Flash RTMP Streaming With No Filename - How To Connect
Mar 9, 2010
I am building a flash media player for an live audio stream. The server is set up to stream from a url that looks like so: [URL]. The instructions I have found say to first set the directory:
nc.connect("rtmp://ourwowzaserver.net:80/ourfiledirectory");
And then later play it:
nc.play(mp3:thefilename)
This works when we have discrete filenames on the server (thefilename), but the live stream is not a file - rtmp://ourwowzaserver.net:80/shoutcast. How in the heck do I get .play to play it?
Writing RTMP Streaming Server for streaming AVC+AAC video. And it works fine with rtmpdump. But I can't force it to work in flowplayer and other flash video players.The message sequence after handshake is similar to FMS / RED5 / erlyvideo / haxevideo servers: I've tried a lot of variations.
From Chrome debug console I can see, what all negotiating messages passed to the flowplayer. The last one is onMetaData. And after this the working sample (rtmp://flash.tvwmedia.net/LiveVideo//Live300) gets NetStream.Buffer.Full. And streaming from my server don't get it.
I'm starting with AVC Header message, containing sps/pps. After it first AVC picture passed. After - AAC header and AAC sample. And then AVC/AAC samples. This dumped OK by rtmpdump - I have working flv on exit. But flowplayer and others does not work.
I am trying to connect to an rtmp site using flash builder 4. I have created a class to handle the connection. This code works (traces a connection error) if connection.connect(null); but if i try to put an rtmp site in the connection, it doesn't do anything. I have it scripted that if the connection fails it should trace and error message but it just seems to stop. any thoughts?
We are attempting to transcode some video to a format suitable for our Flash Streaming Server. In the past we have used Adobe Media Encoder (AME) CS4, but we are working with *.VOB files this time, and AME CS4 does not support them. We are attempting to use Handbrake now, as it has great support for h.264 and MP4 files. Unfortunately while the files play wonderfully on our local machines, they do not stream from the Flash streaming server over RTMP. The files come out of Handbrake with a *.m4v extension. Our old files out of AME CS4 are *.MP4's with h.264 and AAC audio. I can see no differences in video or audio codecs which leads me to believe it may be a container format problem. We are attempting to use Handbrake because it does an excellent job of batch encoding files and produces good quality files.
Usually, flash streaming is done by capturing webcam video/audio and streaming using NetConnection and other objects to servers like FMS,Red5,Wowza etc.I haven't found any example on how to create your own stream of images and stream as a video to the server.I know it would be possible to convert the image to bytes and send via SharedObjects. Then decode on server and create a video file on the server (e.g. using ffmpeg), but I would rather do it in realtime on the client side if possible.
I've created a player for a client in the past using their LimeLight server to stream videos and not had an issue before, however for a new client using different LimeLight server, the videos seem to be ending 3-4 secs too early.
In the video players I check for NetStream.Play.Stop then put a 'reset' type function in there. This function however triggers too early due to this strange bug. Has anyone have seen this before?
private function netStatusHandler(event:NetStatusEvent):void { trace("connected is: " + nc.connected );
[Code].....
The only work-around I see for this is saving the initial number I get from the metadata duration, and running a timer to constantly check for when the current ns.time matches metadata and then run my reset function.
(1) An ec2 instance with an SWF on it - this SWF plays streaming video - i.e. is a video player like JWPlayer (2) A streaming video distribution set up via Cloudfront
If I stream the the video via RTMP from Cloudfront to the SWF (which is on ec2) - would I incur charges for data transfer into the server (i.e. for data being read by the SWF) and out of the server (i.e. for data being displayed by the SWF to the user) on account of streaming the video to users (assuming that data transfer into and out for the server is being charged for)?
how to embed rtmp link to be viewed in a browser? I already tried Moeya but it is still not running. Anyone knows other solution to how can I embed this to a browser?
I want users to be able to upload audio files to my server (say, over HTTP.. mechanism doesn't matter much), and then I'd like to assemble those files into a playlist and stream them out over a live RTMP stream to a Flash widget.The catch is: I'd like to begin streaming the audio files before they completely finish uploading. For example, if a user uploads an hour-long mp3, I'd like it to be able to begin playing on the RTMP stream once, say, the first minute has been uploaded.I'd like to support mp3 and AAC, and would hope to avoid re-encoding the files. Is is there a way to make this work with Wowza, Flash Media Server, or some other clever solution?
