Im trying to create a Speex Voip client and server. I have the basics down and its working OK on the local machine over UDP. I am using JSpeex for portability. Im looking for tips on creating the client and server. What are your thoughts?
The JSpeex library can only encode 320 bytes per call so the packets sent to the server are tiny (in my case ~244 bytes). Would it be better for the client to wait until about 1 or 2 KB of encoded data is ready before sending or let the server handle buffering the packets?
Also, any help on how to implement buffering the data would be nice.
Some of what I have that works on the local machine.
Client:
public void run() {
int nBytesToRead = (m_inputAudioFormat.getFrameSize() * 160);
int nAvailable = 0;
byte[] abPCMData = new byte[nBytesToRead];
byte[] abSpeexData = null;
UserSpeexPacket userSpeexPacket = new UserSpeexPacket("Xiphias3", "TheLounge", null, 0);
while (m_captureThread != null) {
nAvailable = m_line.available();
if (nAvailable >= nBytesToRead) {
int nBytesRead = m_line.read(abPCMData, 0, nBytesToRead);
if (nBytesRead == -1) break;
if (nBytesRead < nBytesToRead)
Arrays.fill(abPCMData, nBytesRead, abPCMData.length, (byte) 0);
abSpeexData = createSpeexPacketFromPCM(abPCMData, 0, abPCMData.length);
//DatagramPacket packet = new DatagramPacket(abSpeexData, 0, abSpeexData.length, m_connection.getInetAddress(), m_nServerPort);
userSpeexPacket.setSpeexData(abSpeexData);
userSpeexPacket.incrementPacketNumber();
DatagramPacket packet = UserSpeexPacket.userSpeexPacketToDatagramPacket(m_connection.getInetAddress(), m_connection.getPort(), userSpeexPacket);
try {
m_connection.send(packet);
}
catch(IOException iox) {
System.out.println("Connection to server lost: " + iox.getMessage());
break;
}
}
}
closeLine();
disconnect();
}
public byte[] createSpeexPacketFromPCM(byte[] abPCMData, int nOffset, int nLength)
{
byte[] abEncodedData = null;
m_speexEncoder.processData(abPCMData, nOffset, nLength);
abEncodedData = new byte[m_speexEncoder.getProcessedDataByteSize()];
m_speexEncoder.getProcessedData(abEncodedData, 0);
return abEncodedData;
}
Server:
DatagramPacket packet = new DatagramPacket(new byte[2048], 0, 2048);
byte[] abPCMData = null;
long lPrevVolPrintTime = 0;
while (m_bServerRunning) {
try {
m_serverSocket.receive(packet);
//System.out.println("Packet size is " + packet.getData().length);
//System.out.println("Got packet from " + packet.getAddress().getHostAddress());
//abPCMData = decodeSpeexPacket(packet.getData(), 0, packet.getLength());
UserSpeexPacket usp = UserSpeexPacket.datagramPacketToUserSpeexPacket(packet);
abPCMData = decodeSpeexPacket(usp.getSpeexData(), 0, usp.getSpeexData().length);
m_srcDataLine.write(abPCMData, 0, abPCMData.length);
if (System.currentTimeMillis() >= (lPrevVolPrintTime + 500)) {
//System.out.println("Current volume: " + AudioUtil.getVolumeLevelForPCM22050Hz16Bit1Channel(abPCMData, 0, abPCMData.length));
lPrevVolPrintTime = System.currentTimeMillis();
}
}
catch (IOException iox) {
if (m_bServerRunning) {
System.out.println("Server socket broke: " + iox.getMessage());
stopServer();
}
}
}
I am working on a similar project. From everything I've read, and personal experience, your best option is to work with small bits of data and send them as soon as you can. You want any jitter buffering to be done on the side of the receiver.
It is typical for a VoIP application to send 50-100 packets per second. For uLaw encoding at 8000Hz, this would result in a packet size of 80-160 bytes. The reasoning for this is that some packets will inevitably be dropped, and you want the impact to the receiver to be as small as possible. So with 10ms or 20ms of audio data per packet, a dropped packet may result in a small hiccup, but not nearly as bad as losing 2k of audio data (~250ms).
Additionally, with a large packet size, you must accumulate all of the data at the sender before sending it. So given a typical network latency of 50ms, with 20ms of audio data per packet, the receiver is not going to hear what the sender says for a minimum of 70ms. Now imagine what happens when 250ms of audio is being sent at once. 270ms will elapse between the sender speaking and the receiver playing that audio.
