RTP transmission in linux

I am using linux for JMF RTP transmission.But the RTP data is not being transmitted.Can anyone suggest me the problem behind it.In windows its working fine.Is there any special configuration to be done on linux side for RTP transmission? I am using linux performance pack of JMF in linux machine.I also tried with cross platform pack of JMF.Thanks in advance...

We finally did resolve this issue!!!!
The thing is that Linux just behaves very different from Windows when you create Datagram or MultiCast's.
Windows just assumes that localhost is the local ip address and Linux does not. When you do InetAddress.getLocalHost().getAddress() ; windows often returns the LAN ip address but linux returns localhost.
     InetSocketAddress local = new InetSocketAddress(InetAddress.getByAddress(new byte[]{(byte)192,(byte)168,(byte)3,(byte)2}),port);
     dataSock.bind(local);
     InetAddress destiny = InetAddress.getByAddress(new byte[]{(byte)224,(byte)1,(byte)1,(byte)0});
     dataSock2.connect(destiny, 9000);
When you want to receive multicast packets you need to specify the network interface you access the multicast group. If you dont then it would just use InetAddress.getLocalHost().getAddress() as the address to reach the multicast group. In Linux this just does not work.
     MulticastSocket socket = new MulticastSocket(4446);
     InetAddress group = InetAddress.getByName("230.0.0.1");
     InetAddress address = InetAddress.getByName("192.168.3.2");
     NetworkInterface interf = NetworkInterface.getByInetAddress(address) ;
     SocketAddress socketAddress = new InetSocketAddress(group,4446) ;
     socket.joinGroup(socketAddress, interf );
We are still working on making it work with broadcast addresses.
Greetings!
Nico and Charly

Similar Messages

Help! How to change the video frame size during RTP transmission

Hi there.
In a RTP transmission:
How can I change the size of the frames of the video while I have the processor in the server running?
i.e. while the server is running, i want to change a video from 400x200 to 200x100
Thanks

I am trying to implement the same functionality. I've tried to reset the format using the trackControl.setFormat method but this only appears to be supported while the processor is in a configured state. I'm still investigating, but if you happen to know a solution please let me know.
Here is what I'm trying to do... In the middle of a session, I'm trying to change the video format from h263/rtp:352x288 to h263/rtp:176x144

RTP Transmission over HTTPS (Internet)

Hello people!
I am currently working on my undergraduate thesis. I need to transmit the feed from a video capture device (i.e. a webcam) and also from an audio capture device towards a client in a secure transmission/session. I am already able to capture the feed from the camera and from the microphone and I also am able to transmit it in a nonsecure transmission/session.
I am stuck with the problem of trasmitting the feed securely. I've tried implementing my own Effect. I override the process method of Effect to encrypt the data. However, there is a problem with the RTP transmission. RTPSession only allows JPEG_RTP and H263_RTP. I already junked this method, and I am now looking for another way, particularly transmitting it using HTTPS (on the internet).
Could somebody please help me how to transmit my RTP session over HTTPS. Some sample codes could really help...
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I think you might have the wrong forum. This is a forum about Forte 4GL.
ka

The RTP transmission of a message is slow?

Hi
I use JMF2.2.1e in windows, and there is the transmission of a message in RTP with a wav file of PCM.
However, it takes time for about 20 seconds when a DNS server of Network Neiborhood is set till a sound is played.
It is played whether you exclude setting of DNS immediately if I fill in /etc/hosts with a transmission of a message former address.
Because I do not get it, in specifications of a program making, as for the measures mentioned above, will not there be any good method?
Transmission of a message former appointment is as follows.
String url="rtp://" + addr + ":" + port + "/audio";
MediaLocator m = new MediaLocator(url);
datasink = Manager.createDatasink(ds,m);

The dialog you are referring to is a ''shell dialog''.
The most likely reason for this to take a long time is that the default location is taking a long time to resolve, as can happen if it's a network location or if the default folder doesn't exist.
To reset the download location, reset '''browser.download.dir''' in the advanced configuration interface:
# Select Firefox's location bar (press CTRL+L or click it)
# Type '''about:config''' and press enter
# Click '''I'll be careful, I promise!'''
# in the Filter box, type '''browser.download.dir'''
# Right click the keyword.url preference below and choose '''Reset'''

