Multocast streams over LWAPP

Hi, Just a questions concerning multicast over wireless. We have remote AP;s homed back over a WAN to a WiSM running 7.0.253. we see an multicast stream develope between the edge router and the group address but is the multicast traffic to the AP's considered part of that stream?
Thanks!

> i tried to find a way to watch them in Mac OS X
I think vp62 can be played on OSX with MPlayer or ffmpeg ...

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Slow streaming over wifi using ipad

Why is streaming over wifi slower than over cellular connection? Other devices don't have the same issue.

generally we all think and expect wifi will be faster than a cell connection. But that will not always be the case. A good uncongested LTE runs pretty fast.
But the wifi should be fast enough that you wouldn't notice any issues unless:
You are using a public wifi spot where the signal is metered. Where I work, they clamp the wifi for guests at 250k.
The wifi cannot run any faster than your internet connection. Is it clean?
Recent stoires here are explaining how verizon and Comcast have been allowing you tube and netflix feeds to become overcongested, in an effort to extract $$ from them for the privilange of sending you the signal.
a good place to start is to get a speed app - like Speed test (there are a bunch in the store) and see what speed your wifi is delivering.
One thing that is almost certain, your wifi is a whole lot cheaper than your cell signal!

HTTP 1.1: Using streams over socket; How ?

Can anyone give me a short example as how to handle streams over a persistent socket-connection ?
I've trouble handling this.
I can't "reget" the stream (create a new input/output stream over an existing socket-connection).
Is there someone who can help me ?
I'm getting confused !!!

Since no one helped I could only try to find out myself:
The problem is I used some example-source that fitted well for HTTP 1.0, but led to erroneous results when applied to HTTP 1.1
It's no problem in HTTP 1.1, but you need to read ALL input (headers and possibly content) before trying to process anything:
From the headers you need to read until an empty line is discovered.... Don't just break on return or newline, this will leave an extra character in the input:
Protocol requires both \r and \n. So you always break 1 char to early when scanning bytes and then concluding on a \r that the line is over.
Then you should possibly read 'contentlength bytes' (I've not tried it out yet) in case of POST.
Flushing is enough to write all data to output.
Open and close can't work. "Regetting" a stream is not possible, so all input for this request must be taken from inputstream before trying to process the next request.

Streaming over HTTP

I'm new to streaming, so hopefully this is a quick answer
question.....due to firewall restrictions I would need to STREAM
via HTTP protocol over (80)....I can't use RTMPT (which I was able
to get working) because of protocol restrictions.....however, all
the documentation keeps hinting that you can only progressive
download via HTTP...is this true?
However, other part of the doco say "you can stream via HTTP
if you have APACHE installed". My assumption was that all this was
doing was allowing Apache to serve the http "pages" and progressive
download the files....so, long story short, can a file be streamed
using FMIS over HTTP? Thanks for any help in advance.....

No... I think the docs are just using the term "streaming" so
the author didn't have to go into a long explanation of progressive
video (to save the non-technical users from information overload
maybe). You can't truly stream over HTTP. All HTTP flv delivery is
progressive.
That said, you can deliver a streaming-like experience using
an HTTP application server. See the following:
http://flashcomguru.com/index.cfm/2005/11/2/Streaming-flv-video-via-PHP-take-two

IPhone HE-AAC Streaming over Mobile Network (3G)

Developed an internet radio streamer using utilizing the html5 audio tags with jQuery and with a flash fall back for unsupported browsers, streaming through an icecast2 server. Upon testing the player on the iPhone Safari (iOS 5.0.1), we ran into a very peculiar issue.
When the iPhone is connected to WiFi, it streams perfectly using the HE-AAC V2 stream @ 64kbps 44.1kHz (the preferred codec for apple products). However, when the iPhone is connected to the 3G mobile network (ATT and Verizon), it "stutters/buffers" or stops streaming for 1-2 secs every 1-2 minutes (does not stop streaming completely). The troubling thing is when the iPhone is forced to use a separate MP3 stream at the same bit rate, it does not have this issue and works very well on 3G.

Hi Lawrence,
Thanks for the fast response. I'm confused as to how AAC requires higher bandwidth over MP3 when it can produce higher quality audio at lower bit rates therefore requiring less bandwidth to stream over 3G. Are you referring to the bandwidth of the audio signal itself (at the codec level)?

Bought a Winegard indoor HDTV antenna for streaming over-the-air TV. Will my Apple TV work

FlatWave Mini
FlatWave
I Am considering cutting my cable service and have bought a Winegard indoor HDTV wall mount antenna for streaming over-the air tv? I would appreciate an advice you might give me on this project. Will my Apple TV work with this setup? Thanks so much.

appetv is a media box not a OTA related product in any way
it use the LAN network to play shared media from your computer
and if it has an Internet connection it offer a range of Internet streaming services
read more
http://www.apple.com/appletv/

Will ipad allow higher quality video streaming over 4G?

