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 !

Similar Messages

MacMinis + SL Server for Intranet live streaming ?

Hi,
is this asking for trouble (not enough horse power) or can the MacMini be used for a live stream into our intranet?
The documentation on this Apple site actually suggests a Macbook (for the Broadcaster) and a MacPro (as streaming server).
We will need to do this only, say, three times per year for 45 Minutes.
Signal will come via Firewire 400 from the camera.
Will two MacMinis be able to handle the load?
How many clients, do you think, can connect at the same time before we run into trouble?
Thanks for any friendly advice,
Barbara

There's a lot to consider when dealing with streaming.
If you're only streaming to the LAN then you might be able to use multicast, in which case this is a no-brainer - the server only sends out a single video stream to which all the clients subscribe (the heavy work is done by your network switches).
If, on the other hand, you can't use multicast, or if your 'intranet' really consists of multiple sites/locations then you need to consider bandwidth since that's likely to be your major bottleneck.
It really doesn't take much CPU to stream a connection (the hard work is in the encoding which is why Apple recommend a separate machine for encoding which passes the stream to another machine for distribution). I don't know what the hard limits are on a Mac Mini as the streamer but I wouldn't be surprised to see at least 100 clients on a reasonably-high quality stream. If you find it can't handle that (or you have way more clients) then QuickTime Streaming makes it easy to add additional streaming servers so all you need to do is balance your clients across the various streaming servers, making it easy to grow incrementally once you work out what your load is.

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...

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

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

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.

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.

Authentication for pushing live streams to FMS3

Hi there, I have setup FMS3 and use it to test my live
streaming. But I was wondering is it possible on the FMS to setup
an application for live stream, so that it checks for
authentication before accepting the live stream for broadcast? What
I mean is if I create a live application on FMS, anyone who knows
this application name would be able to push the stream to FMS. Can
FMS control this with a username and password? Thank you.

The video clip at NAB is a sneakpeak into the upcoming release of the FMS which supports streaming to iPad and iPhone, and its coming up very soon.
Streaming to android and other devices is already supported on the released version of FMS4 technically, so you can give that a try, and with this upcoming release, you can be sure that all the mobile devices would run high performing videos.
i know i have not answered what you asked for, exactly, but be assured it will be very soon. We hope to meet your expecations.
Thank you !

Sound works at 2 locations for a live-streaming video but not on the third.

I am trying to watch live streaming video from a trial. Three television stations offer this video. I am able to see two of the stations, but there is no sound on the third. Since I am not always available to watch, I would like to watch the replays in the evening. These are only offered by one station--of course the one where I can't get the sound. I am able to watch the news videos from this station but not the trial ones. Is this a problem with the station or is it mine? I should add that I have tried in several browsers with the same result.
Sound and video work fine on everything else.

That's odd. It sounds like a problem with the stream but I tried that link out and the sound worked fine for both the commercial and the trial. I did notice that the trial's volume was low for the first 5 - 10 seconds, but I could clearly hear the proceedings once they started.
What browser / OS / version of Flash Player are you using?
Chris

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?

Flash Player Support for HTTP Live Streaming?

Does Flash Player 11 / Air for TV 3.0 support HTTP Live Streaming ?

If you need help with posting messages in the forum there is a test forum available at http://forums.adobe.com/community/general/forum_comments/testing

Only 1st RTMP bitrate for HTTP Live Streaming works

Using this on AWS with cloudfront/cloudformation.
When I created new Cloudformation, my first RTMP will works. i.e I publish at 500kbps.
Then if I stop and send single bitrate at say 800kbps, it will not works. i.e OSMF will keep buffering.
SSH to my EC2, I restart FMS server, also no different. When I republish at 500kbps, it works again.
Then, Create fresh new Cloudformation, and send 800kpbs this time, it works fine.
So my conclusion, I can only send 1 known bitrate, and cannot change to another, ( but can change resolution).
Must be something wrong here. What exactly happened?

Hi,
When you first publish with 500 Kbps and then superimpose with 800 Kbps, it may not allow you to do so. Since on republishing it appends to the previous streams and it might not work with streams of two different bitrates. My suggestion to you is - try publishing in record mode. This will clear old streams and should work like a dream.
To publish in record mode, use this publishing string;
livestream?adbe-live-event=liveevent&adbe-record-mode=record
Thanks,
Shiven

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 ?

We need to optimize FMS for smooth live streaming can anyone help?

So we are coming across the same problem over and over again. I called Adobe FMS tech support and they assisted us to achieve what we need, but we are having two main issues.
First, we need to allow live, clear, and smooth video broadcasting with FMS so we implemented dynamic bandwidth detection so the system can automatically detect a users specifications and adjust video accordingly. This has worked better for us but we are coming across two main issues; first, the video is now very pixilated, and second, we are still get a pause or delay in movement from the stream every 4 seconds or so.
We are looking to get similar video quality as seen on the justin.tv live broadcasting site. I know this video detection and quality is achievable, but not sure what values we need to put in or are missing?
Any help would be greatly appreciated.
Thanks,
Steve

I'm interested in doing something similar - bridging FMS to a
an Asterisk conference. I know breeze has this capability, does
anyone know if this can be done using FMS?

Player for dynamic live streaming

Hello. I run a FMS 3.5 Server and I do not have much knowledge about it. I want to use the dynamic streaming function that FME provides, but I do not have a player that supports the dynamic switching. I looked into ActionScript and Flex but it is all to complicated for me. Can somebody sent me a link where I can download such a player or can somebody give me help so I can figure this out? Thanks!

Thank you for the link but it does not solve my problem. In this thread they work with manifest xml files like a playlist. Which i tryed but also doesn´t work. Adobe declairs FMP with easy config and less code for non geeks. And this is exactly what i want for my project.
Does anybody can post some html sample code especially for dynamic streaming/ MBR wit Flash Media Playback??? For Live or On-Demand.
Best
Hinricht

Disk Management for RTMP live streams

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