To retransmit the rtp streams received

hi !
Program A transmits media data to another program B using RTPSessionMgr.What B has to do is send the recived streams to another receiver program C.This has to play the stream.That is I am trying to implement a client router server model.
So i maintained a session between A & B and also between B & C.Once NewReceiveStream event occurs at B,it retrieves the datasource from the stream and uses the createSendStream() method of the sessionmanager and sends to C.
My problem is that,C receives the audio and video streams.but never plays it.I get a pink screen for video and no audio.
can somebody help me out in pointing out my mistake or tell me a new strategy to implement this.
Actually,i have tried to use only datagram sockets on the router side.ie I created 2 sockets for B to receive data. and forwarded to C using 2 other sockets .And this did work.But the client does not know the sender details.I require to maintain the sender,receiver reports.So i went for the session manager(RTPManager /RTPSessionMgr).
kindly help.

Hi all!
Nice to meet ya, it very exciting that I got a project which will integrating the JMF and JXTA, it got RTP streams from the JMF framework and then sending it to the JXTA framework ,and sent to any peer in some groups to visualizing it .So some aspects of which is similar to yours ,would you mind add me to your contact list of MSN(my MSN account is:[email protected])

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Write(Export) the RTP Stream from a SIP Call to an audio file

I am working on telephony system that need record each call to a file.
I got the RTP Stream in ReceiveStreamEvent handler and start to record
the problem is i got a file and play it but there are any sounds
when i set FileTypeDescriptor.WAVE and AudioFormat.ULAW
I try out the FileTypeDescriptor.WAVE and AudioFormat.IMA4_MS, i got a fixed size file with 60kb
if the FileTypeDescriptor.MPEG_AUDIO and AudioFormat.MPEG, the processor cannot be realize,
the thread is blocked!!
Could anyone save me? thanks in advanced!
=================================Code===================================================
public void update(ReceiveStreamEvent evt){
logger.debug("received a new incoming stream. " + evt);
RTPManager mgr = (RTPManager) evt.getSource();
Participant participant = evt.getParticipant(); // could be null.
ReceiveStream stream = evt.getReceiveStream(); // could be null.
if (evt instanceof NewReceiveStreamEvent)
try
stream = ( (NewReceiveStreamEvent) evt).getReceiveStream();
DataSource ds = stream.getDataSource();
          recorder = new CallRecorderImpl(ds);
recorder.start();
catch (Exception e)
logger.error("NewReceiveStreamEvent exception ", e);
return;
========================================================================================
this is the CallRecorderImpl Class
public class CallRecorderImpl implements DataSinkListener, ControllerListener {
     private Processor processor = null;
     private DataSink dataSink = null;
     private DataSource source = null;
     private boolean bRecording = false;
     FileTypeDescriptor contentType = new FileTypeDescriptor(FileTypeDescriptor.WAVE);
     Format encoding = new AudioFormat(AudioFormat.IMA4_MS);
MediaLocator dest = new MediaLocator("file:/c:/bar.wav");
     public CallRecorderImpl(DataSource ds) {
          this.source = ds;
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          try {
               processor = Manager.createProcessor(source);
               processor.addControllerListener(this);
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               System.out.println("exception:" + e);
     public void stop() {
          try {
               System.out.println("stopping");
               this.dataSink.stop();
               this.dataSink.close();
          } catch (Exception ep) {
               ep.printStackTrace();
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          Processor pr = (Processor) evt.getSourceController();
          if (evt instanceof ConfigureCompleteEvent) {
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               processConfigured(pr);
          if (evt instanceof RealizeCompleteEvent) {
               System.out.println("RealizeCompleteEvent");
               processRealized(pr);
          if (evt instanceof ControllerClosedEvent) {
               System.out.println("ControllerClosedEvent");
          if (evt instanceof EndOfMediaEvent) {
               System.out.println("EndOfMediaEvent");
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          for (int i = 0; i < track.length; i++) {
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          p.realize();
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          System.out.println("Entered processRealized");
DataSource source = p.getDataOutput();
try {
     dataSink = Manager.createDataSink(source, dest);
     dataSink.open();
     dataSink.start();
     dataSink.addDataSinkListener(new DataSinkListener() {
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                              dataSink.close();
} catch (Exception e) {
     e.printStackTrace();
     return;
          p.start();
============================================

Which shows that the output stream cannot be of that particulat format and descriptor.
Look at this code
import javax.swing.*;
import javax.media.*;
import java.net.*;
import java.io.*;
import javax.media.datasink.*;
import javax.media.protocol.*;
import javax.media.format.*;
import javax.media.control.*;
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                    dsk.stop();
                    dsk.close();
                    System.exit(1);
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public class JMFCapture6
     public static void main(String args[]) throws Exception
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Save the rtp streaming

Hi,
If my data source type is
"com.sun.media.protocol.vfw.DataSource@cd5f8b" ,
I can use the following method to grab the image frome the real-time video that my webcam catpured
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cds.start();
Buffer buf = new Buffer();
public void transferData(PushBufferStream pbs) {
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Thanks!

