I'm streaming mp3 on flash media server 3.5 on linux version, and very often come accross with that problem, when the buffer.length become 0 on client side, the streaming will stop with NetStream.Play.Stop event, and will not be continued. I think this case it should wait always till the data arrives and never should stopping. Some more info : I tested on applications/live folder, and the mp3 files are created dynamically during the playing, and the creation speed is almost the same like the playing speed, so those are not fix files.
i have application that streams videos from fms .i try to play video of 240 seconds, but when playback reaches 180 seconds i got NetStream. play. stop . it happens regularly at the same time.why does it happen? is it encoding of the video??
I am experiencing an issue with playback on RTMPE streams. after investigation it seems that the FMS server is firing the NetStream.Play.Complete message at random points, indicating that a stream has ended. This is happening and random points during the stream, not even close to the end.[code]As you can see roughly 17mins into playback...although the stream is 1 hour 24mins long.I have tested this numerous times, and each time it is at a different point in the stream.Intermittently the NetStream.Play.InsufficientBW warning is being fired prior to NetStream.Play.Stop.I am using a player built on OSMF 1.5
I got a bunch of live stream from FMLE, say: "FMLE_channel1", "FMLE_channel2", "FMLE_channel3". And then on the server side, I created several corresponding republished stream called "channel1", "channel2", "channel3".
On periodical basis, we call Stream.get("channel1").play("FMLE_channel1", -1, 10, true) every 10 seconds. Similar things were done on the second channel & third channel.Soon after the above Stream.get("channel1").play() call, I should get the following events in sequence:info :NetStream.Unpublish.Successinfo :NetStream.Publish.Startinfo :NetStream.Play.Resetinfo :NetStream.Play.Start In the above case all are happy. Clients can view channel1, channel2, channel3 well.But then after a while, one of the three channels, in most case it would be channel1, will not be viewable.
With the server trace info, I found that after the Stream.get("channel1").play() call, only the following two events exists:
info :NetStream.Unpublish.Successinfo :NetStream.Publish.Starti.e. I was missing the play.reset and play.start event.I further checked and confirmed that the FMLE was publishing all three channels fine to the server. I was able to view the "FMLE_channel1" from flash clients, but not the republished "channel1". the version is FMS 3.5.0.
I am streaming pre recorded audio files (mp4) to an AIR client. I have tried two different solutions, streaming the file directly with the NetStream.play("mp4:xxxxx.m4a") and creating a server side playlist, adding the same sound clip and then streaming the playlist. The problem is, when streaming the playlist i get a few NetStream.Play.InsufficientBW, this does not happen when streaming the file direct. Both solutions uses bufferTime=1.0
I would like to use a server side playlist to implement a simple key solution so that the client dont know the full path to the file, but instead sends a key to a custom server side function that looks up the file path and creates a stream for the client.
I'm recording audio using FMIS, the file is being saved out as a FLV file and I am able to play it back. Eventually that file may be deleted after some time and I would like to notify the user if the file is no longer available. I was expecting to see NetStream.Play.StreamNotFound after deleting the file on the server but I'm not getting that message. Instead I get:
--> NetStream.Play.Reset--> NetStream.Play.StartAnd nothing plays which is expected since the file is deleted, but I dont get the right message. I'm calling the file like so:ns.play ("flv:file_to_stream");
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?
i've created a sub-folder called dvr under fms 4.0 applications. i made a test of playing a video named sample.flv then at the same time i publish it to the same fms 4.0 server with a different video name: "mergedVideo".the asc at the fms server as following:
NetStream.play(streamName, -1); This seems to be working wrong.if I have recorded an flv on server using FMS and FMLE with only audio with name "myaudio" and then after if I try to play a live stream using NetStream.play("myaudio", -1) then it plays the recorded stream. I believe that documentation says that it should start a live stream instead of playing recorded stream as the second argument is -1. Is this a bug in NetStream.play method?
dvrcast is a live service.mp4: as the service record and I look over to the dvr sample player debug output messages while out NetStream.Buffer.Empty will stop.Screenshot is shown below. Is it possible to do without stopping the service?
I have a series of mp4 videos (H.236 @ 22 fps) streaming from an akamai FMS serve via RTMP protocol in flash player 9+ with AS 3.My goal is to create client side playlists that smoothly switch from one stream to another. I am creating these playlists using a series of netStream. play (filename,start,len) methods. If I don't use an offset for the start parameter, then the stream switches smoothly from one video to the next with not noticeable jump or jerkiness. However if I introduce a start offset, say a few seconds in, I start to see a quick little hiccup or pause between the seams as it switches from one playlist stream to the next.