Users seem to be more forgiving of packet loss here and there, which results in sub-par audio quality, because the audio quality of most telephones isn't that great to begin with. However, users are also used to very low latency on modern telephone circuits, so introducing a round-trip delay of even 250ms can be extremely frustrating.
Now, as far as implementing buffering, I have found a good strategy to use a Queue (whoops, using .NET here :)), and then wrap that in a class that tracks the desired minimum and maximum number of packets in the Queue. Use rigorous locking since you will most likely be accessing it from multiple threads. If the Queue "bottoms out" and has zero packets in it (buffer underrun), set a flag and return null until the packet count reaches your desired minimum. Your consumer will have to check for null being returned and not queue anything into the output buffer, however. Alternatively your consumer could keep track of the last packet and repeatedly enqueue it, which may cause looping audio, but in some cases that may "sound" better than silence. You will have to do this until the producer puts enough packets into the queue to reach the minimum. This will result in a longer period of silence for the user, but that is generally better accepted than short, frequent periods of silence (choppiness). If you get a burst of packets and the producer fills up the queue (reaching the desired maximum), you can either start ignoring the new packets, or drop enough packets out of the front of the queue to return to the minimum.
Picking those min/max values is tough though. You are trying to balance smooth audio (no underruns) with minimum latency between sender and receiver. VoIP is fun but it sure can be frustrating! Good luck!
Related
In the application that I am developing in Android, I send bytes of number 5 using sockets tcp and udp. I would like to know if it is possible to get the amount of payload that was received until a SocketTimeoutException exception was thrown.I'm doing some moving tests with Wi-Fi Direct technology so, when sending, all may not be received because the peers are disconnected before.
Also for the case of UDP when packet loss occurs I would like to know the amount of information that I receive.
To read what I get, I use readFully and recieve. This reception is done in a single step or in a loop in which I receive large amounts of information. I could not receive the bytes one by one because it would be really slow.
TCP RX:
ServerSocket serverSocket = new ServerSocket(SERVERPORT);
Socket client = serverSocket.accept();
DataInputStream DIS = new DataInputStream(client.getInputStream());
int tamMensaje = (100 * 1000 * 1000);
byte[] payload = new byte[tamMensaje];
DIS.readFully(payload);
int failures = 0;
for (int i = 0; i < tamMensaje; i++) {
if (payload[i] != 5) {
failures = failures + 1;
}
}
int nPayLoad = payload.length- failures;
client.close();
serverSocket.close();
UDP RX:
DatagramSocket datagramSocket = new DatagramSocket(SERVERPORT);
byte[] buffer = new byte[1400];
DatagramPacket packet = new DatagramPacket(buffer, buffer.length);
int tamMensaje = 1400 *71428;
int iteration = 71428;
for (int i = 0; i < iteration; i++) {
datagramSocket.receive(packet);
msg_received = msg_received + new String(buffer, 0, packet.getLength());
}
byte[] payload = msg_received.getBytes();
int failures = 0;
for (int i = 0; i < tamMensaje; i++) {
if (payload[i] != 5) {
failures = failures + 1;
}
}
int nPayload = payload.length- failures;
datagramSocket.close();
How can I know the amount of information received if the communication is cut off when sending for TCP and UDP occurs as well as in case UDP does not receive everything it should?
Thanks
You can't if you use any of the compound readXXX() methods, as their APIs don't provide any means of retrieving it.
However as, for example, an int is normally sent in one go, e.g. by writeInt(), there is really little reason why half of it would arrive long before the other half, even if you got really unlucky and segmentation or packetization split it. It could happen, but you would have to be really be extraordinarily unlucky, and in any case half an int is no more use than none of it.
Similar considerations apply to readFully(): presumably you are reading something, all of which is supposed to be there, so half of it wouldn't be of much use; and conversely, if it would, don't use readFully().
If you use the basic read() method, the answer of course is zero: nothing arrived before the timeout.
I could not receive the bytes one by one because it would be really slow.
Of course, but that's not the only alternative. Use read(byte[]), with a buffer size of your choice, say 8192. Or a BufferedInputStream.
I'm implementing sending data through Bluetooth in Android between two Android devices. For simplicity, I'll mostly likely pass strings and parse them as JSON or other string format.