RTP ReceiveStreamListener in Linux

Hi,
I have problem receiving an RTP stream by using a ReceiveStreamListener object, I am adding my object to RTPStreamMgr but the "update" method never gets called. I have changed my host ip in the /etc/hosts file to my "network IP" from loop-back IP so InetAddress.getLocalHost() works as in windows. Before that I wasn't able to send either. I have tested receiving with "JMStudio" and just worked, the program also works in windows. This is how I start my receiving session.
            mngrSession.addReceiveStreamListener(listener);
            InetAddress adr = InetAddress.getByName(destAddress);
            byte[] badr = adr.getAddress();
            InetAddress nadr = InetAddress.getByAddress(badr);
                mngrSession.initialize( new SessionAddress(InetAddress.getLocalHost(),localPort));
            mngrSession.addTarget(new SessionAddress(nadr, destPort));
            mngrSession.startSession();Any ideas on what I am doing wrong?
Thanks in advance,
-nw

Here, it is. Please have a look.
public synchronized void update(ReceiveStreamEvent event) {
        DataSource ds;
     ReceiveStream stream = event.getReceiveStream();
        if (event instanceof NewReceiveStreamEvent) {
            stream = ((NewReceiveStreamEvent) event).getReceiveStream();
            ds = stream.getDataSource();
            try {
                player = javax.media.Manager.createPlayer(ds);
                if (player == null)
                    return;
                player.realize();
                while (player.getState() != Player.Realized){
                    try {
                        Thread.sleep(100);
                        System.err.println("Realizing..");
                    } catch (InterruptedException ex) {
                        Logger.getLogger(RtpSession.class.getName()).log(Level.SEVERE, null, ex);
                player.start();
            } catch (IOException ex) {
                Logger.getLogger(RtpSession.class.getName()).log(Level.SEVERE, null, ex);
            } catch (NoPlayerException ex) {
                Logger.getLogger(RtpSession.class.getName()).log(Level.SEVERE, null, ex);
    }

Problem to create RTP stream on Linux by using JMF studio

Hello All,
I am trying to use JMStudio in order to create RTP stream
I have a problem with capture
I have run Jmfinit I got following messages
JavaSound Capture Supported = true
JavaSoundAuto: Committed ok
java.lang.Error: Can't open video card 0
java.lang.Error: Can't open video card 1
java.lang.Error: Can't open video card 2
java.lang.Error: Can't open video card 3
java.lang.Error: Can't open video card 4
java.lang.Error: Can't open video card 5
java.lang.Error: Can't open video card 6
java.lang.Error: Can't open video card 7
java.lang.Error: Can't open video card 8
java.lang.Error: Can't open video card 9
it seems to me that i have no directsoundcapture
therefore i can not create RTP package stream
How can i add direct sound capture to my capture devices.
is there any suggestion for this issue?
King regards
BEKIR BALCIK
ARGELA TECHNOLOGIES

In $ORACLE_BASE/admin you should have a directory structure for the viton1 instance : create a similar directory structure for the viton2 instance.
In the pfile subdirectory create the initviton2.ora file : you can copy it from initviton1.ora file and change all occurrences of viton1 to viton2.
At OS prompt : export ORACLE_SID=viton2, then
sqlplus /nolog
conn / as sysdba
startup nomount pfile=$ORACLE_BASE/admin/viton2/pfile/initviton2.ora
then CREATE DATABASE.....

RTP Transmission always lose the last packet

Hi,
I transmit a audio file through rtp by using the JMFStudio(V 2.1.1e), then capture the rtp packets on the sending port, but the last rtp packet is missing every time.
Several developers of my team tried on their own machine and got the same result.
Did anyone try this before or if anybody can tell me how to solve it?
Thanks a lot!