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]

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See AVReceive2 in JMF Solutions, in JMF web site

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I want to send the DataSource back to the client, because later, we will be routing different clients, and who they can talk to and such. But I want to start with the basics.

captfoss wrote:
DerNalia wrote:
But what I want to do. Is have send the DataSource back to that client. (The ip address is unknown... but that shouldn't matter, as RTP is two-way).RTP isn't two-way, RTP is one way... RTCP is two-way. The data transport is duplex, the QOS/control messages are duplex...
So you do need to know the IP address of the client in order to send anything to him...
I want to send the DataSource back to the client, because later, we will be routing different clients, and who they can talk to and such. But I want to start with the basics.You should just be able to turn around and give the DataSource that's receiving the RTP stream to an RTPManager and broadcast it back out... assuming you're not transcoding it out of its RTP format or anything.I don't have the IP address. So is this even possible without switching to RTCP? if so if you an example, that would be awesome. Also an example of two way communication with RTCP would be awesome too.
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Hi ppl,
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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.........
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as soon as i try to set the framerate it dies.
any ideas?

i am having problems setting the framerate on the rtp stream, i can set the quality of the jpeg, set the size of the stream, ive tried sizes 160X120, 320X240, 640x480.
as soon as i try to set the framerate it dies.
any ideas?

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Unable to handle format: LINEAR, 8000.0 Hz, 16-bit, Mono, LittleEndian, Signed
Unable to handle format: LINEAR, 8000.0 Hz, 16-bit, Mono, LittleEndian, Signed
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Message was edited by:
kaligula

Hello,
Here is a procedure i wrote to write on text files the content of any table (and also generate the insert SQL order)
http://oracle.developpez.com/sources/?page=developpement#Extraction_table
You can download the script file here http://sheikyerbouti.developpez.com/src/extraction_table.zip
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I have been working long and hard trying to figure out to save an RTP stream from a Cisco phone to a wave. I need the wave file to be in the format ULAW, 8khz, 8 bit mono.
Through much research this is the code I come up with but I does not work. I states that I am trying to get a datasink for null when I try to create the filewriter datasink.
I also have gotten the following error message with other version of the code below:
newReceiveStreamEvent exception Cannot find a DataSink for: com.sun.media.protocol.rtp.DataSource@3680c1
When I get the RTP stream I get the following as the format of the input stream:
- Recevied new RTP stream: ULAW/rtp, 8000.0 Hz, 8-bit, Mono
else if (evt instanceof NewReceiveStreamEvent) {
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Format formats[] = new Format[1];
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MediaLocator f = new MediaLocator("file:/C:/output.wav");
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DataSource tempds = processor.getDataOutput();
filewriter = Manager.createDataSink(tempds, f);
filewriter.open();
Edited by: phodat on Sep 16, 2010 10:17 PM
Edited by: phodat on Sep 16, 2010 10:20 PM

The content descriptor is designed to specify the output format, not the input format. The input format is specified by the DataSource automatically, and you set the content descriptor to whatever you want the output format to be. In this case, you'd want to set it to a WAV file.
You'd then need to go through all of the Track objects on the processor and set their output format to the ULAW specifications as you want there.
[http://www.cs.odu.edu/~cs778/spring04/lectures/jmfsolutions/examplesindex.html]
There's actually an example program that exports RTP streams, you can probably use that code without modification to fit your needs.

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Please help me out on this..
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Meghana

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Hello,
Firstly I am no expert in Cisco telephony as we have just recently migrated to a full Cisco solution, so apologies if I ask a fundamental question.
Client A = Site A
Client B = Site B
Client C = Site C
Site A and Site B are in 1 cluster
Site C is in another cluster
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Client A -> Client B within the same cluster (across 2 sites with a low latency link) RTP stream comes up and the call functions as expected.
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Client A -> Client C this call connects but there is no RTP stream.
We are using G711 across the board and I have captured a wireshark capture from a Client A -> Client C failed call.
I have been going through this capture and noticed that when I search H225 (for the gatekeepers) I see the following –
CS: setup
RAS: admissionRequest
RAS: admissionConfirm
CS: callProeeding
CS: alerting
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Thanks,
Peter

Pat,
I can not talk to the UC540, but I ran into a situation recently where the SIP gateway was sending out the private extension of the phone number instead of the full DID that was registered to the provider. The provider was then blocking call.
In our SIP debugs we saw the RDNIS information of the private extension I believe. The error code we were getting back from SP was code 404 or something along those lines.
I recommend you do some debugs and track where the calls fails, compare the SNR call versus a normal call in the debugs, and then if you still get stuck post running configs and debugs back here.

To retransmit the rtp streams received

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