Media Server :: Unable To Play HTTP Sample,RTMP File Is Not Working?
Aug 12, 2009
i have installed flash media server 3.5 ..windows vista and using FLASH 10right now in Flash Media Server page I am only able to play HTTP sample,RTMP file is not working , i check the sample folder >> and open HelloWorld >> that shows error :: Error #2044: Unhandled NetStatusEvent:. level=error, code=NetConnection.Connect.Failedat HelloWorld/connectHandler()what should i do now need ur help urgent
if I repeat this IIS topic, I couldn't find the answer to my problem any where on the internet. I installed FMS Dev 3.5 on Win 2003 Server with IIS 6.0 enabled. I don't have any issue with port 80 Listening, I used the IP address 192.168.0.21 for my web application (IIS) and 192.168.0.22 for FMS (I only have one network card and port 1935 is open under firewall). I can play the sample videos (RTMP, HTTP, and Dynamic Stream) using the Flash Media Start Screen (or from the location C: Program FilesAdobe Flash Media Server 3.5webrootindex.html) without any problem.
I then modified the IIS Default Website to look at the "webroot" folder (C: Program FilesAdobe Flash Media Server 3.5webroot). From IE, I can access the default web site by enter http://192.168.0.21/index.html. The website loads up correctly, and the RTMP video is playing perfectly. However, if I click Play Video (HTTP) or Dynamic Stream (tab), I receive "Connection Error. Please press Play to try again." I look at the log file (access.01.log) and see the error log "Session disconnect
I am not able to play video on mobile device which is .3gp container and H.263 / AMR_NB encoded. I just want to play my website videos in mobile device also just like[url]....I want to use RTMP and HTTP both. My requirement is as follows - Which codec and container will be best? Should I use FLV to play video on mobile device? RTSP required or can be use RTMP? Is NetStream and NetConnection methods different from Flash Player in Flash Lite Player? How to play 3gp video using RTMP stream ie.ns.play(mp4:mobilevideo.3gp¯, 0, -1, true) is it ok or any thing else required? For mobile browser and computer browser,can I use single player or I have to make different player for computer browser and mobile browser? It would be better if I can do it with single player for both mobile and computer browser. Sample code required for testing. If you can.I got below article in which they mention that we can play video 3gp container in mobile also.
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 had FMS 3.5 developer package installed on my server installed about a month ago. I used to be able to go to the /webroot/ page to see the sample RTMP video play and it worked great. Today, I go to view that page and find that it no longer plays. I get "Connection Error. Please preee Play to try again."
Pressing the play button does nothing, so I click the RTMP thumbnail and it reloads ony to give me that same error again. I can click the HTTP thumbnail and see the video play just fine. But, I want RTMP and after over an hour on the phone with my managed hosting tech, we can't find anything wrong.
Here's the thing: I haven't done anything, I have not changed anything, I have not manipulated the server software whatsoever. It just stopped working and gives me "Connection Error."
i am new on steaming & flash server; when we try to use RTMP over HTTP the outside client gets the internal IP address of the FMS server instead of the NAT one or public IP address, how can we solve this.
I have read some forums posted in here and also searched the web extensively but cannot find a clear answer or get HTTP tunneling to work with Flash Media Server 3.
Q1: Can Flash Media Server 3 be configured for RTMP and HTTP tunneling to work? The reason I need to know if this will work is due to more and more clients are reporting that videos are not playing for them and I have determined that these clients are sitting behind a firewall that has port 1935 blocked. So I would like to configure the FLV playback control to try to stream the file over RTMP and if that does not work, use HTTP.
Here is my asctionscript that I have tried to get this to work, but it does not. Some other notes are, the .FLV files live in this folder: D:AdobeFlash Media Server applicationsvodmedia The videos are recorded and then converted into .FLV files and loaded into this folder. The ultimate solution is I use a FLVPlayback control and pass the location and .FLV file name on the query string "Details.aspx?VIDEO=rtmp://216.203.12.15/vod/flv". I have pasted the object code * to show this example below the asctionscript.
