Rtmp live streaming of videos!

I am loving Azure Platform. I am having certain problems, i hope you can help me resolve them.
Problem : I want to live stream videos using rtmp. The input for rtmp will be using ffmpeg running in my azure VM. I am new to using rtmp. Any help is welcome. Thank you.

hi i met a problem
ffmpeg -v verbose -y -i sample.mp4 -strict -2 -c:a aac -b:a 16k -ar 24000 -r 30 -g 60 -b:v 2m -c:v libx264 -f flv rtmp://livetest1-tracyone.channel.mediaservices.chinacloudapi.cn:1935/live/1ea048c582e34115b9d76cb98c6e9fb6
Parsing...
Parsed protocol: 0
Parsed host : livetest1-tracyone.channel.mediaservices.chinacloudapi.cn
Parsed app : live
RTMP_Connect1, ... connected, handshaking
RTMP_Connect1, handshake failed.
rtmp://livetest1-tracyone.channel.mediaservices.chinacloudapi.cn:1935/live/1ea048c582e34115b9d76cb98c6e9fb6: Unknown error occurred

Similar Messages

Live Streaming from Video File

Hi everyone,,, im newbie in programming streaming video with flash....ater reading documentations i tried the instructions on doc, i got a question that i've not been solved. How to make a "live streaming" from "video file". It likes a live streaming from camera but it uses video file as the source.... many thanks

Do you mean you have video file but you want to publish it as live stream , am i right? If yes following is the solution, but before that let me clarify you will need server to do it , i mean there is no way Flash client (i.e. .swf file running in Flash Player) can publish file from local disk.
Now let me tell you how to do it from server:
Place your video file under "streams" folder of your application i.e. say "myVideo.flv" under <FMS root>/applications/myApp/streams/_definst_
Now write following code in your main.asc file (this file would be under <FMS root>/applications/myApp )
var mystream;
application.onAppStart = function(){
mystream = Stream.get("livestream");
mystream.play("myVideo",0,-1,true);
This will publish "myVideo" file as live stream under name of "livestream". Client would have to subscriber using NetStream and
use ns.play("livestream",-1,-1)
You might have to write some extra code if you want to loop same file else once play of recorded file gets over , live publish will automatically stop.
let me know if this solves your problem

Disk Management for RTMP live streams

Hello,
Can anyone tell me if there is anything similar to DiskManagement functionality (available for HTTP streams) available to RTMP live streams in AMS?
I'm having trouble with INDEX files that are being created for the live+DVR streams (we are serving 24/7 streaming), even though I delete RAW DVR files which are old, INDEX files constantly grow and FMS crashes CPU after few days.
Any suggestions are appreciated.

Hi,
The methods that are mentioned as missing as expected by FMLE as it makes calls to them. They are defined in the sample live application that comes with the installation. The code for these can be picked up from the samples folder under the FMS installation from the application live. You can add these in all of your server side scripts to make them more relevant.
Thank you !

Problem of republishing remote rtmp live stream with AMF

Hi, Guys
I am trying recieving one live stream from a remote rtmp server and republishing this stream in AMF.
a remote rtmp (rtmp://192.168.1.100/live/remotestream)
AMF instance(codes in application/app/main.asc) recieve and republish to another AMF instance.
I can watch with url(rtmp://192.168.1.100/live/remotestream ) in a player (ex. OSMF player).
But,no data from remote rtmp streaming can be recieved in AMF.
Here is the log:
NetConnection.Connect.Success
Sending error message: Method not found (onBWDone).
mystream.onStatus: NetStream.Publish.Start
mystream.onStatus: NetStream.Play.Reset
My codes in application/app/main.asc
application.onAppStart = function()
          mystream = Stream.get("myvideo");
          myRemoteConn = new NetConnection();
          myRemoteConn.connect("rtmp://192.168.1.100/live");
          myRemoteConn.onStatus =function(info){
           trace(info.code);
           if(info.code == "NetConnection.Connect.Success"){
                mystream.play("remotestream", -1, -1, true, myRemoteConn);
                //mystream.play("sample", 0, -1, true);<-----if play a local flv video file ,it works.republish fine.
          nc = new NetConnection();
          nc.connect("rtmp://localhost/demo");
          ns = new NetStream(nc);
          ns.setBufferTime(2);
    mystream.onStatus = function(sinfo)
        trace("mystream.onStatus: "+sinfo.code);
        if(sinfo.code == "NetStream.Publish.Start")
                              attach_retVal = ns.attach(mystream);
            if(attach_retVal==true){
                    ns.publish(mystream.name,"live");

On the Macintosh side there is the Camtwist app which allows you to do this and it works smoothly with FMLE. On the PC side I´m not sure but there just got to be more apps capable of this than on the mac side.