I'm building a Flash mp3 player to handle streaming mp3s served by Amazon CloudFront (which uses Flash Media Server). I'm using an FLVPlayback 2.5 component to do most of the heavy lifting (I tried an all AS3 method, but any stream automatically stopped within 5 - 10 seconds of playing). When I trace out the FLVPlayback's totalTime property, I get NaN. This is causing the seek bar to simply not work. This only seems to affect mp3s, as when I load an FLV I get a real value for totalTime. Does anyone know why this is happening
i had installed a FMS4 developper in server on my centOs. I can connect to my FMS with client with RTMP, but with RTMFP, FMS don't reply. I looked at FMS log and i saw edge. 00.log:
2010-09-30 13:11:53 4984 (e)2631504 RTMFP could not start on edge process for _defaultRoot__edge1. Famille d'adresses non support par le protocole -
I cannot connect to an FMS server from a client on a another domain using RTMP
Locally running a flash client that uses nc.connect(rtmp://...) to connect - works.
But when the client is in another domain I get NetConnection.Connect.Failed in the netconnection onstatus.
I've put crossdomain.xml files on my FMS folders that allow access from any domain (*). Also the allowedHTMLdomains.txt and allowedSWFdomains.txt are set to allow any - though it's not just the live application I can't connect to, it's also any other app.
There are no firewalls blocking on the server. Port 1935 is available. FMS is set to listen to 1935. The server is accessible with other protocols (http) - but with RTMP it only connects when I try to connect locally, as mentioned.
What other reasons or settings can cause the connection to FAIL?
1)I can use rtmp://localhost/vod/mp4:sample2_1000kbps.f4v , it works ok. But when I use rtmp://192.168.8.3/vod/mp4:sample2_1000kbps.f4v, I got Netconnect.connection.rejected. 192.168.8.3 is my fms server IP.
2)When I set fms use port 1935 and 80, rtmp://192.168.8.3/vod/mp4:sample2_1000kbps.f4v works ok on other pc. But using wireshark(Wireshark is a network protocol analyzer for Unix and Windows.) I found , the flash player connect 1935 port at first, it send "connect",the server replay "connection.rejected" and "no scope "vod" on this server". Than it connect to 80 port, so I see the video is not send throw 1935 port with rtmp, but throw 80 port with rtmpt. Why? How can I see video throw rtmp?
Currently, we are streaming using RTMP, but I want to switch to HLS to allow iOS devices to stream our video. I do not want to encode two separate streams though because we are operating with limited bandwidth on mobile data cards. If I stream using the livepkgr, the delay to iOS devices is about 12 seconds which is OK. However, on desktop clients, we have live scoring integrated with our video, so we cannot have a delay that long. My question is, if I connect to the same stream in a Flash client for the desktop, will the delay still be that long? Or will it stream in real time to flash clients with the delay only for iOS devices using HLS? Also, am I correct in assuming that I can do this on 1 stream, or do I need to setup a separate stream for RTMP and HLS?
I herd about RED5, but unfortunately I can't find any examples for feeding it with external source. I know RTSP is in development, but I'm thinking about (named) pipes or something like this. There is is also project named xuggle that is as far as I understand ffmpeg wrapper for Java, but I herd they have also problems with live streaming.
I am in a requirement where I need to calculate the bandwidth usage of rtmp streaming. Is there any way I can calculate the number of total bytes I have download during the rtmp streaming?
I am trying to publish a video to an RTMP Server but it doesn't publish. It might be a pre-release bug. I am able to play a NetStream but not able to publish one. Is there a sandbox issue? I'm not sure. Because, the FMS RTMP Server will not let a client connect unless it has been downloaded from the same host as the server (something like a sandbox condition)
How can I calculate the number of bytes download during RTMP streaming? So that I can keep track of bandwidth usage for particular user? I hope the question is simple enough to understand the requirements.
I have some questions regarding 4.5 and deciding which version we must license (FMIS or FMS). Obviously there is a huge price difference and as this is within a local network we don't require encryption or multicast. Here is the scenerio we need to stream live DVR to mac, windows browser AND iPhone/iPad and devices running Android 2.2(Froyo)+. I need someone from Adobe to help us decide.
1. I see DVR is now listed as a feature in Flash media server page. Does that mean its no longer exclusive to Flash Media Interactive server. For both http streaming and RTMP? 2. Previously in order to stream live to Adobe dynamic streaming, Flash media interactive server was required because a live segmenter had to be used. Is this still the case? Or can we just launch adobe live media encoder (FLME) and stream to flash media server.3. Related to #2 can we switch between serving streams via RTMP or dynamic streaming without changing the format of the material on disk?4.I was at a webcast for 4.x and remember being told that the latency for http live dynamic streaming was 20 seconds because of the repackaging that had to take place. Is this still true? What is the latency for iOS playback from ingest to playback?5. For the iOS playback feature do the streams have to be in Adobe dynamic streaming format or can we just stream via FLME RTMP and trust it to do the right thing?6. Does the iOS playback support DVR?7. Does the Android plugin have good support for RTMP live streaming including DVR?