Will there be improved video quality on youtube and other video streaming on 4G compared to existing 3G connctions? The ipad appears to dumb down video streaming in 3G compared to streaming the same thing on wifi. It is often highly pixelated and blocky, Will the new ipad allow higher quality video streaming over 4G, i.e. the same quality as wi-fi?

The YouTube app is made by Google. Any suggestions you have about that app should be directed to them.

RTP Streaming over the internet

Having some problems streaming over the internet - probably because im doing it wrong.
I can stream over a network fine using Transmit2.java.
But when i try over the internet cant pick up the stream.
Im using www.whatismyip.com to get the internet IP of the 2 computers im trying to stream to and from - and using these in the code, but nothing happens.
Help?

natdeamer wrote:
Having some problems streaming over the internet - probably because im doing it wrong.That's correct, you're doing it wrong.
Im using www.whatismyip.com to get the internet IP of the 2 computers im trying to stream to and from - and using these in the code, but nothing happens.99% of the time, your public IP actually addresses your router, rather than your computer. That means your computer is not publically addressably by it's IP address alone. You'll need to do something called a "NAT holepunch", which you can look up online. Also, I've included two links to discussions I've had with people about the same issue.
[http://forums.sun.com/thread.jspa?forumID=28&threadID=5355413]
[http://forums.sun.com/thread.jspa?forumID=28&threadID=5356672]

App-V 5.0 Streaming over SCCM 2012 SP1 - High Availability / Load Balancig

Hello Anybody,
the default App-V 5.0 Streaming (SharedContentStoreMode = on) Deployment over SCCM belongs to a HTTP-Streaming over http:\\SCCM-Server\... . The whole content flows from SCCM to the App-V-Client. Is it possible to create there a high availability solution?
Yes, we can set a second distribution point. But in th App-V-Client are always only one Distribution Source.
Is it possible to set there an clustered adress of an other IIS? Is it possible to change the publishing Distribution Source with an SMB-Path to File-Cluster? Is there a trick with Source-Root-Parameter in App-V-Client? Or are ther anything else to create
a high availibility App-V-Streaming-Solution?
Yes, with App-V-Management-Server there is a high availability solution. - But we have only SCCM-Infrastruktur for App-V-Distribution.
Thank you, best regards
Daniel

Daniel, do you found solution?

Passing serialized object and data as byte stream over same stream

I am writing a chat program, I am keeping online user List as Vector A would like to pass online user List to client as Vector object also the client message is sended as byte stream. Is it possible to pass object and data as byte stream over the same stream.

I am writing a chat program, I am keeping online user
List as Vector A would like to pass online user List
to client as Vector object also the client message is
sended as byte stream. Is it possible to pass object
and data as byte stream over the same stream.Why are you sending the client message as a byte stream?
String seems a much more logical type. I would assume that you're reading strings from one client, and printing them on the others. If you translate to bytes in the middle, you're going to have to ensure that you decode those streams correctly on the other side.
And if I were implementing this, I probably wouldn't send the raw Vector or List ... instead, I'd create a Message object, which wraps the Vector/String, and a MessageDispatcher interface, which the client implements to handle incoming messages.

