Help forwarding an RTP stream

I need help in forwarding an RTP audio stream from one client to another.I know i need to create a datasource from the newReceicveStream event and use it as the ds for the processor in transmitter but i have not had success.Can people help me with the code that does forwarding?

Well ill try to help you. I dont really know what you are trying to do but if you want to pass the the rtp object to another client then have a look at this article on how to send objects over sockets.
http://java.sun.com/developer/technicalArticles/ALT/sockets/
Why dont you just send the information required for the rtp (ip, port) to the other client and have them create a new rtp session?

Similar Messages

Help reqd fr rtp streaming in j2me....

hi,
i am new to the world of J2ME n my doing a project fr my final year.i am developing an application, wherein i can chat via bluetooth.This involves chatting via text mssgng,audio n video chat.I hv implemented text module via client/server architecture.
I tried the same method fr live audio/video transfer,but m facing problems..I can save the file temporarily but
The stream parameters read null values while streaming..can't figure out the problem..
However ,i thought of an alternative ie. to use RTP fr streaming those files.
But nw im clueless as to whr to begin,as i read j2me does nt support rtsp/rtp.also vry few devices support it..
pls ppl,i need ur timely help,n il be vry grateful fr all tht u can share..

HI all,...
appearantly we all have the same problems...
i want 2 implement rtp player for ipaq h5550. Ive used j9 but i dont know how to implement mmapi to ipaq so j9 can run it....
i want 2 use the jeode vm with jmf, but i dont have any jeode coz ive lost my cd...and they (HP) doesnt support it anymore,
i heard in the forum that rtp player worked with j9 and jmf crossplatform. so would anyone who have made it please help all of us,..
im beginning 2 felt sick about this.....
thx
regards ,
hendra

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])

RTP Streaming seek & stop question.

Is it possible to seek in a rtp stream received at the client. Since rtp works on push data concept,is there a way to implement this.
I want the server to stop streaming when the client exits. Normally the sender sends Bye Event to Receiver. Is it possible for other way around to tell the sender to stop streaming.
Thanks.

From what I understand RTP on it's is basically for a blind transmission, like watch tv or the listening to the Radio, if the RTP stream (analogous to the tv signal or radio waves) is being broadcast you can receive it. With this method you have no control really, you are just given what is being streamed.
However RTSP, allows you send commands to a server that will stream RTP content, which you can start, stop, pause, and in some cases move forward or backward to a particular time in the sequence.
You may find the following link useful: http://www.cs.columbia.edu/~hgs/teaching/ais/slides/RTSP.pdf
hope this helps,
Stef

Play and save rtp streams

Hello
I need help can someone tell me how i play and save at the same time an rtp stream?
Thanks

Thanks for your response.
Very much appreciated. Was very informative.
This is my current situation with 3 your suggestions:
Daniele,
Your suggestion 1’s result:
In Wireshark ----> Under Statistics --->I have VoIP calls.
(I don’t see VoIP calls under Telephony –> may be a different version of Wireshark).
Anyway, there is only one call because the Wireshark had a Capture Filter to track information between one source and one destination IP address. So I select that call and click on Player button and then click on Decode button. Then I select the forward stream (From IP1 to IP2) and click on play and I don’t hear anything at all. All silence. Same when I select the reverse stream from IP2 to IP1 and play.
Your suggestion 2’s result:
In Wireshark ---> Under Statistics ---> I Selected Stream Analysis (Did not select Show All Streams – not sure what the difference is) then ---> Save Payload ----> Select “au” instead of raw and it says – “Can’t save in a file:saving in au format supported only for alaw / ulaw stream
Your suggestion 3’s result:
Saved the file in .raw format. Opened Audacity and imported the file as raw and specified FIRST the A-Law codec for G.711A and selected 8000hz and that didn’t work and SECOND tried the u-Law coding for G.711u and selected the sample frequency again equal to 8000 Hz and that didn't work.
Didn't work means:
When I played the imported information I get all noise (like heavy metallic sound) and no voice.
So my guess is that this capture is neither A-Law or u-Law codec - right. This capture was given to me by a customer.
Any other suggestions – much appreciated Daniele.