Several thoughts...
If the last "packet" isn't big enough, IE, a full payload worth of data, the underlaying transport mechanism may keep it trapped in the buffer forever. If it sends packets based on a BufferFull event of some sort, then the last packet won't ever be sent because it'll get trapped in the buffer. This is a big problem with TCP sockets, and in that instance, the buffer is flushed automaticlly when a close operation is called on the socket.
I have no idea how you're managing the sending of RTP packets, but that's a possible reason.
Another possible reason would be how you're handling the EndOfMedia event. If you were to, say, stop the entire process down before the final RTP packet got sent out, then you'd also not be getting the last packet sent out. Everything in JMF pretty much spawns a new execution thread, so it's possible the main thread is shutting those other threads down before they're done doing work with calls to whatever.Stop() and whatever.Close()...
Those are just 2 ideas off the top of my head to explain why a last packet may not be sent out...

RTP Transmission, Receive Woes

Hi ppl,
I developed an app to transmit captured video and audio as RTP streams. I made a monitor of the video and audio being captured which is displayed on the Frame. I also made a player from the RTP data received from another system in the network which transmits the data as RTP streams.
My problem is as soon as the player is realized and starts, the other components in the window, say a JList control, JButton controls are invisible and the JFrame upon which the Monitor and player is placed is also unresponsive. But the Monitor still displays video/audio being captured and the Player plays the video/audio being received from the network.
Can anybody offer some help on this? Please ...

Hi!
You r probably suffering from the problem of light weight component(tat r swing components) n heavy weight component(getVideo..... method components). Here heavy weight components r not working properly with these light weight components so..... not working properly for it more read article available title something as "Heavy weight n light weight components" as this site.........
I m not sure much but this might show u some direction to work properly..............

RTP transmission problem.

Hi,
I developed a audio/video conferencing application which transmit the rtp data from server to all clients. The problem I facing is the system working perfectly on my home network. but if I try in my university network it doesnt work.
I use the IP address 192.168.0.255 for the transmitting server, and the client would use the IP of the server to access the data. In univasersit environment, I change the IP address accordingly.
What would be the cause of this problem. Is it probably because of my university network settings?
Plz help me ...thanks

I've understood my problems...My error was to not close the Player objects created by Receiver...
I thought that closing RTPManager, i'll close the associated Player object, but this is not true..
Each Player object need to be closed...

It takes 3-5 seconds to start a RTP Transmission?

Hi there,
Has anyone experienced the above? I can't hear anything on the receiving end until like about 3-5 seconds after the transmission started.

Could be some time lapse in identifying the right device to play the sound/media file.
Do you have a Streaming Server installed to transmit audio files where?
1. you have written a transmit program to transmit all audio files
2. you have a receiver program to receive the same
If this is the case can you ellaborate on the Streaming Server details.
rgds,
seetesh

Is jmf required on recieving side during rtp transmission

hi to all,
iam new to jmf.
I am transmitting media through rtp using AVTransmit2.
But i dont want to use jmf on recieving side .
can someone suggest me, how to do that .
An early response would be a great help to me .

rohit_rdx wrote: hi to all,
iam new to jmf.
I am transmitting media through rtp using AVTransmit2.
But i dont want to use jmf on recieving side . >By 'I' I assume you actually mean you do not want to require your end user to install JMF?
Note that it is easy (for the user) if you (the developer) deploy the JMF* using webstart. Installing JMF can then be as easy as 'one click' for some types of deployments** for most users***.
* JMF, or an application that uses JMF.
** E.G. If you have a sandboxed app. that suggests desktop shortcuts and menu items, the user will be prompted in dialogs - no longer a 'one click' install.
*** If the user has a suitable minimum version of Java to run the application, otherwise the Java plug-in might prompt them to update their Java.

Re: RTP transmission problem

Is it necessary to use AVTransmit3 and RTPSocketAdapter classes for your project..
If not, for your requirement, you can use AVTransmit2 and you can add a remoteSessionAddress whenever you need.
If you want to use the AVTransmit3 and RTPSocketAdapter, keep the createTransmitter in separate class, and create a object whenever you want (say new avater met), here u have to careful to use the local datasource...
Karthikeyan R

I've understood my problems...My error was to not close the Player objects created by Receiver...
I thought that closing RTPManager, i'll close the associated Player object, but this is not true..
Each Player object need to be closed...