IOS RTMP live stream audio

Hi guys hopefully some of you will be able to help me.
I am trying to stream audio from a live RTMP feed using the NetConnection and NetStream classes. I've managed to get my app running no problem on Android, however I am having some major difficulties getting it to play the audio back on iPad. Interestingly it works in the device emulators when debugging, however I'm assuming this is not really an accurate representation. I've tried streaming the RTMP in both AAC and MP3, but with no luck from either. I can verify through debug that it has connected to the stream, however I just get no audio playing.
Everything I've read about seems to suggest that this is possible on IOS as I'm only interested in audio and not video. Can anyone help?
Code sample below (it's quick and dirty! ).
THanks in advance!
<?xml version="1.0" encoding="utf-8"?>
<s:View xmlns:fx="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;
                              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;
                                        //Attach the video to the stage
                                        videoHolder = new UIComponent();
                                        videoHolder.addChild(vid);
                                        addEventListener(SecurityErrorEvent.SECURITY_ERROR, onSecurityError);
                                        grpVideo.addElement(videoHolder);
                                        connect();
                              public function onSecurityError(e:SecurityError):void
                                        trace("Security error: ");
                              public function connect():void
                                        nc = new NetConnection();
                                        nc.client = this;
                                        nc.addEventListener(NetStatusEvent.NET_STATUS, netStatusHandler);
                                        nc.objectEncoding = flash.net.ObjectEncoding.AMF0;
                                        nc.connect(defaultURL);
                              public function 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 - App shutdown");
                                                            break;
                                                  case "NetConnection.Connect.InvalidApp":
                                                            trace("audio - Connection invalid app");
                                                            break;
                                                  default:
                                                            trace("audio - " + e.info.code + "-" + e.info.description);
                                                            break;
                              public function createNS():void
                                        trace("Creating NetStream");
                                        ns=new NetStream(nc);
                                        //nc.call("FCSubscribe", null, "live_production"); // Only use this if your CDN requires it
                                        ns.addEventListener(NetStatusEvent.NET_STATUS, netStreamStatusHandler);
                                        vid.attachNetStream(ns);
                                        //Handle onMetaData and onCuePoint event callbacks: solution at http://tinyurl.com/mkadas
                                        //See another solution at http://www.adobe.com/devnet/flash/quickstart/metadata_cue_points/
                                        var infoClient:Object = new Object();
                                        infoClient.onMetaData = function oMD():void {};
                                        infoClient.onCuePoint = function oCP():void {};
                                        ns.client = infoClient;
                                        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 function 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 - Play start:");
                                                            break;
                                                  default:
                                                            trace("audio - " + e.info.code + "-" + e.info.description);
                                                            break;
                              public function monPlayback():void {
                                        // Print current buffer length
                                        trace("audio - Buffer length: " + ns.bufferLength);
                                        trace("audio - FPS: " + ns.currentFPS);
                                        trace("audio - Live delay: " + ns.liveDelay);
                              public function onBWDone():void {
                                        //Do nothing
                              public function onFCSubscribe(info:Object):void {
                                        // Do nothing. Prevents error if connecting to CDN.
                              public function onFCUnsubscribe(info:Object):void {
                                        // Do nothing. Prevents error if connecting to CDN.
                    ]]>
          </fx:Script>
          <s:Group id="grpVideo">
          </s:Group>
</s:View>

Just an update on this for anyone coming along after me.
I've managed to get this working on MP3 but not AAC (I guess AAC just isn't supported?).
My problem was the buffertime. The docs seemed to indicate it shoudl be set to 0 for live streaming, however switching it to "1" solved my problem on the particular stream I was pointing at.
So essentially justchanged the above line:
                                        ns.bufferTime = 0;
to
                                        ns.bufferTime = 1;
Would be great to find out if anyone has gotten AAC working however...