Mixing audio RTP-Streams to conference

I'm trying to create a telefon conference applikation with jmf.
My problem:
I need to mix multiple incomming RTP-Streams (Audio/G711_ulaw) together into one outgoing RTP-Stream.
If I use a MergingDataSource i can get one DataSource with multiple Tracks, but i'm not able to send them out together over one RTP-Session.
Is there any way to do this?
I have also thaught about using RawBufferMux or WAVMux to multiplex the tracks included in the MergingDataSource ,but i cant find out how to use them.
Or is there an easyer way to create an conference over RTP?
I need to connect to cisco ip-phones, so i can only use the g711_ulaw codec.
Is there anyone out who can help me?

Im sorry, but I met this problem in past and find it impossible to resolve.
Im also thought about MergingDataSource, about Many2One class and more and more but nothing I could use. And nobody could answer this question in this forum half of year ago.
Oh, nearly forgot: it is possible to write merged streams to file, seems possible to write them into file and then to transmit it to net from file. But how to realize more than one Processor?..

Can I play and save the incoming RTP streams simultenously

Hi,
I want to whether i can play and save the incoming RTP stream simultenously using
JMF 2.1. The idea is by saving in the file, i can playback the same file later time
Is there any example code aviliable.
Thanks in Advance
bye
Srikanth

I think this, today, it's impossible lamentably. By but that I have tried it, I have not been able to obtain it.
If exists someone that it has obtained it, please, help us with this subject. It's very important.
Thanks

Receiving Video RTP Stream (JMF) in JME ( MMAPI ) - URGENT !!!

Hi Folks...
I�m trying to develop an application that sends the images from a web cam connected to the computer, to the PDA that the images can be viewed by the user...
My code for the JMF RTP Video Stream is as follows.
Processor proc = null;
          javax.media.protocol.DataSource ds = null;
          TrackControl[] tc = null;
          int y;
          boolean encodingOk = false;
          Vector<javax.media.protocol.DataSource> datasources = new Vector<javax.media.protocol.DataSource>();
          for( int x = 0 ; x < camerasInfo.length ; x++ ){
               try {
                    proc = Manager.createProcessor(camerasInfo[x].getLocator());
               }catch (NoProcessorException e) { System.out.println("Erro ao int�nciar PROCESSOR: " + e);     }
                catch (IOException e) { System.out.println("Erro ao int�nciar PROCESSOR: " + e); }
               proc.configure();
               try {
                    Thread.sleep(2000);
               } catch (InterruptedException e1) {
                    // TODO Auto-generated catch block
                    e1.printStackTrace();
               proc.setContentDescriptor( new ContentDescriptor(ContentDescriptor.RAW_RTP) );
               tc = proc.getTrackControls();
               for( y = 0; y < tc.length ; y++ ){
                    if (!encodingOk && tc[y] instanceof FormatControl){
                         if( ( (FormatControl) tc[y] ).setFormat( new VideoFormat(VideoFormat.RGB) ) != null ){
                              tc[y].setEnabled(true);
                         }else{
                              encodingOk = true;
                    }else{
                         tc[y].setEnabled(false);
               proc.realize();
               try {
                    Thread.sleep(2000);
               } catch (InterruptedException e1) {
                    e1.printStackTrace();
               try{
                    ds = proc.getDataOutput();
               }catch(NotRealizedError e){
                    System.out.println("ERRO ao realizar datasource: " + e);
               }catch(ClassCastException e){
                    System.out.println("Erro ao realizar datasource: " + e);
               datasources.add(ds);
               System.out.println( ds.getLocator() );
               encodingOk = false;
          MediaLocator ml = new MediaLocator("rtp://10.1.1.100:99/video");
          try {
               DataSink datSink = Manager.createDataSink(ds , ml);
               datSink.open();
               datSink.start();
          } catch (NoDataSinkException e) {
               System.out.println("Erro ao instanciar DataSink: " + e );
          } catch (SecurityException e) {
               System.out.println("Erro ao iniciar DataSink: " + e);
          } catch (IOException e) {
               System.out.println("Erro ao iniciar DataSink: " + e);
          }I�m not sure if this code is correctly... it is ?
So... the next part of the systems runs on the PDA..
The code that access this RTP Stream is as follows..
          VideoControl c = null;
          try {
               player = Manager.createPlayer("rtp://10.1.1.100:99/video");
               c = (VideoControl) player.getControl("VideoControl");
               tela = (Item) c.initDisplayMode( GUIControl.USE_GUI_PRIMITIVE, null);
               player.start();
               append(tela);
          } catch (IOException e) {
               str.setText(e.toString());
               append( str );
          } catch (MediaException e) {
               str.setText(e.toString());
               append( str );
          }So when the APP try to create a player for "rtp://10.1.1.100:99/video" an MediaException is throwed..
javax.microedition.media.MediaException: Unable to create Player for the locator: rtp://10.1.1.100:99/video
So... I don�t know what happen =/
The error is in the PDA module ? Or in the computer�s initialization off the RTP Video Streaming ?
I need finish this job at next week... so any help is usefull..
Waiting for answers
Rodrigo Kerkhoff

