Microphone streaming over SIP protocol.

Hello,
can I stream microphone voice over SIP protocol?
Is there a library to use SIP protocol without RTMP communication?
thanks

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)...

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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)...

Flash sockets over SIP-protocol: HTTP/1.1 501 Not Implemented

Hi there.
I'm developing a chat app based on IRC server and flash client (flash sockets). Everything is OK but for some ISP flash client doesn't connect to IRC server: There are some logs from chat while connecting to IRC server:
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Hi there.
I'm developing a chat app based on IRC server and flash client (flash sockets). Everything is OK but for some ISP flash client doesn't connect to IRC server: There are some logs from chat while connecting to IRC server:
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    Send: NICK nickname
    Send: USER username
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    Receive: Content-type: text/html
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There is only thing I'm aware - this issue occurs for SIP mobile network provider only. For other non-mobile providers issue doesn't occur and chat works good.
I would really appreciate for any thoughts on this matter.

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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:
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I'm new to streaming, so hopefully this is a quick answer
question.....due to firewall restrictions I would need to STREAM
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the documentation keeps hinting that you can only progressive
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However, other part of the doco say "you can stream via HTTP
if you have APACHE installed". My assumption was that all this was
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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
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Hi All,
This is kapil from India and i want to implement VoIP over SIP in j2me.Any help will be appreciated.
I know about SIP API in J2ME. I want to use VoIP over it.Please help.I will be grateful to you.
regards
kapil

Why don't you give us the money your Boss pays you.. then we'll see what can be done ;)
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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.
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Hi
I am working on EDI X12 (835) over File Protocol.
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Thanks & Regards,
Praveena.

Hi Praveena,
Yes, TP name and directory name should be same.
Put oracle.tip.adapter.b2b.allTPInOneDirectory = true (Idetifies the Trading partner by the directory name) in your tip.properties file.
Check below thread for more information on TP identification -
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Regards,
Anuj
Edited by: Anuj Dwivedi on May 5, 2009 4:31 PM

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Hi,
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Appreciate any help on this.
Thanks
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Once more, missleading advices. It is not me who pursues the 'Hollier than thou' attitude, it is you who put yourself in the target.
You cannot connect directly to Oracle using the HTTP
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If you could this would have been documented.
Did you read the documentation?Have you? It would not be the first time you recommend something you don't do.
Ref.
* FTP, HTTP(S), and WebDAV Access to Repository Data.
* Managing Network Connections
* Configuring and Using mod_oradav
-- ~ Madrid
http://hrivera99.blogstpot.com

How to implement SIP protocol?

hi there
how to implement SIP protocol?
I just want to know if there is API for it?
I want to design a instant messaging service.
thx
varun

Look for jain sip:
jain-sip.dev.java.net

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Hello. I noticed the other day that google voice is compatible with "SIP" friendly adapters, like obihai style ones, but skype is not by default. Could I make a friendly request that Skype add SIP functionality so that I can use more devices with it?
Thank you.
-roger-

Skype has supported the SIP protocol for quite some time now. It is however not part of their normal service. If you want to use the SIP version of Skype you will have to pay a premium. The service is called Skype Connect. http://www.skype.com/en/features/skype-connect/

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Hi Lawrence,
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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.

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

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But when i try over the internet cant pick up the stream.
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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]

Microphone streaming over SIP protocol.

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