We've upgraded our production servers to 3.5.3 on Windows 2008, and now I'm seeing a lot of "NetStream.Play.InsufficientBW" statuses in our client app's log The server is streaming a very low bandwidth live video/audio stream, and the client has more than plenty of bandwidth to play it The stream doesn't exhibit any audio drops or skipped frames, and there are no message drops on the server side We have had clients complaining of random disconnects. Does anyone know why I might be getting this message
I'm using OSMF v.95 to handle my video playback. When my streaming videos finish, the following error is triggered:"NetStream.Play.StreamNotFound"I want to be able to re-play the videoor scrub back, but the stream is unusable after that error.I've tested against a few different streams - here is one:
I am working on the Audio-Video application having two Flash Media Server, One is Flash Media Interactive Server and another is Flash Media Streaming Server.I am publishing stream on Flash Media Streaming Server and receiving stream from Flash Media Streaming Server. Flash Media Interactive Server publish stream to live folder "rtmp://xxx.xxx.xx.xx/live" Flash Media Streaming Server from server side code.when application.onPublish of Interactive server dispatch then the stream published by interactive server has been republished to live of the streaming server and from there client receive stream by making connection with live folder of Streaming server. Stream from streaming server received successfully but
I have recently installed FMIS 3.5.3. In checking the access logs I find data in both logs that display the same stream stop and stream play time .I'm not sure why the time is the same (00:19:27 example below). Videos play fine when testing from work (T3 connection). However, occasionally a very slight hesitation when playing video from home (I have cable connection). [code]...
I'm creating a video chat app using Flash Media Server 4 and Flex, using RTMFP for peer-to-peer.
No matter what I try, I cannot mute a user's NetStream. I tried receiveAudio(false) and that does not work.
I tried capturing the stream SoundTransform and setting the volume of that to 0, that does not work.
I tried setting the mx VideoDisplay volume to 0 and that does not work.
I can set the alpha of the VideoDisplay, so its really strange that I cannot set the volume.
BTW, I am not trying to mute the mic of the user who launched the app, that I can do. I'm trying to mute one of the other users stream in the Flex app, that way the app makes it possible to not hear people who are perhaps being offensive.
Is anyone having trouble with Chrome getting NetStream.Play.Stop early in Chrome? Seems to fire around the time the buffer has filled. Same code is and has been working fine in all other browsers for some time.
I'm using the following code to attempt to get netStream.time on the serverside.....i've tried various ways googled it but have come up short...what code do i need to use so i can track a streams progress from the server side?
myStream = Stream.get("home1"); setInterval(time,5000,myStream.time); function time(myStream){ trace(myStream); }
I have an FMIS 3.5 app where I am attempting to publish multiple streams to an edge server.I have found that if I loop through my list of streams and attempt to publish to the edge server (i.e. as fast as possible), the publishing will fail They succeed if I have some delay between the publish calls.
Video broadcast using VLC to flv file. I am using the NetConnection, NetStream and Video to play it in a flash. In Chrome and IE everything works fine, but in Firefox and Opera NetStream often dispatch event NET_STATUS with info.code = NetStream.Buffer.Flush and NetStream.Play.Stop and video is a slowdown.
I am developing a video player for FMS 3.0 and my client complains that when he leaves the player running overnight or after computer goes to sleep when he comes back the elapsed time displayed as NaN and when he tries to play, nothing happens. Why is this happening? Does it mean that the connection with the FMS dropped and I have to reconnect? How can test for the dropped connection?
I'm working with a custom made Flash video player using FMS (Wowza, actually).The video is working, yet when I access the BytesLoaded and BytesTotal,I get 0 (zero) for each.This even persists as the video is loading and playing.It's as if it is not even being registered at all.Here's the code:
I have application that uses the send functionally in the NetStream on the server-side. When a connect to the app using rtmp I am able to see the send come thru but if the app uses rtmfp it does notI gone thru all the examples in setting up a mutlicast app and I know the app works because when I publish a video out everyone in the group sees the video. The only thing not working is send. I've also set dataReliable to true.
I'm trying to get the metaData as usual with NetStream object from FMS but in RTMFP protocol but doesn't work.However, with RTMP my code (which is coming from onMetaData example in livedocs)it works.
Is it need any special metchod/properties to get the metadata with rtmfp ?
I'm trying to send metaData from client to FMS server when recording Webcam video and have implemented sending of metaData as stated on Adobe's help page at: http:[url]....
However when I play the recorded video from FMS, my custom metaData properties trace "undefined".I can trace default metaData properties such as duration or videocodecid successfully,but not my custom properties such customProp,width or height. Here is part of my code that is related to the issue: private function