Now I'm wondering, how should I read the data to be sure, that I've received all of them? Currently I'm taking the following approach:
byte[] buffer = new byte[1024];
while (!finished) {
// My class
MemoryStream ms = new MemoryStream();
int bytesRead = 0;
do {
bytesRead = input.read(buffer, 0, buffer.length);
ms.write(buffer, 0, bytesRead);
} while (bytesRead == buffer.length);
// Now process data
}
However, this will work only if inputStream.read() will always return as many bytes as were sent on the other device.
For example, I'm assuming, that if first device sends 1234 bytes, first call to read will return 1024 and second 210. Or if first device sends 1024 bytes, first call to read will return 1024 bytes and second - -1 (stream end).
Am I right? Or should I implement my own mechanism of determining, whether all sent data was received (or should I wait for more to complete current chunk)?
The answer is: no. It is possible, that transmission will end (in terms of input.read) and not the whole sent buffer is transferred.
One has to guard transmission, preferably by preceeding the data with their size in bytes and then read data until all of them are transferred.
As a hobby project, I'm writing an android voip client. When writing voice data to the socket (Vars.mediaSocket), many times, the data isn't immediately sent out over the wifi but just stalls and then all at once it will send 20 seconds worth of voice. Then it will stall again and wait for 30 seconds and then send 30 seconds of voice. The wait is not consistent but after a while it will continuously send voice data immediately. I've tried everything from using DataOutputStream to setting the socket output buffer size, setting the sendbuffer size huge, small, and lastly, buffering the voice data from its 32 byte chunks to anything from 128bytes to 32kb.
Utils.logcat(Const.LOGD, encTag, "MediaCodec encoder thread has started");
isEncoding = true;
byte[] amrbuffer = new byte[32];
short[] wavbuffer = new short[160];
int outputCounter = 0;
//setup the wave audio recorder. since it is released and restarted, it needs to be setup here and not onCreate
wavRecorder = null; //remove pointer to the old recorder for safety
wavRecorder = new AudioRecord(MediaRecorder.AudioSource.MIC, SAMPLESWAV, AudioFormat.CHANNEL_IN_MONO, FORMAT, 160);
wavRecorder.startRecording();
AmrEncoder.init(0);
while(!micMute)
{
int totalRead = 0, dataRead;
while(totalRead < 160)
{//although unlikely to be necessary, buffer the mic input
dataRead = wavRecorder.read(wavbuffer, totalRead, 160 - totalRead);
totalRead = totalRead + dataRead;
}
int encodeLength = AmrEncoder.encode(AmrEncoder.Mode.MR122.ordinal(), wavbuffer, amrbuffer);
try
{
Vars.mediaSocket.getOutputStream().write(amrbuffer);
Vars.mediaSocket.getOutputStream().flush();
}
catch (IOException i)
{
Utils.logcat(Const.LOGE, encTag, "Cannot send amr out the media socket");
Utils.dumpException(tag, i);
}
Is there something I'm missing? To simulate a second cell phone, I have another client which just simply reads the voice data, throws it away, and reads again in a loop. I can confirm in the simulated second cell phone when the real cell phone stops sending voice, the simulated one's socket.read hangs until the real one starts sending voice again.
I'm really hoping not to have to write a jni for the socket as I don't know anything about that and was hoping I could write the app as a standard java app.
CASE CLOSED: turned out to be a server side bug but the simplifying back to basics suggestions is still a good idea.
You are adding most of the latency yourself by reading large amounts of data before writing any of it. You should just use the standard Java copy loop:
byte[] buffer = new byte[8192];
int count;
while ((count = in.read(buffer)) > 0)
{
out.write(buffer, 0, count);
}
You need to adapt this to incorporate your codec step. Note that you don't need a buffer the size of the entire input. You can tune its size to suit yourself but 8192 is a good starting point. You can increase it to say 32k but don't decrease it. If your codec needs the data in fixed-size chunks, use a buffer of that size and DataInputStream.readFully(). But the larger the buffer the more the latency.
EDIT Specific issues with your code:
byte[] amrbuffer = new byte[AMRBUFFERSIZE];
byte[] outputbuffer = new byte [outputBufferSize];
Remove (see below).
short[] wavbuffer = new short[WAVBUFFERSIZE];
int outputCounter = 0;
Remove outputCounter.