Not able to Transmit and receive RTP in Linux

Friends,
I am not able to receive the RTP stream transmitted from another Linux machine though I was trying doing it with JMStudio. The transmit shows as if it is working properly and JMStudio in the receiver machine keeps waiting. Tranmission and receiving between the two same machines when in windows is fine.
Can anyone tell me what the problem is, do I have to add any feature to make this work in Linux.
Sreekant

I too am unable to transmit OR recieve RTP from Linux (RedHat 8.0). I am using JMF-2.1.1e and J2SDK 1.4.2.
I can play files on Linux and I can transmit and recieve between Win boxes (98 and 2K) but my RedHat (8.0 and/or 7.3) can not send or recieve to anyone or anything. I have tried JMStudio and RTPPlayerApplet.
TCPDUMP says udp port 22224 unreachable [tos 0xc0]
JMStudio says that no data has arrived.
I have all firewalling turned off and have tested on two different networks without success. All other networking functions are operational.
My problem is identicle to the problem decribed in the following linked post.
http://forum.java.sun.com/thread.jsp?thread=154947&forum=28&message=966551
I also found another discussion that referenced big/little endian problems in the JMF implementation. (scary stuff)
http://www.vovida.org/pipermail/rtp/2001-April/000112.html
Any help or tips on resolving this issue are welcome. I will be happy to supply more info if needed. Also, if anyone has managed to get RTP working on Linux then what versions Linux/JDK/JMF, what program?
Thanks in advance.

