RTP streaming

Hi guys,
what I need to do is to stream YUV images between two JMF applications. I am using VideoTransmit application (taken from this site) to transmit and JMStudio to receive. From a little research I have noticed the following to transmit:
1) Create the DataSource using the Manager.CreateDataSource(MediaLocator)
2) create processor using the above created datasource
3) create player etc.
My question now is the following:
How can I transmit a YUV file? How can I create the DataSource which has this YUV file without having problems for the further creation of processor etc?
Thanks in advance guys,
Chris

Ok, I think I know what was confusing me. Apparently the setRate() isn't effecting the rate of stream. The audio file I am streaming runs about five minutes usually, so I sped the rate up to 10. It sped the process of writing to a file up significantly, but writing to an RTP stream seems to take the full run-time of the file. Does this sound right or is there something else I'm missing?
Thanks,
Khanathor

Similar Messages

How to extract data from Buffer and create a RTP stream

Hi
I'm working on a project where I need to interrupt a media stream (video and audio) and extract the data. Send the data with a custom protocol. On the recieveing side I would like to reconstruct the stream using only the data chunks.
I'm currently looking at [DataSourceReader.java|http://java.sun.com/javase/technologies/desktop/media/jmf/2.1.1/solutions/DataSourceReader.java] and more specifically at a method like printDataInfo(Buffer buffer) to extract the data.
Is it possible to create a RTP stream, only having access to the byte array "data" in Buffer ?
Thanks in advance.

camelstrike wrote:
Hi
I'm working on a project where I need to interrupt a media stream (video and audio) and extract the data. Send the data with a custom protocol. On the recieveing side I would like to reconstruct the stream using only the data chunks.
I'm currently looking at [DataSourceReader.java|http://java.sun.com/javase/technologies/desktop/media/jmf/2.1.1/solutions/DataSourceReader.java] and more specifically at a method like printDataInfo(Buffer buffer) to extract the data.
There are a couple of different ways to get the data. Reading it from inside a DataSink is perfectly fine...
Is it possible to create a RTP stream, only having access to the byte array "data" in Buffer ?Yes and no.
[http://java.sun.com/javase/technologies/desktop/media/jmf/2.1.1/solutions/CustomPayload.html]
You need to know the format of the media in addition to the actual media data...

Writing a conference server for RTP streams

Hello,
I'm trying to write a conference server which accepts multiple RTP streams (one for each participant), creates a mixed RTP stream of all other participants and sends that stream back to each participant.
For 2 participants, I was able to correctly receive and send the stream of the other participant to each party.
For 3 participants, creating the merging data source does not seem to work - i.e. no data is received by the participants.
I tried creating a cloneable data sources instead, thinking that this may be the root cause, but when creating cloneable data sources from incoming RTP sources, I am unable to get the Processor into Configured state, it seems to deadlock. Here's the code outline :
    Iterator pIt = participants.iterator();
    List dataSources = new ArrayList();
    while(pIt.hasNext()) {
      Party p = (Party) pIt.next();
      if(p!=dest) {
        DataSource ds = p.getDataSource();
        DataSource cds = Manager.createCloneableDataSource(ds);
        DataSource clone= ((SourceCloneable)cds).createClone();
        dataSources.add(clone);
    Object[] sources = dataSources.toArray(new DataSource[0]);
    DataSource dataSource =   Manager.createMergingDataSource((DataSource[])sources);
    Processor p = Manager.createProcessor(dataSource);
    MixControllerListener cl = new MixControllerListener();
    p.addControllerListener(cl);
    // Put the Processor into configured state.
    p.configure();
    if (!cl.waitForState(p, p.Configured)) {
        System.err.println("Failed to configure the processor.");
        assert false;
    }Here are couple of stack traces :
"RTPEventHandler" daemon prio=1 tid=0x081d6828 nid=0x3ea6 in Object.wait() [98246000..98247238]
        at java.lang.Object.wait(Native Method)
        - waiting on <0x9f37e4a8> (a java.lang.Object)
        at java.lang.Object.wait(Object.java:429)
        at demo.Mixer$MixControllerListener.waitForState(Mixer.java:248)
        - locked <0x9f37e4a8> (a java.lang.Object)
        at demo.Mixer.createMergedDataSource(Mixer.java:202)
        at demo.Mixer.createSendStreams(Mixer.java:165)
        at demo.Mixer.createSendStreamsWhenAllJoined(Mixer.java:157)
        - locked <0x9f481840> (a demo.Mixer)
        at demo.Mixer.update(Mixer.java:123)
        at com.sun.media.rtp.RTPEventHandler.processEvent(RTPEventHandler.java:62)
        at com.sun.media.rtp.RTPEventHandler.dispatchEvents(RTPEventHandler.java:96)
        at com.sun.media.rtp.RTPEventHandler.run(RTPEventHandler.java:115)
"JMF thread: com.sun.media.ProcessEngine@a3c5b6[ com.sun.media.ProcessEngine@a3c5b6 ] ( configureThread)" daemon prio=1 tid=0x082fe3c8 nid=0x3ea6 in Object.wait() [977e0000..977e1238]
        at java.lang.Object.wait(Native Method)
        - waiting on <0x9f387560> (a java.lang.Object)
        at java.lang.Object.wait(Object.java:429)
        at com.sun.media.parser.RawBufferParser$FrameTrack.parse(RawBufferParser.java:247)
        - locked <0x9f387560> (a java.lang.Object)
        at com.sun.media.parser.RawBufferParser.getTracks(RawBufferParser.java:112)
        at com.sun.media.BasicSourceModule.doRealize(BasicSourceModule.java:180)
        at com.sun.media.PlaybackEngine.doConfigure1(PlaybackEngine.java:229)
        at com.sun.media.ProcessEngine.doConfigure(ProcessEngine.java:43)
        at com.sun.media.ConfigureWorkThread.process(BasicController.java:1370)
        at com.sun.media.StateTransitionWorkThread.run(BasicController.java:1339)
"JMF thread" daemon prio=1 tid=0x080db410 nid=0x3ea6 in Object.wait() [97f41000..97f41238]
        at java.lang.Object.wait(Native Method)
        - waiting on <0x9f480578> (a com.ibm.media.protocol.CloneableSourceStreamAdapter$PushBufferStreamSlave)
        at java.lang.Object.wait(Object.java:429)
        at com.ibm.media.protocol.CloneableSourceStreamAdapter$PushBufferStreamSlave.run(CloneableSourceStreamAdapter.java:375)
        - locked <0x9f480578> (a com.ibm.media.protocol.CloneableSourceStreamAdapter$PushBufferStreamSlave)
        at java.lang.Thread.run(Thread.java:534)Any ideas ?
Thanks,
Jarek