Microphone Streaming over Network is Choppy

Hello.
Part of a program in development allows audio to be streamed over the Internet via microphone capture. The current solution seems to be working alright... however, at times the audio gets extremely choppy on the receiving side, almost to the point of being unable to hear it at all... 15 second gaps etc...
I was wondering if you would look at this code and had any suggestions how I could improve this to make it flow better.
Any help is appreciated. First is the server code. Second is the client (receiving code)
public class MicrophoneManager implements Runnable {
private static volatile ByteArrayOutputStream byteArrayOutputStream;
private static String communicatorIP = null;
public static volatile boolean interrupt = false;
private AudioFormat getAudioFormat() {
float sampleRate = 8000.0F;
//8000,11025,16000,22050,44100
int sampleSizeInBits = 8;
//8,16
int channels = 1;
//1,2
boolean signed = true;
//true,false
boolean bigEndian = false;
//true,false
return new AudioFormat(
sampleRate,
sampleSizeInBits,
channels,
signed,
bigEndian);
}//end getAudioFormat
public static void setCommunicatorIP(String communicatorIP) {
MicrophoneManager.communicatorIP = communicatorIP;
@Override
public void run() {
try {
AudioFormat audioFormat = getAudioFormat();
DataLine.Info dataLineInfo = new DataLine.Info(TargetDataLine.class, audioFormat);
TargetDataLine targetDataLine = (TargetDataLine) AudioSystem.getLine(dataLineInfo);
targetDataLine.open(audioFormat);
targetDataLine.start();
ArrayList<Byte> byteArrayList = new ArrayList<Byte>();
byte tempBuffer[] = new byte[10000];
byteArrayOutputStream = new ByteArrayOutputStream();
Thread mcThread = new Thread(new MicrophoneCommunicator());
mcThread.start();
//bos = new BufferedOutputStream(byteArrayOutputStream);
while (!interrupt) {
while (byteArrayList.size() < 10000) {
targetDataLine.read(tempBuffer, 0, tempBuffer.length);
for (byte b : tempBuffer) {
byteArrayList.add(b);
byte[] writerArray = new byte[byteArrayList.size()];
tempBuffer = new byte[10000];
for (int i = 0; i < byteArrayList.size(); i++) {
writerArray[i] = byteArrayList.get(i);
byteArrayList.clear();
byteArrayOutputStream.write(writerArray, 0, writerArray.length);
targetDataLine.close();
} catch (Exception e1) {
global.debug(e1);
private class MicrophoneCommunicator implements Runnable {
public void run() {
try {
Socket microphoneSocket = new Socket(communicatorIP, 5751);
BufferedOutputStream netStreamOut = new BufferedOutputStream(microphoneSocket.getOutputStream());
while (!interrupt) {
if (byteArrayOutputStream.toByteArray().length >= 10000) {
netStreamOut.write(byteArrayOutputStream.toByteArray());
byteArrayOutputStream.reset();
microphoneSocket.close();
} catch (Exception e1) {
global.debug(e1);
}Now, here is the other code.
public class Microphone implements Runnable {
public void run() {
try {
byte[] tempArray = new byte[10000];
ServerSocket micSocket = new ServerSocket(5751);
micSocket.setSoTimeout(0);
Socket microphoneSocket = micSocket.accept(); //Now connected
BufferedInputStream bis = new BufferedInputStream(microphoneSocket.getInputStream());
ByteArrayInputStream bais = new ByteArrayInputStream(tempArray);
AudioFormat audioFormat = getAudioFormat();
AudioInputStream audioInputStream =
new AudioInputStream(bais,
audioFormat,
tempArray.length / audioFormat.getFrameSize());
DataLine.Info dataLineInfo =
new DataLine.Info(
SourceDataLine.class,
audioFormat);
SourceDataLine sourceDataLine = (SourceDataLine) AudioSystem.getLine(
dataLineInfo);
sourceDataLine.open(audioFormat);
sourceDataLine.start();
ArrayList<Byte> byteArrayList = new ArrayList<Byte>();
boolean interrupted = false;
while (!interrupted) {
int nextByte = 0;
while ((nextByte = bis.read()) != -1 && (byteArrayList.size() < 10000)) {
byteArrayList.add((byte) nextByte);
for (int i = 0; i < byteArrayList.size(); i++) {
tempArray[i] = byteArrayList.get(i);
byteArrayList.clear();
sourceDataLine.write(tempArray, 0, tempArray.length);
interrupted = global.micInterrupt;
microphoneSocket.close();
micSocket.close();
sourceDataLine.close();
} catch (Exception e1) {
e1.printStackTrace();
private AudioFormat getAudioFormat() {
float sampleRate = 8000.0F;
//8000,11025,16000,22050,44100
int sampleSizeInBits = 8;
//8,16
int channels = 1;
//1,2
boolean signed = true;
//true,false
boolean bigEndian = false;
//true,false
return new AudioFormat(
sampleRate,
sampleSizeInBits,
channels,
signed,
bigEndian);
}//end getAudioFormat
}If you guys have any ideas on how to make this smoother... please feel free... Also, please feel free to use this code however you wish. I'd really appreciate this... It's hard for people to communicate when it's so choppy this way.
However, the stream is currently only one way though, it only goes from one system to another... not two-way. One talks, the other types... it's complicated, but just might be useful info.
Both sides are communicating via lan... so speed of connection isn't an issue...
Thanks in advance.
Best,
Chris

I'd recommend using UDP streams instead of TCP streams, and I'd recommend discarding out-of-order packets and packets that are delayed beyond a certain threshold (which you can play with)...