Tutor/live streaming surf video

I'm looking for a tutor in the Santa Cruz Ca or bay area. I want to be able to do live streaming video on the web. Film surfing in the day, edit it & show it at night on the web.
I'm currently taking a HTML class & have asked the professors at my college, but no one could help me.
The language & everything about this, is new & foriegn to me.
thank you pat

Yes, TeraGlobal Communications has live, streaming video using Java Media Framework. See http://www.teraglobal.com/ and download the TeraMedia demo for Windows.

RTMP Live Streaming Works, HTTP HLS/HDS Streaming Doesnt

I am running Flash Media Server on my Windows 2008 R2 server, and am trying to get HTTP live streaming working. I am using Flash Media Live Encoder to stream to the server. I can get a basic RTMP stream to work without issue, however anytime I setup to stream to a basic HTTP live stream, I get the generic "We are having problems with playback. We apologize for the inconvenience." in the basic Flash Media Playback player. I can confirm the stream is being published to the server in the Admin Console using the livepkgr application successfully, and I can even get an RTMP stream to work when it is streaming to the livepkgr application.
Here are the FMLE settings I am using to publish the stream:
Stream URL: rtmp://184.69.238.58/livepkgr
(yes, thats my IP, the stream is live continuously, test for yourself!)
Stream Name: livestream?adbe-live-event=liveevent
my embeded html for viewing the stream is as follows:
<object width="640" height="480"> <param name="movie" value="http://fpdownload.adobe.com/strobe/FlashMediaPlayback_101.swf"> </param> <param name="flashvars" value="src=http://<myipgoeshere>/livepkgr/liveevent/livestream.f4m"></param> <param name="allowFullScreen" value="true"></param> <param name="allowscriptaccess" value="always"></param> <embed src="http://fpdownload.adobe.com/strobe/FlashMediaPlayback_101.swf" type="application/x-shockwave-flash" allowscriptaccess="always" allowfullscreen="true" width="700" height="500" flashvars="src=http://<myipgoeshere>/livepkgr/liveevent/livestream.f4m">
While this gives me the error, the RTMP version (publishing to the livepkgr) works fine like this:
<object width="640" height="480"> <param name="movie" value="http://fpdownload.adobe.com/strobe/FlashMediaPlayback_101.swf"> </param> <param name="flashvars" value="src=rtmp://<myipgoeshere>/livepkgr/livestream"> </param> <param name="allowFullScreen" value="true"></param> <param name="allowscriptaccess" value="always"></param> <embed src="http://fpdownload.adobe.com/strobe/FlashMediaPlayback_101.swf" type="application/x-shockwave-flash" allowscriptaccess="always" allowfullscreen="true" width="700" height="500" flashvars="src=rtmp://<myipgoeshere>/livepkgr/livestream"> </embed> </object>
Can anyone explain to me what is happening here? I need HTTP streaming so I can eventually handle multi-bit rate streaming. Is there any further configuration beyond the standard "out of the box" FMS installation I need to consider on my server? There are no firewall issues at play here as all ports for standard streaming are open.

I had about 2 hours of downtime last night to move my server. Try again? It will be live all day.
Dustin Rogers
[email protected]
780.293.6632

Using netStream.time on rtmp live stream (broadcaster!=receiver)

I am trying to synchronize a live stream (which is
broadcasted from the flash player plugin) with some scripted
actions. The problem is, that the displayed frames of the rtmp
stream do not correlate to the netStream.time property on the
receiver side. In fact, i do not understand at all, how the
property is rendered on the receiver, since i can not recognize any
dependencies to e.g. bufferLength.
Does anybody now, how it is calculated? I tried with Red5 and
Wowza and they seem to behave similar. I guess, FMS would not make
any difference (if so: lease let me know!), since i assume that the
property is rendered during the encoding process i.e. by the
plugin.