bgl,
I was able to get past the cloning issue by following the Clone.java example to the letter :)
Turns out that the cloneable data source must be added as a send stream first, and then the clonet data source. Now for each party in the call the conf. server does the following :
Party(RTPManager mgr,DataSource ds) {
      this.mgr=mgr;
      this.ds=Manager.createCloneableDataSource(ds);
   synchronized DataSource cloneDataSource() {
      DataSource retVal;
      if(getNeedsCloning()) {
        retVal = ((SourceCloneable) ds).createClone();
      } else {
        retVal = ds;
        setNeedsCloning();
      return retVal;
    private void setNeedsCloning() {
      needsCloning=true;
    private boolean getNeedsCloning() {
      return needsCloning;
     private synchronized void addSendStreamFromNewParticipant(Party newOne) throws UnsupportedFormatException, IOException {
    debug("*** - New one joined. Creating the send streams. Curr count :" + participants.size());
    Iterator pIt = participants.iterator();
    while(pIt.hasNext()) {
      Party p = (Party)pIt.next();
      assert p!=newOne;
      // update existing participant
      SendStream sendStream = p.getMgr().createSendStream(newOne.cloneDataSource(),0);
      sendStream.start();
      // send data from existing participant to the new one
      sendStream = newOne.getMgr().createSendStream(p.cloneDataSource(),0);
      sendStream.start();
    debug("*** - Done creating the streams.");So I made some progress, but I'm still not quite there.
The RTP manager JavaDoc for createSendStream states the following :
* This method is used to create a sending stream within the RTP
* session. For each time the call is made, a new sending stream
* will be created. This stream will use the SDES items as entered
* in the initialize() call for all its RTCP messages. Each stream
* is sent out with a new SSRC (Synchronisation SouRCe
* identifier), but from the same participant i.e. local
* participant. <BR>
For 3 participants, my conf. server creates 2 send streams to every one of them, so I'd expect 2 SSRCs on the wire. Examining the RTP packets in Ethereal, I only see 1 SSRC, as if the 2nd createSendStream call failed. Consequently, each participany in the conference is able to receive voice from only 1 other participant, even though I create RTPManager instance for each participany, and add 2 send streams.
Any ideas ?
Thanks,
Jarek

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

How to synchronize audio and video rtp-streams

Hi everyone!
I've tried to make one of the processors, processor1, control the other processor, processor2, with processor1.addController(processor2) but I get a
javax.media.IncompatibleTimeBaseException. I need to synchronize audio and video rtp-streams because they are sometimes completely out of sync.
In JMF API Guide I've read:
"2. Determine which Player object's time base is going to be used to drive
the other Player objects and set the time base for the synchronized
Player objects. Not all Player objects can assume a new time base.
For example, if one of the Player objects you want to synchronize has
a *push-data-source*, that Player object's time base must be used to
drive the other Player objects."
I'm using a custom AVReceive3 to receive rtp-streams and then I create processors for the incoming stream's datasource's, and they are ALL PushBufferDataSource's.
Does this mean I can't synchronize these. If so is there any way I can change them into Pull...DataSources ?
How are you supposed to sync audio and video if not with as above ?

