Firewall latency for rtp streams

Does anyone have test info on PIX or other firewall latency changes as the number of voice or video connections increases? 1, 10, 100, 1000 connections? Any reference documents? Thanks in advance.

You can refer to the document on Low Latency Queuing (LLQ) for IPSec Encryption Engines
http://www.cisco.com/en/US/products/ps6350/products_configuration_guide_chapter09186a00804dfa7f.html

Similar Messages

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

No Inter-Cluster RTP Stream with Gatekeepers

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
Intra-Cluster Traffic Works
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.
Inter-Cluster Traffic Fails (GK to GK)
Client C -> Client A this works, the RTP stream comes up and the call functions as expected.
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
CS: notify
RAS: registrationRequest
RAS: registrationConfirm
CS: notify
CS: connect
CS: notify
CS: releaseComplete
RAS: disengageRequest (DISCONECT_REASON=2,TIME=1321266127,DURATION=24,DISCONNECT_STRING=no resource,ORIGIN=0,LINE_NUMBER=GK,OUTBUND_GW_IP=..
RAS: disengageConfirm
There are firewalls inbetween and these were the first thing I looked at, but I dont even see any RTP stream trying to be initiated from the far side. Would anyone have any ideas where I could start looking?
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.

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

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

Get a event for a new received rtp stream in a player

Hi!
I'm trying to implement a RTP-player, that receives a AV-stream and play it. The special thing about this player should be, that even if the stream interrupts, the player wait's on the same IP and port for a new stream and open it in the SAME frame (not like jmstudio in new window).
I try to catch the "ReceiveStreamEvent", so i can restart the player, but i don't get eny events for this. I tried to do it with a "RTPManager", but i don't know how.
Does anybody has a example how to get the "ReceiveStreamEvent", so i know, the RTP-stream has been interrupted?
Thanks
Adam

See AVReceive2 in JMF Solutions, in JMF web site

Send RTP stream to NAT address

Hi,
i want to transmit a RTP stream from a server to a host in a LAN.
This host has a NAT address and it's non real IP address, so i can't send any stream trought usage of SessionManager API because it need to know a public IP.
The other issue is that in a LAN, in most popular cases, there is a firewall that close the connection from internet to their hosts.
I think this solution:
1) LAN's hosts can intiate the connection with server sending a non real RTP data
2)Server store the SessionManager of this connection
3)server can send your RTP stream now
Someone have a more good solution or any suggestion?
Thank for all
[email protected]

I have one appletTransmitter that capture video from webcam and transmit it to other client on internet.
I try to transmit medialocator from appletTransmitter to servlet1 and then save MedialLocator as servlet attribute, then other client can connect to servlet2 that send saved MediaLocator to appletClient.
APPLETTRANSMITTER:
URL url=null;
MediaLocator media=new MediaLocator("vfw://0");
try{
url = new URL("http://localhost:8080/servlet1");
catch(MalformedURLException mue){mue.printStackTrace();}
URLConnection conn=null;
try{
conn = url.openConnection();
catch(IOException ioe){ioe.printStackTrace();}
conn.setDoOutput(true);
OutputStream os=null;
ObjectOutputStream oos=null;
InputStream in=null;
ObjectInputStream iin=null;
MediaLocator mResp=null;
String r=null;
try{
os=conn.getOutputStream();
oos=new ObjectOutputStream(os);
oos.writeObject(media);
//oos.writeObject("Prova Servlet");
oos.flush();
catch(IOException io){io.printStackTrace();}
catch(ClassNotFoundException cn){cn.printStackTrace();}
SERVLET1
ObjectInputStream objin = new ObjectInputStream(request.getInputStream());
MediaLocator ml =null;
try{
ml = (MediaLocator) objin.readObject();
context.setAttribute("media",ml);
catch(ClassNotFoundException e)
{e.printStackTrace()}
But on servlet1 there is a ClassNotFoundException: MediaLocator
What do we think about the solution and exception problem?
Best Regards,
Nico from Italy

Problem while sending RTP stream to QuickTime player

Hello,
I am trying to send a RTP video stream from a local file to a Quicktime player.
The player is waiting for the stream in its default ports (6970/6971). I archieve this by using my own RTSP server.
I start the transmission by using a processor and a datasink. It seems that I send one or two RTCP packages to port 6971, but not RTP packages are sent.
This is my code:
pr = Manager.createProcessor( new MediaLocator("file:/rootDirectory/file.mpg"));
pr.configure();
while(pr.getState()==Processor.Configuring){}
TrackControl[] tracks = pr.getTrackControls();
tracks[0].setFormat(new VideoFormat(VideoFormat.JPEG_RTP));
tracks[0].setEnabled(true);
pr.realize();
while(pr.getState()==Processor.Realizing){}
DataSource ds = null;
ds = pr.getDataOutput();
String url1 = "rtp://193.147.59.231:6970/video/10"
MediaLocator m1 = new MediaLocator(url1);
DataSink d1 = Manager.createDataSink(ds, m1);
d1.open();
d1.start();JMF doesn't show any error, but my application just is not sending RTP packages. What could be wrong in my code?
Kind regards.

Any clue please? I am frustrated!!

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])

AVReceive2 gives error: No format has been registered for RTP Payload type

Hello,
I am using the AVReceive2 program (http://java.sun.com/products/java-media/jmf/2.1.1/solutions/AVReceive.html#requirements) as posted to hopefully play back RTP in real time. I have it set to receive a known good RTP stream and I receive the following error:
No format has been registered for RTP Payload type 8
I have seen posts for a similar error but it seemed to be in the context of an A-law add on which am I not doing. Am I missing a plug-in or modification to the code?
thanks in advance,
Erich

Hi Erich,
I have the same problem when I using AVReceive2.java .Could yuo describe your solution in order to solve the issue?
In order to create the RTP stream I used VLC and is impossible to change the encapsulation method and then the payload type .
What kind of streaming server are yuo used?
My e-mail address is [email protected]
If you can , please write me your proceeding in order to solve the problem.
Thank you

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.

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

Firewall latency for rtp streams

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