//setup the wave audio recorder. since it is released and restarted, it needs to be setup here and not onCreate
wavRecorder = null; //remove pointer to the old recorder for safety
Pointless. Remove.
wavRecorder = new AudioRecord(MediaRecorder.AudioSource.MIC, SAMPLESWAV, AudioFormat.CHANNEL_IN_MONO, FORMAT, WAVBUFFERSIZE);
wavRecorder.startRecording();
AmrEncoder.init(0);
OK.
try
{
Vars.mediaSocket.setSendBufferSize(outputBufferSize);
}
catch (SocketException e)
{
e.printStackTrace();
}
Pointless. Remove. The socket send buffer should be as large as possible. Unless you know that its default size is < outputBufferSize there is no benefit to this. In any case we are getting rid of outputBuffer altogether.
while(!micMute)
{
int totalRead = 0, dataRead;
while(totalRead < WAVBUFFERSIZE)
{//although unlikely to be necessary, buffer the mic input
dataRead = wavRecorder.read(wavbuffer, totalRead, WAVBUFFERSIZE - totalRead);
totalRead = totalRead + dataRead;
}
int encodeLength = AmrEncoder.encode(AmrEncoder.Mode.MR122.ordinal(), wavbuffer, amrbuffer);
OK.
if(outputCounter == outputBufferSize)
{
Utils.logcat(Const.LOGD, encTag, "Sending output buffer");
try
{
Vars.mediaSocket.getOutputStream().write(outputbuffer);
Vars.mediaSocket.getOutputStream().flush();
}
catch (IOException i)
{
Utils.logcat(Const.LOGE, encTag, "Cannot send amr out the media socket");
Utils.dumpException(tag, i);
}
outputCounter = 0;
}
System.arraycopy(amrbuffer, 0, outputbuffer, outputCounter, encodeLength);
outputCounter = outputCounter + encodeLength;
Utils.logcat(Const.LOGD, encTag, "Output buffer fill: " + outputCounter);
Remove all the above and substitute
Vars.mediaSocket.getOutputStream().write(amrbuffer, 0, encodeLength);
This also means you can get rid of 'outputBuffer' as promised.
NB Don't flush inside loops. As a matter of fact flushing a socket output stream does nothing, but the general principle still holds.
I just made two threads communicating in a simple client-server fashion. Here's the code for the sender:
//Open and configure the socket
SocketChannel channel = SocketChannel.open();
InetSocketAddress address = new InetSocketAddress("localhost", 5050);
channel.connect(address);
channel.socket().setTcpNoDelay(true);
channel.socket().setKeepAlive(false);
int count = 0;
OutputStream os = channel.socket().getOutputStream();
int amount = 50;
//Prepare a simple buffer to send.
byte[] data = new byte[amount];
Arrays.fill(data, (byte)1);
double start = System.currentTimeMillis(); //Time it.
//Send and print the throughput each 50000 tuples.
while(true){
count++;
IOUtils.write(data, os);
if(count % 50000==0){
double totalsize = count;
double end = System.currentTimeMillis();
double time = (end-start)/1000;
System.out.println("Writer:"+(totalsize/time)/1000000+"Tuples/second");
}
}
And for the receiver:
.....//Just accepting the channel.
//After accepting the channel. Read the tuples.
readC.socket().setTcpNoDelay(true);
readC.socket().setKeepAlive(false);
byte[] readbuf = new byte[50];
InputStream is = readC.socket().getInputStream();
double start = System.currentTimeMillis();
while(true){
IOUtils.read(is, readbuf);
}
The output says that the average tuples per second is only 0.1 x 10^6, which means each tuple = 50 bytes, I could only get 5MB/s throughput.
Both threads run on local computer, so I assume the ideal throughput is far more larger than I got.
I also check the wireshark for the reason, and found, when I send one tuple, let's say, from port 71581 to port 5050. Then there will be a bunch of tcp packets sent, roughly 300 packets!!!!! AND, among those 300 packets, only 1 packet is really sending the data between 71581 to 5050.
For the other 299 packets, a weird thing is that the communications are between port 71581 and 71580 (Neighbor port). Any body could explain why it is doing such a stupid thing? Can I disable it?
Any suggestions, comments, and ideas would be very appreciated!!!!!!!!!!! Thank you!
I set up a server with a ServerSocket, connect to it with a client machine. They're directly networked through a switch and the ping time is <1ms.
Now, I try to push a "lot" of data from the client to the server through the socket's output stream. It takes 23 minutes to transfer 0.6Gb. I can push a much larger file in seconds via scp.