camelstrike wrote:
Hi everyone!
I've tried to make one of the processors, processor1, control the other processor, processor2, with processor1.addController(processor2) but I get a
javax.media.IncompatibleTimeBaseException. I need to synchronize audio and video rtp-streams because they are sometimes completely out of sync.
In JMF API Guide I've read:
"2. Determine which Player object's time base is going to be used to drive
the other Player objects and set the time base for the synchronized
Player objects. Not all Player objects can assume a new time base.
For example, if one of the Player objects you want to synchronize has
a *push-data-source*, that Player object's time base must be used to
drive the other Player objects."
I'm using a custom AVReceive3 to receive rtp-streams and then I create processors for the incoming stream's datasource's, and they are ALL PushBufferDataSource's.
Does this mean I can't synchronize these. If so is there any way I can change them into Pull...DataSources ?The RTP packets are timestamped when they leave, and they are played in order. You can't change the timebase on an RTP stream because, for all intensive purposes, it's "live" data. It plays at the speed the transmitting RTP server wants it to play, and it never gets behind because it drops out of order and old data packets.
If your RTP streams aren't synced correctly on the receiving end, it means they aren't being synced on the transmitting end...so you should look into the other side of the equation.

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])

How can I manage controls in an RTP Streaming

Hello,
I'm currently using an RTP Streaming in order to play some music. I started with AVReceive2 and AVTransmit2 from sun and by adding a plugin, I managed to play MP3. So far so good.
Because I didn't want to use view components given by sun, I created my own GUI for my player and I use it instead of using the one in sun's examples.
I use listeners for my "buttons" (I prefer using mouselistener and Panel like in the sun's example Jamp) but I can't find a way to execute those buttons.
If I press pause, it has to pause the streaming in AVTransmit2.
If I press next, or previous, it has to change the process AVTransmit2.
Each control has to be done in AVTransmit2 but my player is part of AVReceive2.
My first guess was to check if I could use an event to tell my AVTransmit2 object to execute my controls but I havn't found a way to tell my AVTransmit2 object that it was this button I pressed.
I finally stopped trying transmitting my orders thanks to the events.
I tried then to find a way to get my AVTransmit2 object in my AVReceive2 object so that I could call methods but i failed :(
How can i manage my controls so that it will call methods on my server and not my client ?
Thanks :)
Shad.
Edited by: Shadwolf on Feb 9, 2010 2:15 AM

Hi, again,
For the next & previous button, I was thinking in something.
Do you think it could be good to create a new class RTPClientManager which extends from RTPManager and get 2 booleans previous & next that are set to true when I press on buttons ?
From here, couldn't I modify my function in AVTransmit2 like this (I implements ReseiveStreamEvent on AVTransmit2) :
@Override
     public void update(ReceiveStreamEvent evt)
          RTPClientManager mgr = (RTPClientManager )evt.getSource();
          Participant participant = evt.getParticipant();     // could be null.
          ReceiveStream stream = evt.getReceiveStream(); // could be null.
           * Détection de la fermeture de connexion du client afin de fermer la transmission streaming
          if (evt instanceof ByeEvent)
               System.err.println(" - Got \"bye\" from: " + participant.getCNAME());
                     if(mgr.isNext())
                 this.stop();
                         /* SOME STUFF FOR NEXT */
                    else if(mgr.isPrevious())
                 this.stop();
                         /* SOME STUFF FOR PREVIOUS*/
                else //means it's the stop button called
                          this.stop() ;
     }Would it work ?
If you have better ideas I am ready to hear theam :P
But if this works, I will manage stop, next and previous, but how could I manage play & pause button ?
I can't find a proper solution to manage my controls :(

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.

RTP stream being stripped

How can I check to see if 802.1Q VLAN tagging information is being stripped from the RTP stream by the workstation's NIC?

Some network cards strip the dot1q headers, so your packet capture won't have this data. If you're not seeing dot1q headers please check the following link from wireshark:
http://wiki.wireshark.org/CaptureSetup/VLAN#head-81781716144f2855ab0aff2f8b752e95f2562efb
Apply the previous settings to the machine you're capturing with.

Sequence Error in RTP Stream

What anomalies are we expecting if we are getting Sequence error and high Delta and Jitter values in RTP stream.

You can expect voice quality issues mostly choppy voice.
Regards,
Anup

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

Setting RTP Stream Properties FrameRate/Size/Format

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?

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?

Evesdrop on RTP stream

Hi,
I was wondering does anyone know good tools (for WIndows) to reconstruct captured RTP stream into audible format?
I am trying to assess the level of IPT security in our Enterprise and one thing I would like to show is how easy it is for hackers to capture unencrypted RTP traffic, reconstruct the conversation and play it out.
I think that such tools are available, so I am just looking for the recommended ones.
Thanks,
David

There are freeware sniffers like ettercap that can be used to perform man in the middle attacks and capture voip traffic. I dont remember the tool available in linux that can decode a voip sniffer dump.i will see if i can find the name for it.
Its not at all hard for a person with enough knowledge to be dangerous to sniff on your network. Again voip security is something that very much depends on your lan network security as well.
HTH
Sankar

RTP streaming

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