Media Server :: Video Delay Continuously Increasing Latency?
Jun 3, 2010
I'v made an application with all the default codec and settings provided by adobe to publish live video using FMS. I've made two swfs one is for publishing live video and the other is for viewing the published stream (telecast). When I'm publishing the live video on My laptop using the default web Cam and mic attached to the Dell laptop then after 2- 3 minutes delay on the telecast side start increasing and it keep on increasing continuously .At the same time when I use my desktop PC by attaching external Web Cam and mic to publish, delay on my Telecast side is consistent ,it never increase. I found the problem is with the Laptop's mic. Can anybody please suggest me why it is happening, and is there any solution to it.
We have one functionality in which one user can communicate with the other user at the same time through flash media server3.5. In this fucnitonality we are catching stream from both end through camera and pass it to flash media server. On FMS side this streams gets stored also we are displaying those recorded flv files side by side to the corresponding user so that they can communicate with each other. In this situation we are facing a video and audio latency upto 1.5 to 2 sec.
I have made a sample application for publishing video through FMS using as3.In that I had used the same code as given in the sample provided by the adobe,
m_nc = new NetConnection("rtmpt://localhost:1935/Test"); private function onNetStatus(event:NetStatusEvent):void { switch (event.info.code) case "NetConnection.Connect.
I need to increase my buffer size in FMS. My video is running from RTMP and I faces the problem to load. Video stalled between the running. So I need to increase buffer size for running video smoothly.
I've a site where consumer can take a live meeting with the beauty consultant. The problem I'm facing is during the video chat. Basically there is a delay of 5 seconds between Video and Audio when I access the site from out of my network but when I access it within my network (VPN) then it's work fine.
I've build a simple video chat application. The problem is that there is a delay of seconds between video display i.e. user1 video motion change displayed bit late at the user2 window. I'm using FMS 3.5 and FLEX.. IS THIS PROBLEM RELATED TO THE BANDWIDTH. My FMS bandwidth is 256kbps.
I can't find the setttings for have a least latency as possible, either the video quality is bad or the latency is up than 5 secondes...What should i do?
How can I play flv files continuously..?? I have developed code for playing flv files in my browser. Those flv files are streaming from flash media server. I can play files individually, but I want to add one button to play multiple flv file continuously when one button clicked. How can I play those multiple flv files continuously...??
iam streaming my web cam to FMS .... i have got a latency of 1 -2 second,,,,,, What should i do to reduce the latency,,,,,, is ter any change i can make in FMS files(xml files) in order to reduce the latency......
We developed one video chat application in which one user can chat communicate with anohter user. Latency is coming on one side in peer to peer connection.
Apparently I seem to have upset one of the forum members by cluttering up his space with rantings about FMS not being up to speed or scratch or whatever. So i'll keep it simple. Since i'm not inexperienced in the deployment of Flash via a streaming provider, I just want to know if other users here, find the FMS software to be brilliant in concept and delivery. In other words are the streaming companies going to be worried that my 1000mb of date transfer per month are going to stream just as well as the $360 a month they would like to charge me for doing the same job?!
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.
what the best way would be to add a delay of about 120s. This is for a videogame shout casting company and in order to prevent cheating we need to make sure that the stream is at least delayed 120s.
I have an application that uses FMS 3.5 to stream live audio between 2 computers.Currently, the delay between when the person speaks and the time the other person hears it is about 2 - 3 seconds.This makes it difficult to have a conversation because one person regularly steps on the other.I am using a NetStream object that connects to the fms server.I then attach audio.What configurations can be set so that the least amount of delay is experienced between the two computers? Can fms ever relay audio as quick as something like Google Talk?
when i try to live stream with FMS! I can stream video with Flash media live encoder to the server but when i create the player to recieve the livestream from server,i can not recieve the live stream,can anyone give me a step by step tutorial of how to do it?
I'm doing some tests with FMS and live streaming, but I observe quite long delays. When the stream is first accessed, the latency is around 3-5 seconds, which is acceptable. But as we watch the stream for about 30 minutes or more, the latency grows to around 20 seconds, which is just too much. What settings should I look at to keep the latency / delay minimal?
I use FMS 3.5 and i have same problem.....the delay of live video from the studio to clients is big, 10 seconds if i restart the server. After one day of staying on, the delay reaches 1 minute or 2. The FMS is connected to 1 Gibabyte internet fiber, and the stream is VP6, 500kbps video and 128kbps audio, 720x576 resolution. Server Quad 3.0Ghz, 4 Gb Ram.
Please help me to minimize the delay. What i must write in application.xml from live folder?????? Is there a command. I don't want to make buffer, i want the clients to receive live stream, if the stream is buffering i want to play it jumping that sequence, to maintain the live aspect,not to delay it if the clients doesn't have a good internet conection.
Let's suppose i have 1 FMS, 1 Media Encoder, and 2 computers with flash player on the same Switch ( Local area Network). I've made some tests and the delay i get is minimum 5 seconds.
HTTP broadcast delay is too long, I do a test in the LAN, send video using FME 3.2 the flow FMS4.5 server, use the sample folder under the videoplayer to play live video streams, using HTTP delay of 10 seconds or more,(RMTP play live rarely delay), who can tell me, how can I do this problem can solve the delay!
Flash media server 3.5 to stream live video to a webpage with flex GUI embedded in it.For this I have used Adobe flash media server start screen .I could able to stream video correctly.I want to know how long it could be do like this.ie suppose I have to stream like this for 3 or 6 months
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?
I want to have a website where users can record short video clips (using a webcam) and then have those videos saved on the FMS for viewing later. I've seen a lot about streaming video, but I actually want to save the video.
I have an *.flv file on a FMS. When I play it on the client side the video plays just fine, but when I call Stream.play(filename, 0, -1, false) on the server side the video turns out really choppy.I both cases I use NetConnection to connect to an rtmp and NetStream to play the stream, but in one case I connect to a stream and request the server to play my file on that stream. Apparently that doesn't work with files? It works just fine for live streams.