I'm setting up my fmis to deliver video through rtmp and http.I'm on a locked down network.What ports need to be open to allow people outside of the network to access the rtmp/http streams? I am correct to say only port 80 and 1935?
I am connecting FMS over RTMP from a firewalled network where RTMP is blocked.Connecting to my FMS server from there takes (a too long) one minute before the Flash client switches from RTMP to encapsulated RTMP into HTTP.Is there any way to by pass this delay or is it a plug-in inner behavior that cannot be short-cut ?
I can get which part of file in bytes is loaded using videoDisplay component for RTMP protocol for VOD, I can get the current playing position using videoDisplay.playheadTime. But I want to know how many seconds of video are already loaded (not the length of bufferTime, which remains constant). i used videoDisplay.bytesLoaded when using RTMP it returns nothing ,if we uses HTTP it displays number of bytes loaded
The loaded size in bytes is not directly proportional to running time of the video, and while using rtmp im unable to get bytesLoaded too, how i can calculated the Video already loaded.
My swf is playing one video in using NetConnection, NetStream and Video object. If I want to stream one more video simultaneously in the same swf I have a few problems. It works when I create more NetConnection, NetStream and Video objects but is that necessary? The code rapidly becomes complex to handle.
Is there an easier way like perhaps share the NetConnection, or something (same FMS server)?
Question 2
The two videos on the stage are suppose to have different size, placement etc. Still the last one created inherit the properties of the first video display. It also starts playing for an annoying couple of seconds before the first one. How can I avoid that (inherit and delay)?
var ns1:NetStream; var ns2:NetStream; var nc1:NetConnection = new NetConnection();
I installed the FMS 4.5 on a debian system already running an apache web server. I configured the httpd.conf as mentioned in d6093a7e2f8312a374a1bde-8000.html but when i call a streaming url e.g. http://<IP>/hls-live/livepkgr/_definst_/liveevent/livestream.m3u8 the request is still handled by the old apache, not by the FMS
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 use video player from the flash media server and it will playback url.. as this originates from Flash media encoder then i have to enter this on the videoplayer but when i replace localhost with my IP address i get no joy at all. Also i can run url...ok from videoplayer in the media server but it allows the user to use http so how is this achieve as i cannot get it to work at all. tells me there is no connection. also this i try using local host and my IP address as well but no joy.
Same file streamed from FMS4 (same on 3.5) with rtmp protocol has very poor quality compared to real file quality. Is there some low level configuration to do on Flash Server? Is the streaming server making some kind of transcoding before sending the stream? Or it can be Flash Player? Tried unchecking "hardware acceleration" but nothing changes. I am on a local gigabit network, so no network bottlenecks. Video seems very pixelated on the edges (not soft/antialiased). These are the file details:
I'm using FMS Streaming 3.5.0 r405 on linux servers to stream videos with RTMP/RTMPT. We have decided to use the new proxying function to redirect HTTP requests to a Web server. We did not install the included Apache server, but used a lighttpd server installed on the same machines and configured to listen on port 81. So when a client connects with our player, it tries first RTMP, RTMPT and if it times out on these attemps, we try to go through with HTTP. FMS proxies the HTTP request received on port 80 to the lighttpd server on port 81.
My fms.ini file contains: # Whether to start and stop the included HTTP server along# with FMS.#SERVER.HTTPD_ENABLED = false
# IP (address and) port that Flash Media Server should proxy# unknown HTTP requests to. Leave empty to disable proxying.# With no address, specifies a localhost port.# For example:# HTTPPROXY.HOST = webfarm.example.com:80#HTTPPROXY.HOST = :81
This works well, but after a few hours, the HTTP proxy of FMS does not work anymore. Lighttpd is still responding on port 81. There is no error in all log files. Just stops working! I have to restart FMS to enable again the tunnel. This behavior happens on all my streamers.