Any idea what I might be doing wrong? I'm basically just looping and calling writeInt on the socket. The speed issue doesn't matter where the data is coming from, even if I'm just sending a constant integer and not reading from disk.
I tried setting the send and receive buffer on both sides to 4Mb, no dice. I use a buffered stream for the reader and writer, no dice.
Am I missing something?
EDIT: code
Here's where I make the socket
System.out.println("Connecting to " + hostname);
serverAddr = InetAddress.getByName(hostname);
// connect and wait for port assignment
Socket initialSock = new Socket();
initialSock.connect(new InetSocketAddress(serverAddr, LDAMaster.LDA_MASTER_PORT));
int newPort = LDAHelper.readConnectionForwardPacket(new DataInputStream(initialSock.getInputStream()));
initialSock.close();
initialSock = null;
System.out.println("Forwarded to " + newPort);
// got my new port, connect to it
sock = new Socket();
sock.setReceiveBufferSize(RECEIVE_BUFFER_SIZE);
sock.setSendBufferSize(SEND_BUFFER_SIZE);
sock.connect(new InetSocketAddress(serverAddr, newPort));
System.out.println("Connected to " + hostname + ":" + newPort + " with buffers snd=" + sock.getSendBufferSize() + " rcv=" + sock.getReceiveBufferSize());
// get the MD5s
try {
byte[] dataMd5 = LDAHelper.md5File(dataFile),
indexMd5 = LDAHelper.md5File(indexFile);
long freeSpace = 90210; // ** TODO: actually set this **
output = new DataOutputStream(new BufferedOutputStream(sock.getOutputStream()));
input = new DataInputStream(new BufferedInputStream(sock.getInputStream()));
Here's where I do the server-side connection:
ServerSocket servSock = new ServerSocket();
servSock.setSoTimeout(SO_TIMEOUT);
servSock.setReuseAddress(true);
servSock.bind(new InetSocketAddress(LDA_MASTER_PORT));
int currPort = LDA_START_PORT;
while (true) {
try {
Socket conn = servSock.accept();
System.out.println("Got a connection. Sending them to port " + currPort);
clients.add(new MasterClientCommunicator(this, currPort));
clients.get(clients.size()-1).start();
Thread.sleep(500);
LDAHelper.sendConnectionForwardPacket(new DataOutputStream(conn.getOutputStream()), currPort);
currPort++;
} catch (SocketTimeoutException e) {
System.out.println("Done listening. Dispatching instructions.");
break;
}
catch (IOException e) {
e.printStackTrace();
}
catch (Exception e) {
e.printStackTrace();
}
}
Alright, here's where I'm shipping over ~0.6Gb of data.
public static void sendTermDeltaPacket(DataOutputStream out, TIntIntHashMap[] termDelta) throws IOException {
long bytesTransferred = 0, numZeros = 0;
long start = System.currentTimeMillis();
out.write(PACKET_TERM_DELTA); // header
out.flush();
for (int z=0; z < termDelta.length; z++) {
out.writeInt(termDelta[z].size()); // # of elements for each term
bytesTransferred += 4;
}
for (int z=0; z < termDelta.length; z++) {
for (int i=0; i < termDelta[z].size(); i++) {
out.writeInt(1);
out.writeInt(1);
}
}
It seems pretty straightforward so far...
You do not want to write single bytes when you are transferring large amounts of data.