Hello Jay,
thank you for your answer! NetStream.send() seems to be at
least a possibility to solve my problem with a workaround.
I just want to synchronize the time information of the up-
and downstream: both should have the same time information when
they show the same visual content (which i called "frame" in the
former post).
What i would suppose the netstream.time to be is that if i
shake my camera on upstream.time=10.0 then downstream.time should
equal 10.0 when this camera shaking is played back. This is the
behaviour that i am used to from streaming prerecorded FLVs. But
with live streams things work out diffferently.
In fact my downstream.time is bigger than upstream.time. And
i can not imagine how this can be. If i streamed up for let's say
20 seconds and i start the downstream after about 10 seconds, i
would expect my downstream.time to start with 10 seconds and then
increase continually. But against this, my downstream.time starts
with something above 20. How can this be? Or back to my initial
question: how is the downstream.time rendered?
This behaviour seems not to be dependent on the
downstream.bufferTime.
With netStream.send() i could send the upstream.time
information via the upstream to render an offset for
downstream.time on receiver side. This should work (have to check
it), but is a workaround, no "clean" solution.

Live Stream No Video

I am trying to watch a live stream on
www.mtv.com.lb
I am unable to get an image I only get sound.
I tried dowloading the lates flash player
and I also tried using google chrome but nothing seems to work.
can anyone suggest a solution please,
Thanks

Have you got notifications muted ? Only notifications (including games) get muted, so the Music/iPod and Videos apps, and headphones, still get sound.
Depending on what you've got Settings > General > Use Side Switch To set to (mute or rotation lock), then you can mute notifications by the switch on the right hand side of the iPad, or via the taskbar : double-click the home button; slide from the left; and it's the icon far left; press home again to exit the taskbar. The function that isn't on the side switch is set via the taskbar instead : http://support.apple.com/kb/HT4085

How to play rtmp live streaming through VideoElement??

Hi , i'm trying to play live streaming by using VideoElement
like this :
<s:VideoElement>
<s:StreamingVideoSource serverURI="rtmp://192.168.0.136/tv" live="true">
<s:StreamItem streamName="stream" />
</s:StreamingVideoSource>
</s:VideoElement>
but nothing occurred ........did i use the wrong syntax @@a or....?

Anyone?

Timecode/TimeStamp in RTMP live streaming using Adobe Media Server and FMLE

HI There,
Am trying to stream a video on to Adobe media server(using RTMP) through FMLE and play it on my web site using JWPlayer. i could stream the video successfully, but know i want to get the timestamp/timecode on the stream. does Adobe media server and FMLE support this kind of use-case ? if not is there any other way to achieve the same. any comments/suggestion/pointers appreciated .
Thanks in Advance
Regards
Deepak

If you're talking about nonstop, continuous streaming, your subscribing client will need to close and reconnect the stream every couple of days, as the server will run out of timestamps for the stream.

RTMP Live Stream does not resume with Spark VideoPlayer Component

Hi all
The Flex 4 framework brought along the VideoPlayer component, which supposedly can automatically choose the best stream in terms of bitrate from a list of available multi-bitrate streams, play live videos and everything.
See http://help.adobe.com/en_US/flex/using/WSc78f87379113c38b-669905c51221a3b97af-8000.html for some information about it.
Here's a working example for an MXML application under the Flex 4.1 framework:
<s:VideoPlayer width="100%" height="100%">
    <s:source>
        <s:DynamicStreamingVideoSource streamType="live" host="rtmp://88.87.56.214:1935/live/">
            <s:DynamicStreamingVideoItem streamName="euronews_tmo_h.stream" bitrate="19200"/>
            <s:DynamicStreamingVideoItem streamName="euronews_tmo_h.stream" bitrate="9000"/>
            <s:DynamicStreamingVideoItem streamName="euronews_tmo_h.stream" bitrate="3600"/>
        </s:DynamicStreamingVideoSource>
    </s:source>
</s:VideoPlayer>
The streams are valid and they play. However, I'm bumping into a problem when the stream is paused: it will not resume.
The player state alternates correctly between 'playing' and 'pause' when you keep clicking the play/pause button. However, once the first stream pause(or stop) occurs, the image and sound seem to freeze...forever.
Any help ore guidance are most appreciated!

I'm also facing the same problem.. Any developments in that ?