I have to a problem using the Flash Media Interactive Server Feature Explorer. I want use the sample: RecordStream. I can see the instance "RecordStream" in console FMS 3.5. and show me the video in app AIR, but does not save the .FLV in my server.
We are using influxis fms connect, we need to upload videos and audios using php script, i dont know what to do exactly with this fms connecti need the answer for the following query, so that i can get into it, How to upload audio or video files to rtmp server using php script, since my php scripts is in one server and the fms is in another rtmp server?
I want to use RTMP based HTTP DVR functionality and HLS based IOS functionality, So I need to know how to MPP from my existing DVR app to the livepkgr app so that the stream being recorded at the DVR app can be used by the Flash and Stream MPP over to livepkgr app and can be used by the IOS HLS.
I'm trying to make a software which sends video and audio data to a flash media server by using RTMP protocol. Currently, my program can communicate with a flash media server correctly. RTMP specifications does not describe about the raw data in video/audio messages, so I muxed raw H.264 and AAC data into video/audio messages and sent to the server. The server seems to accept them, but a video player cannot playback the stream sending from the server. The player just says "Loading..." For a test purpose, I sniffed the network packets between Wirecast and the flash media server and ripped off only video and audio data. Then, I muxed those data into video/audio message and sent to the flash media server. In this case, the video player connected to the server can playback the stream correctly.
I checked the stream sent from Wirecast, the stream seems not to be H.264 raw data because those data are not started from 0x17 instead of H.264 start code. With those situation, I am wondering what kind of container format I should use for H.264/AAC data to the flash media server.
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.
We had FMS2 installed before and the paths to all our videos are like rtmp://ServerName/sites/.... (the default path on FMS2) Now we upgrade to FMS4 and we would like to keep these paths the same because we have many HTMLs that reference these videos. However, the default path on FMS4 is rtmp://ServerName/vod/... Is there a way to change "vod" to "sites"?
I tried to change VOD_COMMON_DIR in fms.ini from /install_dir/webroot/vod to /install_dir/webroot/sites, and also changed the document root in httpd.conf, but rtmp://ServerName/sites/ is still not working.
where is the source code for the multicastplayer sample that is located in <installed directory>AdobeFlash Media Server 4 oolsmulticastmulticastplayer directory? The StrobeMediaPlayer does not use the same html code and does not playback a rtmfp multicast stream.This is the code used in the multicast sample player to pass the manifest.f4m file into the player to playback a Multicast Stream which is not used in the StrobeMedia player sample.
I've placed some of my videos flv and f4v in the vod/media and webroot/vod folders with sample files provided. My files were checked with flvcheck and passed. When I load the embed code into a Dreamweaver HTML file and save the page to the webroot folder as instructed, the provided sample files play in my Firefox browser. However, when I edit the code to insert my files in strict accordance with the instructions concerning codecID and extension, I get the following message on the Browser "We are unable to connect to the content you've requested." My videos are less than 30 minutes in length and were encoded H.264 using Adobe Media Encoder and as I said passed the flvcheck.
I would like to have the possibility to detect when NetStream is not working (i.e. the name used in initialization can't be found on the server side). The StreamNotFound doesn't work at all. I've read that it's the fault of the Flash Media Server, which automaticaly creates new stream, if it cannot be found (of course it's an empty stream - what is wrong from my point of view). Is it true? If yes, can I disable it on the server side, so I could easily detect if the stream name is correct?
As titled, what is the way to record video/audio files using Flash Meida Server through rmtp, and allow users to access the recorded files through http?What I am trying to do, is to record a user's microphone's input and save it to the server.fterwards, I would like other users to be able to access the recorded files and mainuplating the audio data, by computeSpectrum(), to do some visualization of the audio. As I know computeSpectrum() cannot work on streaming files, so I think I need to access the recorded files using http instead of rmtp. Is that true?