First of all: before going onto the j2me part, make sure the server works, before doing anything else! Apparently, it doesn't...
The MediaLocator is generally used to specify a kind of URL depicting where the data put into the datasink should go to. In your case, This cannot be just some URL where you want it to act as a rtps server. You'll need to implement that server yourself I guess.

Adding Effect on an incoming H.263/RTP stream

Hi,
I am playing an h.263/rtp stream which is multicasted over a network. I want to play the file with rotation or some other effect added to it. I tried to see following link for some help(as the forums suggested):- http://java.sun.com/products/java-media/jmf/2.1.1/solutions/RotationEffect.html. But the link is disabled and new link which they are providing does not contain the example.
Can any of me help me how to add the effect on h263/rtp.
Though i already created my effect class and tried adding it. But could not add the effect.
Kindly give me some directions.

Is there some error in above code or if there is some other issue.There's an error in the above code because you're assuming I'm being imprecise with my language. I am not.
You must copy the data from the input buffer to the output buffer...
Buffer objects are wrappers around byte[], and you cannot copy data from one array to another by modifying the object references...which is what your code does.
public int process(Buffer inBuffer, Buffer outBuffer) {
    // Make sure the output buffer will hold data
    if (!(outBuffer.getData() instanceof byte[])) {
        outBuffer.setData(new byte[inBuffer.getLength()]);
        outBuffer.setLength(inBuffer.getLength());
        outBuffer.setOffset(0);
    // Make sure the output buffer is long enough
    if (outBuffer.getLength() + outBuffer.getOffset() < inBuffer.getLength())) {
        int offset = outBuffer.getOffset();
        int length = inBuffer.getLength() + outBuffer.getOffset();
        byte[] oldData = (byte[])outBuffer.getData();
        byte[] newData = new byte[length];
        // Copy the previous data
        for (int i = 0; i < offset; i++) {
            newData[i] = oldData;
// Set the variables for the output buffer
outBuffer.setData(newData);
outBuffer.setLength(length);
outBuffer.setOffset(offset);
// Copy the data from in to out
int j = outBuffer.getOffset();
for (int i = inBuffer.getOffset(); i < inBuffer.getOffset() + inBuffer.getLength(); i++) {
((byte[])outBuffer.getData())[j++] = ((byte[])inBuffer.getData())[i];
return BUFFER_PROCESSED_OK;
That's untested code above, fix my dumb mistakes as necessary.

Having multiple threads for receiving RTP streams

Hello,
Developing an audio conference server, I have come to think that if I manage to separate the different audioReceivers who receive the RTP streams the performance could improve.
At this moment I have the main program, so the main thread let�s say, who initializes a new audioRx object for each remote client.
Would separation of the different receivers into different threads improve the applications performance?
has anyone thought of this? or done something similar?
Thanks for your help.
bgl

i need help from about the RTP stream from the same port

Cisco RTP Stream Capture Utility

Hi everybody
I'm working on VoIP CME
& as i want to identify the noises on my line during conversations.
I read in the current Cisco doc that there are a Cisco Utility to capture RTP stream to be able to send it to the TAC.
http://www.cisco.com/en/US/tech/tk652/tk698/technologies_white_paper09186a00801545e4.shtml
Anybody knows where i can download it ? or link to this tool ?
thx a lot
have a nice day