Live streaming audio/video

I'm looking for advice on how to stream live video, full
screen at high quality 640x480 30 fps. I need to be able to carry
on a teleconference between two instances of the program, but also
support 640x480 at 30 fps. What programs might be needed (encoder?,
server?)

Hi tony45371,
The replies from in2mobile are 100% correct.
For live video, some server is required. Flash Media Server
is the required server offered by Adobe (there are competitors
including open source competitors to Flash Media Server). Also an
alternative to putting up your own Flash Media Server, may CDNs
offer Flash Media Server services. You can find a list of CDN
partners here:
http://www.adobe.com/products/flashmediaserver/fmsp/
Are your live events webcasts (e.g. 1 presenter, w/ many
attendees)? Or 2-way (e.g. multiple presenters)?
I assume that you require 2-way since your post said "I need
to be able to carry on a teleconference between two instances of
the program".
Following below are links into the browsable version of the
product documentation (adobe.com/livedocs). Easier to read PDF
versions can be downloaded here:
Flash Media Server Developer Guide
http://livedocs.adobe.com/flashmediaserver/3.0/hpdocs/flashmediaserver_dev_guide.pdf
All Flash Media Server Documentation
http://www.adobe.com/support/documentation/en/flashmediaserver/
2-WAY VIDEO CONFERENCE
For 2-way you can find a related sample application (with
explanation and downloadable source code) here:
Video Conference with Flex & FMS
http://renaun.com/blog/2006/11/08/147/
Video Conference with Flex & FMS, Live Demo and Source
Code
http://renaun.com/blog/2006/10/28/139/
You can also find additional code examples in the Flash Media
Server product documentation here:
http://www.adobe.com/livedocs/flashmediaserver/3.0/hpdocs/00000082.html
(Above link, in the PDF version, is in Chapter 4: Developing
live video applications)
1-WAY WEBCASTS
For live webcast events, Flash Media Server includes a free
companion program "Flash Media Live Encoder 2.5" (aka FMLE). FMLE
offers higher quality video (H.264 & VP6), and audio (MP3 and
with a 3rd party plug-in AAC). Details on FMLE here:
http://www.adobe.com/products/flashmediaserver/flashmediaencoder/
Following is a link into the product documentation on using
FMLE 2.5 w/ FMS3:
Using the live service
http://www.adobe.com/livedocs/flashmediaserver/3.0/hpdocs/00000031.html
(Above link, in the PDF version, is in Chapter 2: Streaming
services)
You can also find an article written against the older Flash
Media Server 2 with an earlier version of FMLE here:
http://www.adobe.com/devnet/flashmediaserver/articles/webcasting_fme.html
Please post back regarding whether the above answers your
question, and if you find it helpful :-)
Best regards,
g

Live streaming, no video only audio

Hello
I have 3 pc: in the first I'm encoding a camera with Adobe
Flash Encoder 2.5 and send the signal to the second PC
in the second PC have installed Flash Media Server
in the third have a player.
Every thing is connected and in the player i can hear only
the audio, but not the video. Tryed with different players.
Any help
Sincerly
Saimir

Sounds to me like a h.264 issue. If you're using the h.264
codec in FME, you need to make sure you're using the latest version
of FMS3. H264 support was fixed in 3.0.2. You also need Flashplayer
9.0.115 or better.
Try switching the video codec in FME to VP6. If that makes
the problem go away, then you know it's an issue with either FMS or
your flashplayer being outdated.

AS3 - 1) Countdown Timer to the designated Date for Live-Broadcast-Event yet to take place; 2) Detect End of Live Stream on FLVPlayback with FLVPlayback.isLive = true - ActionScript 3 - flash cs3 cs4

Hi folks,
Ronny's here again on forums, having particularly 2 (two) questions/problems to resolve:
1) Countdown Timer to the designated Date for Live-Broadcast-Event yet to take place
2) Detect End of Live Stream on FLVPlayback with FLVPlayback.isLive = true
attached is the .zip file (as3_Countdown Timer_ver 1.0.1_by Ronny Depp.zip) with all flash source files containing:
a) The FLash Source (file: timer_module.fla) - (FLA flash source file - Flash CS3 Professional, Flash Player 9, actionscript 3.0)
b) com.othenticmedia.utils.dateAndTimeManagement package including 2 .as actionscript 3.0 Class files.
       i) com.othenticmedia.utils.dateAndTimeManagement.DateAndTimeManager Class in the said package. (file: DateAndTimeManager.as)
       ii) com.othenticmedia.utils.dateAndTimeManagement.CountdownTimer Class in the package. (file: CountdownTimer.as)
c) The compiled SWF file version of this Application's blueprint. (file: timer_module.swf).
What i need to confirm is: ........................................................ see the next post of mine. (for Problems need to be Resolved)

