event.Text);

  }

  intervalMonitorBufferLengthEverySecond = setInterval (monPlayback, 1000);

  }

  public void netStreamStatusHandler(e:NetStatusEvent):void

  {

  Switch (e.info.code) {}

  case "NetStream.Buffer.Empty":

  trace ("audio - buffer empty :");

  break;

  case "NetStream.Buffer.Full":

  trace ("audio - buffer full :");

  break;

  case "NetStream.Play.Start":

  trace ("audio - beginning of the game :"); ")

  break;

  by default:

  trace ("audio-" + e.info.code + "-" + e.info.description);

  break;

  }

  }

  public function monPlayback (): void {}

  Print the current length of the buffer

  trace ("audio - buffer length:" + ns.bufferLength);

  trace ("audio - FPS:" + ns.currentFPS);

  trace ("audio - delay Live:" + ns.liveDelay);

  }

  public function onBWDone (): void {}

  Do nothing

  }

  public void onFCSubscribe(info:Object):void {}

  Do nothing. Prevents the error if connection to CDN.

  }

  public void onFCUnsubscribe(info:Object):void {}

  Do nothing. Prevents the error if connection to CDN.

  }

  []] >

  < / fx:Script >

  < s:Group id = "grpVideo" >

  < / s:Group >

  < / s:View >

  Just an update on this for all of you who come along after me.

  I managed to get this working on the MP3 but not AAC (I guess just AAC is not taken in charge?).

  My problem was the buffertime. Docs seemed to indicate that it should be set to 0 for live streaming, however he switch to "1" has solved my problem on the particular flow I pointed at.

  So, essentially above the justchanged line:

  ns.bufferTime = 0;

  TO

  ns.bufferTime = 1;

  Would be nice to know if someone got AAFC works though...