JMF How to stream rtp from udp packet

I implemented an rtsp client and use the client to setup two rtp session(audio, vedio). But when I use the example pramgram "AVReceive3" to stream the udp packet, it doesn't work.
When the AVReceive3 receive the udp packet from the rtp port, it will call the update(ReceiveStreamEvent evt) function, and the event type is StreamMappedEvent, and the call to evt.getReceiveStream().getDataSource() return null.
I thought the first event should be NewReceiveStreamEvent. Please help to solve the problem.
What's rtp packet will cause the event StreamMappedEvent.
Following is the code of AVReceive3.java:
* AVReceive3.java
* Created on 2007年10月30日, 下午4:11
* To change this template, choose Tools | Template Manager
* and open the template in the editor.
package PlayerTest;
import java.io.*;
import java.awt.*;
import java.net.*;
import java.awt.event.*;
import java.util.Vector;
import javax.media.*;
import javax.media.rtp.*;
import javax.media.rtp.event.*;
import javax.media.rtp.rtcp.*;
import javax.media.protocol.*;
import javax.media.protocol.DataSource;
import javax.media.format.AudioFormat;
import javax.media.format.VideoFormat;
import javax.media.Format;
import javax.media.format.FormatChangeEvent;
import javax.media.control.BufferControl;
* AVReceive3 to receive RTP transmission using the RTPConnector.
public class AVReceive3 extends Thread implements ReceiveStreamListener, SessionListener,
ControllerListener
String sessions[] = null;
RTPManager mgrs[] = null;
Vector playerWindows = null;
boolean dataReceived = false;
Object dataSync = new Object();
public AVReceive3(String sessions[])
this.sessions = sessions;
public void run()
initialize();
public boolean initialize() {
try {
mgrs = new RTPManager[sessions.length];
playerWindows = new Vector();
SessionLabel session;
// Open the RTP sessions.
for (int i = 0; i < sessions.length; i++) {
// Parse the session addresses.
try {
session = new SessionLabel(sessions);
} catch (IllegalArgumentException e) {
System.err.println("Failed to parse the session address given: " + sessions[i]);
return false;
System.err.println(" - Open RTP session for: addr: " + session.addr + " port: " + session.port + " ttl: " + session.ttl);
mgrs[i] = (RTPManager) RTPManager.newInstance();
mgrs[i].addSessionListener(this);
mgrs[i].addReceiveStreamListener(this);
// Initialize the RTPManager with the RTPSocketAdapter
mgrs[i].initialize(new RTPSocketAdapter(
InetAddress.getByName(session.addr),
session.port, session.ttl));
// You can try out some other buffer size to see
// if you can get better smoothness.
BufferControl bc = (BufferControl)mgrs[i].getControl("javax.media.control.BufferControl");
if (bc != null)
bc.setBufferLength(350);
} catch (Exception e){
System.err.println("Cannot create the RTP Session: " + e.getMessage());
return false;
// Wait for data to arrive before moving on.
long then = System.currentTimeMillis();
long waitingPeriod = 30000; // wait for a maximum of 30 secs.
try{
synchronized (dataSync) {
while (!dataReceived &&
System.currentTimeMillis() - then < waitingPeriod) {
if (!dataReceived)
System.err.println(" - Waiting for RTP data to arrive");
dataSync.wait(1000);
} catch (Exception e) {}
if (!dataReceived) {
System.err.println("No RTP data was received.");
close();
return false;
return true;
public boolean isDone() {
return playerWindows.size() == 0;
* Close the players and the session managers.
protected void close() {
for (int i = 0; i < playerWindows.size(); i++) {
try {
((PlayerWindow)playerWindows.elementAt(i)).close();
} catch (Exception e) {}
playerWindows.removeAllElements();
// close the RTP session.
for (int i = 0; i < mgrs.length; i++) {
if (mgrs[i] != null) {
mgrs[i].removeTargets( "Closing session from AVReceive3");
mgrs[i].dispose();
mgrs[i] = null;
PlayerWindow find(Player p) {
for (int i = 0; i < playerWindows.size(); i++) {
PlayerWindow pw = (PlayerWindow)playerWindows.elementAt(i);
if (pw.player == p)
return pw;
return null;
PlayerWindow find(ReceiveStream strm) {
for (int i = 0; i < playerWindows.size(); i++) {
PlayerWindow pw = (PlayerWindow)playerWindows.elementAt(i);
if (pw.stream == strm)
return pw;
return null;
* SessionListener.
public synchronized void update(SessionEvent evt) {
if (evt instanceof NewParticipantEvent) {
Participant p = ((NewParticipantEvent)evt).getParticipant();
System.err.println(" - A new participant had just joined: " + p.getCNAME());
* ReceiveStreamListener
public synchronized void update( ReceiveStreamEvent evt) {
System.out.println("\nReceive an receiveStreamEvent:"+evt.toString());