Ethereal is one tool that will allow you to do this.
http://www.ethereal.com/
Hope this helps. If so, please rate the post.
Brandon

Simultaneous Recording in file and playing incoming rtp stream

I want to Simultaneous Recording in file and playing incoming rtp stream.I have cloned the data source. After cloning I created two data sinks and two processors.One processor/datasink is meant for recording the stream in a file and other processor/data sink is used to play the rtp through speaker.
But at a time I am only able to achieve one operation i.e either recording or playing.For recording into the wav file I have to put Thread.sleep(30) so that I can record the rtp into the file.During this time I am not able to play the audio through speaker.I also create separate thread but this does not help.
Please give suggestion on how to resolve this issue.

thanks for the reply, but I think I've already solved the problem.

Merging RTP streams and writing to file

I'm am in the middle of project using JMF i'm trying to merge incoming RTP streams and write them to file but i always have problem with formats. I have created processor with merged DataSource i set ContentDescriptor to different FileTypeDescriptors and tried to transcode the RTP streams to different formats, but always processor couldnt realize because of this error:
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
Could someone tell me what content descriptors i need to use and what formats to solve this problem. Any help will be appreciated.
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
Hope this help you.
Francois

Using RTP stream as data source

I have successfully written a program that reads G.711 audio from a wav file and sends it out over the network.
I'm trying to modify that program so that instead of getting its source audio from a file, it instead receives the audio from an RTP stream. So, in effect, it is simply receiving audio from one socket and sending it back out another.
The error I'm encountering:
[java] javax.media.NoProcessorException: Cannot find a Processor for: rtp://172.30.18.140:32916
[java] at javax.media.Manager.createProcessorForContent(Manager.java:1663)
[java] at javax.media.Manager.createProcessor(Manager.java:627)
Here is the code fragment where the exception occurs:
processor = javax.media.Manager.createProcessor(new MediaLocator("rtp://172.30.18.140:32916"));
Can anybody help? Thanks!
<removed funky formatting><br>
Message was edited by:
feh

Hi,
I Had a very similar problem, but in addiction I had to store data in a buffer before retransmission. I resolved it employing low level classes. At this time, I still do not know any other solution, and nobody helped me in any way, also in this forum...
So, first you have to access single frames of the stream (Buffer or ExtBuffer objects) using a RawBufferParser, then you have to reconstruct a DataSource by multiplexing the frames (use an appropriate Multiplexer class, such as RTPSynchBufferMux).
The big problem is that there is no documentation avalaible about how to use these classes: I have lerned all I know by "reverse engineering" (lots of hours spent in reading very long annoying code). A little help can be given by the PlugIn Viewer of JMStudio, that tells you which classes are to be employed.
I have not tried to retransmit data "straightforward", without access single frames, because that was not my task. Maybe if you use RTPManager, it is possible to get a DataSource by the ReceiveStream object, and then to pass it to another RTPManager to create a SendStream.
I do not know any simpler way to do that, but this does not mean that it does not exist...!
good luck.
Alberto M. (Italy)

RTP-Streaming (MP3)

Dear Java-Users,
we like to build a little RTP-Streaming radio.
We found a lot of code samples for example the AVTransmit/Recieve from java.sun.
Our client is working well and is able to recieve RTP-Streams.
But our server isn't running at the moment.
When we start both (server and client) on one local machine, the client can recieve the Stream. When we start the server on an other machine from the same network, the client cannot recieve the stream.
Well there are two problems left :
1) As I said the server does not work except from localhost.
2) We read that there is no MP3 support on Java-RTP-Streaming. But as you know MP3 is the most used audio-format and that's why we like to implement it.
If somebody could help me, I would be really glad.
Please send me an email or post here. ([email protected])
I appreciate for your time. with good greetings,
Tobias Belch

Really I have to pay to use MP3 ?
No I didn't knew this.
I want to use MP3, for I want to build a simple and small client/server for RTP-Streaming over a standard network. And MP3 is the most used format and that's why it is just comfortable for the user to use mp3.
But thank you very much for your hint.
greetings

Help forwarding an RTP stream

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