Problems to Resolve:
Problem#1) - Countdown Timer to the designated Date for Live-Broadcast-Event yet to take place.
Problem#2) - Detect End of Live Stream on FLVPlayback with FLVPlayback.isLive = true;
Problem#1 Description:
I need to pinpoint the Logical TimeSync Exception, i am still unable to figure out. That is I'm using a webservice in my Application to Synchronize the Time with the actual ET (eastern time) with accomodation of auto-adjustment for EDT GMT-4 (eastern daylight time) & EST GMT-5 (eastern standard time), times. I am using the zipcode: "10012" to pass it to the Web Service in urlRequest object, to retrieve the Current ET eastern time according to EDT & EST time settings for Manhattan/Brooklyn areas or others within New York, NY 10012.
Currently the Web Service is returning accurate date/time based on local EDT GMT-4 daylight time.
Is there some defined set of dates for EDT & EST times for New York region that I can check for to ensure the correct Dates/Times for Eastern Time in New York area ??? I am using NY zipcodes because i am sure to get correct ET values.
The Major Problem Part: is I need to correct the time by 2 seconds or approx. 2 secs, some millisecs.
When I retrieve the Time Value from WebService, it lags behind for 2 seconds as compared to DateObj i create using computer's local time, on my Windows XP Service Pack 2 with Automatic Updates turned-on. And I'm sure about my Windows will be having latest updates for Time Management already installed. I also added the 2 secs. to the TimeSync(ed) Date to make correction to this Date obj.
I call my custom fucntion addSeconds(dateObj:Date, secs:int) to add 2 seconds to the Date by Converting Seconds to Milliseconds.
Please comb through the as code in files attached and Help Me Out !!!
Problem#2 Description:
Secondly I need to Detect the End of Stream state while using FLVPlayback component, an rtmp:// live Stream from FLASH MEDIA SERVER.
I need to Play a YuMe Post-Roll Ad when Steam Finishes/Ends.
Live Broadcast Stream Event starts every night on Wednesdays & Saturdays exactly at 10:59 PM EDT GMT-4.
Live Events only Streams/Broadcasts the stream for 50secs. exactly. When [playback stopped] it plays a PostRoll Ad and after the CountdownTimer again comes back to life. The Next upcoming Event is calculated & the Countdown begins until Next Event's time/date is reached.
Here is the code on the frame 1 on the MainTimeline: (rest of the params like source, volume, skinAutoHide are Set using Property Inspector for FLVPlayback instance on Stage)
//myStream instance of FLVPlayback is on the Stage
myStream.isLive = true;// Frame 1 Actions in the FLA
myStream.addEventListener(VideoEvent.COMPLETE, onEndOfStream);
myStream.addEventListener(VideoEvent.STATE_CHANGE, onState);
myStream.addEventListener(String(VideoError.NO_CONNECTION), onStreamError);
myStream.addEventListener(NetStatusEvent.NET_STATUS, netStatusHandler);
/*if(myStream.stopped){
     trace("tracy: "+myStream.state);
} else if(myStream.state == VideoState.STOPPED){
     trace("tracy: "+myStream.state);
function onStreamError(event:VideoError) {
     trace(event.code + "\n\t" + event);
function onState(event:VideoEvent) {
     trace(event.state + "\n\t" + event.toString());
function onEndOfStream(event:VideoEvent) {
     trace(event.state + "\n\t" + event.toString());
function netStatusHandler(event:NetStatusEvent):void {
     switch (event.info.code) {
          case "NetConnection.Connect.Success":
               //connectStream();
               break;
          case "NetStream.Play.StreamNotFound":
               trace("Stream not found: "/* + myStream.source*/);
               break;

Rtmp live streaming of videos!

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