RTPManager mgr = (RTPManager)evt.getSource();
Participant participant = evt.getParticipant(); // could be null.
ReceiveStream stream = evt.getReceiveStream(); // could be null.
System.out.println("The RTPManager is:");
if (evt instanceof RemotePayloadChangeEvent) {
System.err.println(" - Received an RTP PayloadChangeEvent.");
System.err.println("Sorry, cannot handle payload change.");
System.exit(0);
else if (evt instanceof NewReceiveStreamEvent) {
try {
stream = ((NewReceiveStreamEvent)evt).getReceiveStream();
DataSource ds = stream.getDataSource();
// Find out the formats.
RTPControl ctl = (RTPControl)ds.getControl("javax.media.rtp.RTPControl");
if (ctl != null){
System.err.println(" - Recevied new RTP stream: " + ctl.getFormat());
} else
System.err.println(" - Recevied new RTP stream");
if (participant == null)
System.err.println(" The sender of this stream had yet to be identified.");
else {
System.err.println(" The stream comes from: " + participant.getCNAME());
// create a player by passing datasource to the Media Manager
Player p = javax.media.Manager.createPlayer(ds);
if (p == null)
return;
p.addControllerListener(this);
p.realize();
PlayerWindow pw = new PlayerWindow(p, stream);
playerWindows.addElement(pw);
pw.setVisible(true);
// Notify intialize() that a new stream had arrived.
synchronized (dataSync) {
dataReceived = true;
dataSync.notifyAll();
} catch (Exception e) {
System.err.println("NewReceiveStreamEvent exception " + e.getMessage());
return;
else if (evt instanceof StreamMappedEvent) {
if (stream != null)
if(stream.getDataSource()!=null)
DataSource ds = stream.getDataSource();
// Find out the formats.
RTPControl ctl = (RTPControl)ds.getControl("javax.media.rtp.RTPControl");
System.err.println(" - The previously unidentified stream ");
if (ctl != null)
System.err.println(" " + ctl.getFormat());
System.err.println(" had now been identified as sent by: " + participant.getCNAME());
else if (evt instanceof ByeEvent) {
System.err.println(" - Got \"bye\" from: " + participant.getCNAME());
PlayerWindow pw = find(stream);
if (pw != null) {
pw.close();
playerWindows.removeElement(pw);
* ControllerListener for the Players.
public synchronized void controllerUpdate(ControllerEvent ce) {
Player p = (Player)ce.getSourceController();
if (p == null)
return;
// Get this when the internal players are realized.
if (ce instanceof RealizeCompleteEvent) {
PlayerWindow pw = find(p);
if (pw == null) {
// Some strange happened.
System.err.println("Internal error!");
System.exit(-1);
pw.initialize();
pw.setVisible(true);
p.start();
if (ce instanceof ControllerErrorEvent) {
p.removeControllerListener(this);
PlayerWindow pw = find(p);
if (pw != null) {
pw.close();
playerWindows.removeElement(pw);
System.err.println("AVReceive3 internal error: " + ce);
* A utility class to parse the session addresses.
class SessionLabel {
public String addr = null;
public int port;
public int ttl = 1;
SessionLabel(String session) throws IllegalArgumentException {
int off;
String portStr = null, ttlStr = null;
if (session != null && session.length() > 0) {
while (session.length() > 1 && session.charAt(0) == '/')
session = session.substring(1);
// Now see if there's a addr specified.
off = session.indexOf('/');
if (off == -1) {
if (!session.equals(""))
addr = session;
} else {
addr = session.substring(0, off);
session = session.substring(off + 1);
// Now see if there's a port specified
off = session.indexOf('/');
if (off == -1) {
if (!session.equals(""))
portStr = session;
} else {
portStr = session.substring(0, off);
session = session.substring(off + 1);
// Now see if there's a ttl specified
off = session.indexOf('/');
if (off == -1) {
if (!session.equals(""))
ttlStr = session;
} else {
ttlStr = session.substring(0, off);
if (addr == null)
throw new IllegalArgumentException();
if (portStr != null) {
try {
Integer integer = Integer.valueOf(portStr);
if (integer != null)
port = integer.intValue();
} catch (Throwable t) {
throw new IllegalArgumentException();
} else
throw new IllegalArgumentException();
if (ttlStr != null) {
try {
Integer integer = Integer.valueOf(ttlStr);
if (integer != null)
ttl = integer.intValue();
} catch (Throwable t) {
throw new IllegalArgumentException();
* GUI classes for the Player.
class PlayerWindow extends Frame {
Player player;
ReceiveStream stream;
PlayerWindow(Player p, ReceiveStream strm) {
player = p;
stream = strm;
public void initialize() {
add(new PlayerPanel(player));
public void close() {
player.close();
setVisible(false);
dispose();
public void addNotify() {
super.addNotify();
pack();