Am developing an application with Flex 4.5 & Flash Media Server. I need a visualization curresponding to the plaing track . It is possible with SoundMixer.computeSpectrum(bytes, true, 0) in case of progressive downloading. But not working with rtmp streaming. And also need audio wave curresponding to the track which is also working with progressive ownloading usinf Sound object.
I am trying to use BitmapData.draw() on a video object, but using the OSMF framework.My hosting service has set up the following so that I can access my rtmp video:
At a certain point in the app flow, a m4v video is streamed from a remote location via rtmp. This works well, except for two little problems:
1. In very rare cases, the video starts playing, but you see no images and only hear the audio. Seems like this happens more often with Chrome, but that is only a guess.
2. In the first frame or so of the video, you hear a noise in the audio. It really is very short, but still a nuisance. This chirp is not there when viewing the same video with VLC or QuickTime player. It IS there when viewing with the Windows Media Player.
Are there any known problems with codecs and the flash as3 video streaming (via NetStream/NetConnection)? The video is composed by using the ffmpeg library (one mess of a library, I can tell you...) to bring together a live-rendered video (using a 3D Engine) and a pre-created *.wav (which does not have the chirp).
I've freshly installed the FSM demo on a redhat linux box, and have everything working. From that start screen running on Local Host, I click the "Play video (HTTP)" and that video of a train shows up. Cool. But when I click the link above it ("Play Video (RTMP)") I get an error message: "Connection Error. press Play to try again." and no matter how many times I hit play, I get that same message. The Dynamic stream doesn't work either, and the Interactive sucessfully displays webcam feeds, but doesn't show the "Play Live Stream" button thing. Is there anything special you have to do to get the RTMP stuff working? Some special command or server you have to run?
Would firewalls intefere with things (I'm pretty sure there isn't one on the machine, but I'm flailing wildly here) or would permissions mess things up? I'm completely lost ^_^;; I guess I should also add that there doesn't seem to be any log files. I'm looking under the server install directory, and there isn't even a "log" folder. There isn't one under Apache, either. It confuses me. So far all I can find on the internet is instructions to look at the log files...but if they aren't there... Am I just looking in the wrong places, or are they just not being generated yet? I did a tcpdump with wireshark, and the web app IS pinging port 1935 (for RTMP), but the packets are failling the checksum and are refusing to be reassembled because of that. Is this making sense to ANYBODY?
I am developing an application in Flash that runs locally and it uses FMS 4.01, locally. I have been using adobe FMS 4.01 for months with no problem. Today I cannot connect to my server and I cannot even play the sample video on the Flash Media Server Start Screen. The sample video for the http plays after I changed the permissions for flash, but rtmp does not play. I have reinstalled the server; 3.5, 4.0 and 4.01, none of them will connect to rtmp. I receive this error on the flash media server start screen that says "the connection timed out".
I have some .mov files want to stream to Flash media server by ffmpeg. i have already tried to stream a single .mov by FFMPEG command in terminal and it works, the FMS can display the thing i streaming in live.Now i want to stream multiple files as one source, i tried to use above command one by one, but it seems Flash media server stop the streaming when file1 is finished, then start the stream with file2.It makes the stream player stopped when file1 is finish, and i have to refresh the web page in order to continue on file2.i am calling the FFMPEG command by a C program in linux, i wonder is there any method that i can prevent the FMS stopped when i switch the file source in FFMPEG? or is that possible to let FFMPEG constantly deliver the stream by multiple files source without stopped when a file finish
I'm running Flash Media Streaming Server and have only been serving VOD up until now. I had my network administrator open up port 1935 to the outside world during the setup process and now I can't remember if that was actually required for streaming VOD to clients. Most documentation I've read says that this port should be open, but I seem to recall reading something at one point that suggested it wasn't necessary.
I've just started messing around with publishing live streams using Flash Media Live Encoder to the Flash Media Streaming Server. I have that working without issue but was surprised to find that no authentication is required before a client running the live encoder can publish a stream to the Flash Media Streaming Server. An authentication module is available however it only works with Flash Media Interactive Server and Flash Media Development Server.
If I leave port 1935 open to the outside world, there would be nothing to stop anybody anywhere from streaming video via my server. Anyone else running a default install of Flash Media Streaming Server and with port 1935 open to the outside should see that this is true of their setup as well. I'm wondering if I can safely close port 1935 without limiting the functionality of the server or if there's some way I can require authentication prior to publishing a live stream even though I'm not on the four-and-a-half-times-more-expensive edition of the product.