import java.io.FileInputStream;
import java.io.IOException;
import java.io.InputStream;
import java.io.OutputStream;
import java.net.ServerSocket;
import java.net.Socket;
public class Transfer {
public static void main(String[] args) {
final String largeFile = "/home/dr/test.dat"; // REPLACE
final int BUFFER_SIZE = 65536;
new Thread(new Runnable() {
public void run() {
try {
ServerSocket serverSocket = new ServerSocket(12345);
Socket clientSocket = serverSocket.accept();
long startTime = System.currentTimeMillis();
byte[] buffer = new byte[BUFFER_SIZE];
int read;
int totalRead = 0;
InputStream clientInputStream = clientSocket.getInputStream();
while ((read = clientInputStream.read(buffer)) != -1) {
totalRead += read;
}
long endTime = System.currentTimeMillis();
System.out.println(totalRead + " bytes read in " + (endTime - startTime) + " ms.");
} catch (IOException e) {
}
}
}).start();
new Thread(new Runnable() {
public void run() {
try {
Thread.sleep(1000);
Socket socket = new Socket("localhost", 12345);
FileInputStream fileInputStream = new FileInputStream(largeFile);
OutputStream socketOutputStream = socket.getOutputStream();
long startTime = System.currentTimeMillis();
byte[] buffer = new byte[BUFFER_SIZE];
int read;
int readTotal = 0;
while ((read = fileInputStream.read(buffer)) != -1) {
socketOutputStream.write(buffer, 0, read);
readTotal += read;
}
socketOutputStream.close();
fileInputStream.close();
socket.close();
long endTime = System.currentTimeMillis();
System.out.println(readTotal + " bytes written in " + (endTime - startTime) + " ms.");
} catch (Exception e) {
}
}
}).start();
}
}
This copies 1 GiB of data in short over 19 seconds on my machine. The key here is using the InputStream.read and OutputStream.write methods that accept a byte array as parameter. The size of the buffer is not really important, it just should be a bit larger than, say, 5. Experiment with BUFFER_SIZE above to see how it effects the speed but also keep in mind that it probably is different for every machine you are running this program on. 64 KiB seem to be a good compromise.
Hey, I figured I'd follow up for anyone that was interested.
Here's the bizarre moral of the story:
NEVER USE DataInputStream/DataOutputStream and sockets!!
If I wrap the socket in a BufferedOutputStream/BufferedInputStream, life is great. Writing to it raw is just fine.
But wrap the socket in a DataInputStream/DataOutputStream, or even have DataOutputStream(BufferedOutputStream(sock.getOutputStream())) is EXTREMELY SLOW.
An explanation for that would be really interesting to me. But after swapping everything in and out, this is what's up. Try it yourself if you don't believe me.
Thanks for all the quick help, though.
Maybe you should try sending ur data in chunks(frames) instead of writing each byte seperately. And align your frames with the TCP packet size for best performance.
Can you try doing this over loopback, it should then transfer the data in second.
If it takes minutes, there is something wrong with your application. If is only slow sending data over the internet it could be you network link which is slow.
My guess is that you have a 10 Mb/s network between your client and your server and this is why your transfer is going slowly. If this is the case, try using a DeflatoutOutputStream and an InflatorInputStream for your connection.
How are you implementing the receiving end? Please post your receiving code as well.
Since TCP is a reliable protocol, it will take steps to make sure the client is able to receive all of the data sent by the sender. This means that if your client cannot get the data out of the data receive buffer in time, then the sending side will simply stop sending more data until the client has a chance to read all the bytes in the receiving buffer.
If your receiving side is reading data one byte at a time, then your sender probably will spend a lot of time waiting for the receiving buffer to clear, hence the long transfer times. I'll suggest changing your receiving code to reading as many bytes as possible in each read operation . See if that will solve your problem.
Since I cannot yet comment on this site, I must write answer to #Erik here.
The problem is that DataOutputStream doesn't buffer. The whole Stream-thing in Java is based on decorators design pattern. So you could write
DataOutputStream out = new DataOutputStream(new BufferedOutputStream(socket.getOutputStream()));
It will wrap the original stream in a BufferedOutputStream which is more efficient, which is then wrapped into a DataOutputStream which offers additional nice features like writeInt(), writeLong() and so on.
#Erik: using DataXxxputStream is not the problem here. Problem is you were sending data in too small chunks. Using a buffer solved your problem because even you would write bit by bit the buffer would solve the problem.
Bombe's solution is much nicer, generic and faster.
You should download a good packet sniffer. I'm a huge fan of WireShark personally and I end up using it every time I do some socket programming. Just keep in mind you've got to have the client and server running on different systems in order to pick up any packets.
Things to try:
Is the CPU at 100% while the data is being sent? If so, use visualvm and do a CPU profiling to see where the time is spent
Use a SocketChannel from java.nio - these are generally faster since they can use native IO more easily - of course this only helps if your operation is CPU bound
If it's not CPU bound, there's something going wrong at the network level. Use a packet sniffer to analyze this.
I was using PrintWriter to send data. I removed that and sent data with BufferedOutputStream.send(String.getBytes()) and got about 10x faster sending.
How is your heap size set? I had a similar problem recently with the socket transfer of large amounts of data and just by looking at JConsole I realized that the application was spending most of its time doing full GCs.
Try -Xmx1g
USe Byte buffer for sending the data