* GUI classes for the Player.
class PlayerPanel extends Panel {
Component vc, cc;
PlayerPanel(Player p) {
setLayout(new BorderLayout());
if ((vc = p.getVisualComponent()) != null)
add("Center", vc);
if ((cc = p.getControlPanelComponent()) != null)
add("South", cc);
public Dimension getPreferredSize() {
int w = 0, h = 0;
if (vc != null) {
Dimension size = vc.getPreferredSize();
w = size.width;
h = size.height;
if (cc != null) {
Dimension size = cc.getPreferredSize();
if (w == 0)
w = size.width;
h += size.height;
if (w < 160)
w = 160;
return new Dimension(w, h);
public static void main(String argv[]) {
//if (argv.length == 0)
// prUsage();
String sessions[]= new String[] {"127.0.0.1/6670","127.0.0.1/6672"};
AVReceive3 avReceive = new AVReceive3(sessions);
if (!avReceive.initialize()) {
System.err.println("Failed to initialize the sessions.");
System.exit(-1);
// Check to see if AVReceive3 is done.
try {
while (!avReceive.isDone())
Thread.sleep(1000);
} catch (Exception e) {}
System.err.println("Exiting AVReceive3");
static void prUsage() {
System.err.println("Usage: AVReceive3 <session> <session> ");
System.err.println(" <session>: <address>/<port>/<ttl>");
System.exit(0);
}// end of AVReceive3
Following is the code of RTPSocketAdapter.java:
* RTPSocketAdapter.java
* Created on 2007年10月30日, 下午4:13
* To change this template, choose Tools | Template Manager
* and open the template in the editor.
package PlayerTest;
import java.io.IOException;
import java.net.InetAddress;
import java.net.DatagramSocket;
import java.net.MulticastSocket;
import java.net.DatagramPacket;
import java.net.SocketException;
import javax.media.protocol.DataSource;
import javax.media.protocol.PushSourceStream;
import javax.media.protocol.ContentDescriptor;
import javax.media.protocol.SourceTransferHandler;
import javax.media.rtp.RTPConnector;
import javax.media.rtp.OutputDataStream;
* An implementation of RTPConnector based on UDP sockets.
public class RTPSocketAdapter implements RTPConnector {
DatagramSocket dataSock;
DatagramSocket ctrlSock;
InetAddress addr;
int port;
SockInputStream dataInStrm = null;
SockInputStream ctrlInStrm = null;
SockOutputStream dataOutStrm = null;
SockOutputStream ctrlOutStrm = null;
public RTPSocketAdapter(InetAddress addr, int port) throws IOException {
this(addr, port, 1);
public RTPSocketAdapter(InetAddress addr, int port, int ttl) throws IOException {
try {
if (addr.isMulticastAddress()) {
dataSock = new MulticastSocket(port);
ctrlSock = new MulticastSocket(port+1);
((MulticastSocket)dataSock).joinGroup(addr);
((MulticastSocket)dataSock).setTimeToLive(ttl);
((MulticastSocket)ctrlSock).joinGroup(addr);
((MulticastSocket)ctrlSock).setTimeToLive(ttl);
} else {
dataSock = new DatagramSocket(port, InetAddress.getLocalHost());
ctrlSock = new DatagramSocket(port+1, InetAddress.getLocalHost());
} catch (SocketException e) {
throw new IOException(e.getMessage());
this.addr = addr;
this.port = port;
* Returns an input stream to receive the RTP data.
public PushSourceStream getDataInputStream() throws IOException {
if (dataInStrm == null) {
dataInStrm = new SockInputStream(dataSock, addr, port);
dataInStrm.start();
return dataInStrm;
* Returns an output stream to send the RTP data.
public OutputDataStream getDataOutputStream() throws IOException {
if (dataOutStrm == null)
dataOutStrm = new SockOutputStream(dataSock, addr, port);
return dataOutStrm;
* Returns an input stream to receive the RTCP data.
public PushSourceStream getControlInputStream() throws IOException {
if (ctrlInStrm == null) {
ctrlInStrm = new SockInputStream(ctrlSock, addr, port+1);
ctrlInStrm.start();
return ctrlInStrm;
* Returns an output stream to send the RTCP data.
public OutputDataStream getControlOutputStream() throws IOException {
if (ctrlOutStrm == null)
ctrlOutStrm = new SockOutputStream(ctrlSock, addr, port+1);
return ctrlOutStrm;
* Close all the RTP, RTCP streams.
public void close() {
if (dataInStrm != null)
dataInStrm.kill();
if (ctrlInStrm != null)
ctrlInStrm.kill();
dataSock.close();
ctrlSock.close();
* Set the receive buffer size of the RTP data channel.
* This is only a hint to the implementation. The actual implementation
* may not be able to do anything to this.
public void setReceiveBufferSize( int size) throws IOException {
dataSock.setReceiveBufferSize(size);
* Get the receive buffer size set on the RTP data channel.
* Return -1 if the receive buffer size is not applicable for
* the implementation.
public int getReceiveBufferSize() {
try {
return dataSock.getReceiveBufferSize();
} catch (Exception e) {
return -1;