Video not played using JMF while tramitting stream over RTP

video not played using JMF while tramitting stream over RTP
I am using code to tranmit video using RTP using JMF and i am able to successfully get the video on client side
Code to send the video over rtp
http://java.sun.com/javase/technologies/desktop/media/jmf/2.1.1/solutions/AVTransmit2.java
code to recieve video over rtp
http://java.sun.com/javase/technologies/desktop/media/jmf/2.1.1/solutions/AVReceive.html#source
My problem is i am not able to play video in AVTransmit2.java.
If i create a player using datasouce, eigther i get player already occupied or a black screen.
please help me running the video on Transmit side as well.
NOTE: i am using web cam for video source
Edited by: qasitouch on Dec 23, 2009 5:55 AM

I am not getting any error on consol, rather video is not displayed on media screen but black rectangle.
On Transmitter
     // Try to create a processor to handle the input media locator
     processor = Manager.createProcessor(ds);
createPlayer(Manager.createCloneableDataSource(ds)); // here i am craeting a player to show the video on transmitter end aswell.
Player code:
     private void createPlayer(DataSource ds) {
          try {
               Player player;
               if(processor!=null)
                    System.out.println("processor: " + processor);
                    player = processor;
               }else{
                    System.out.println("null is processor");
               player = Manager.createRealizedPlayer(ds);
               Component video = player.getVisualComponent();
               JFrame mediaFrame = new JFrame("Camera Screen");
               mediaFrame.setDefaultCloseOperation(JFrame.EXIT_ON_CLOSE);
               mediaFrame.add(video);
               mediaFrame.setSize(300, 300);
               mediaFrame.setVisible(true);
               player.realize();
               player.start();
               player.prefetch();
          } catch (NoPlayerException e) {
               // TODO Auto-generated catch block
               e.printStackTrace();
          } catch (CannotRealizeException e) {
               // TODO Auto-generated catch block
               e.printStackTrace();
          } catch (NotRealizedError e) {
               // TODO Auto-generated catch block
               e.printStackTrace();
          } catch (IOException e) {
               // TODO Auto-generated catch block
               e.printStackTrace();
          } catch (Exception e) {
               // TODO Auto-generated catch block
               e.printStackTrace();
I am not getting any error and not running on the same side. My problem is showing video on Tranmitter end.
Using above code i can create player and a black screen is showing, i am ubable to find, why video is not played in screen.

Slow Fast Forward or stalls - streaming over wifi.

To anyone who is interested,
I was having problems trying to fast forward an HD film over my Wifi(g) network. For some reason, apple tv stalls if the fast forwarding out paces your buffering speed. In a nutshell, my network is fast enough to watch hd over but too slow to fast forward faster than X2.
Does anyone know how to improve this setting? Apart from opting for a new router, wired and removing other wireless devices?
For a short term solution for anyone out there, the remote feature on the iphone allows you to simply drag along the length of the film, similar to that of a film, and play from where you stop. EXTREMELY USEFUL when compared to the response-less rubbish that is the white remote!

I have that problem all the time with synced HD video. If I fast forward or rewind, the video will start sputtering after that. I don't think it's your network, it's probably a glitch in the ATV.
In any case, if you're streaming from a Mac via wifi, make sure you turn Time Capsule off. If it starts running while you're streaming, it will definitely degrade performance.

Reading input stream over the tcp socket in unix

I have a program that reads data from input stream from the socket. If the data is over 1500 bytes it is sent in multiple TCP packets. Whats weird is, if I run the program in windows environment it waits till it receives all the packets but when I run the same program in unix environment it only reads the first packet and go further without waiting for all the TCP packets!!
The line that reads from input stream is
datalen = inStr.read(byteBuffer);is there anyway I can make it wait till it receives all the packets on unix system? I do not understand why it works fine for windows in this case but not for unix.
I'll appreciate any help..
Thanks

When the network is busy there can be any amount of dleay between packets. If this is ever 100 ms , then this will break.
If you send more than one packet per 100 ms you will get two packets at once which will look like one longer packet. Unless you check for this the second packet may get ignored.
The safe way is to send the packet size before sending the packet. Then on the client read the packet to the correct length. Otherwise you will have a program which just happens to work rather than one which will always work.

Streaming over WiFi is not fluent

When I stream video from my mac or iPad over wifi, video is not fluent, it is stucking. When I stream it from my mac over ethernet (apple tv is connected to my network by ethernet), it works just fine.
These problems started after updating my Apple TV few months ago.
Any idea? :/

Hmm, I am trying to insert picture but it does not react at all..
So I used my dropbox
https://www.dropbox.com/s/bsxn4lb95qxj031/Sn%C3%ADmek%20obrazovky%202014-12-05%2 0v%C2%A011.23.22.png?dl=0
https://www.dropbox.com/s/0oibbrgiy1hf2kb/Sn%C3%ADmek%20obrazovky%202014-12-05%2 0v%C2%A011.23.09.png?dl=0

Multocast streams over LWAPP

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