* Set the send buffer size of the RTP data channel.
* This is only a hint to the implementation. The actual implementation
* may not be able to do anything to this.
public void setSendBufferSize( int size) throws IOException {
dataSock.setSendBufferSize(size);
* Get the send buffer size set on the RTP data channel.
* Return -1 if the send buffer size is not applicable for
* the implementation.
public int getSendBufferSize() {
try {
return dataSock.getSendBufferSize();
} catch (Exception e) {
return -1;
* Return the RTCP bandwidth fraction. This value is used to
* initialize the RTPManager. Check RTPManager for more detauls.
* Return -1 to use the default values.
public double getRTCPBandwidthFraction() {
return -1;
* Return the RTCP sender bandwidth fraction. This value is used to
* initialize the RTPManager. Check RTPManager for more detauls.
* Return -1 to use the default values.
public double getRTCPSenderBandwidthFraction() {
return -1;
* An inner class to implement an OutputDataStream based on UDP sockets.
class SockOutputStream implements OutputDataStream {
DatagramSocket sock;
InetAddress addr;
int port;
public SockOutputStream(DatagramSocket sock, InetAddress addr, int port) {
this.sock = sock;
this.addr = addr;
this.port = port;
public int write(byte data[], int offset, int len) {
try {
sock.send(new DatagramPacket(data, offset, len, addr, port));
} catch (Exception e) {
return -1;
return len;
* An inner class to implement an PushSourceStream based on UDP sockets.
class SockInputStream extends Thread implements PushSourceStream {
DatagramSocket sock;
InetAddress addr;
int port;
boolean done = false;
boolean dataRead = false;
SourceTransferHandler sth = null;
public SockInputStream(DatagramSocket sock, InetAddress addr, int port) {
this.sock = sock;
this.addr = addr;
this.port = port;
public int read(byte buffer[], int offset, int length) {
DatagramPacket p = new DatagramPacket(buffer, offset, length, addr, port);
try {
sock.receive(p);
} catch (IOException e) {
return -1;
synchronized (this) {
dataRead = true;
notify();
System.out.println("RTPSocketAdapter receive RTP packet from port:"+port);
System.out.println("The received RTP packet:"+new String(buffer));
return p.getLength();
public synchronized void start() {
super.start();
if (sth != null) {
dataRead = true;
notify();
public synchronized void kill() {
done = true;
notify();
public int getMinimumTransferSize() {
return 2 * 1024; // twice the MTU size, just to be safe.
public synchronized void setTransferHandler(SourceTransferHandler sth) {
this.sth = sth;
dataRead = true;
notify();
// Not applicable.
public ContentDescriptor getContentDescriptor() {
return null;
// Not applicable.
public long getContentLength() {
return LENGTH_UNKNOWN;
// Not applicable.
public boolean endOfStream() {
return false;
// Not applicable.
public Object[] getControls() {
return new Object[0];
// Not applicable.
public Object getControl(String type) {
return null;
* Loop and notify the transfer handler of new data.
public void run() {
while (!done) {
synchronized (this) {
while (!dataRead && !done) {
try {
wait();
} catch (InterruptedException e) { }
dataRead = false;
if (sth != null && !done) {
sth.transferData(this);
Thanks.

The error of No format has been registered for RTP Payload type 96
is caused by the dynamic payload mapping, when I add the dynamic mapping between dynamic payload and format. The Player cann't work yet. I think it because JMF doesn't support the format of my clips. For example:
video: a=rtpmap:96 H263-2000/90000
audio:a=rtpmap:97 MP4A-LATM/12000/1
Is there some available plugin to support these format?
Thanks

Cloning RTP DataSource

I'm having difficulty making the Clone.java example (from JMF Solutions) clone an RTP transmission. The Clone program works fine with a video file, but when I attempt this:
java Clone rtp://localhost:550/video 2
where I have an RTP transmission from JMStudio on port 550, it complains:
create player for: rtp
Failed to create a player from the given DataSource: javax.media.NoPlayerException: Cannot find a Player for: com.ibm.media.protocol.CloneablePushBufferDataSour
ce@551f60
I must be missing something, and I'd be very appreciative for any pointers.
Cheers,
Frank

You can see JMF guide. There're some helpful examples.
Also, I have some problems when capture a RTP session to a file while playing it concurrently. I cloned the datatsource and use one for playing , one for capturing but the playback quality is very bad.
What I have to do.

RTP transmission in linux

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