RTP Packet Data

Hi again,
As anyone who's been following my posts can tell, I've been working on an RTP application. I've got a Server-Client program set up (mainly it's just the AVTransmit and AVReceive stuff on this website) and I'm looking for a bit of code that will tell the clien that RTP packets are arriving
eg
while ( RTPArriving () == true ) System.out.println ( "RTP Arriving" );
Does any such code exist?
Thanks ;-)

This is copied directly from AVReceive2.java, which
you claim to base your app on...
if (evt instanceof RemotePayloadChangeEvent) { ... }
else if (evt instanceof NewReceiveStreamEvent) { ...
else if (evt instanceof StreamMappedEvent) { ... }
else if (evt instanceof ByeEvent) { ... }Might this be what you where searching, but not
lookin, for? Did you look at all?Yes, I looked through the code. The code you printed above tells you that, yes a Stream event of some sort or a "bye" event has occured. It doesn't tell you how many RTP packets are arrving. As far as I'm aware it just tells you that a Stream is open.
Is it actually possible to inspect or count the number of RTP packets arriving at all?

Similar Messages

How to use jmf convert the rtp packet (captured by jpcap) in to wav file?

I use the jpcap capture the rtp packets(payload: ITU-T G.711 PCMU ,from voip)
and now I want to use JMF read those data and convert in to wav file
How to do this? please help me

pedrorp wrote:
Hi Captfoss!
I fixed it but now I have another problem. My application send me this message:
Cannot initialize audio renderer with format: LINEAR, Unknown Sample Rate, 16-bit, Mono, LittleEndian, Signed
Unable to handle format: ALAW/rtp, Unknown Sample Rate, 8-bit, Mono, FrameSize=8 bits
Failed to prefetch: com.sun.media.PlaybackEngine@1b45ddc
Error: Unable to prefetch com.sun.media.PlaybackEngine@1b45ddc
This time the fail is prefetching. I have no idea why this problem is. Could you help me?The system cant play an audio file / stream if it doesn't know the sample rate...somewhere along the way, in your code, the sample rate got lost. Sample rates are highly important, because they tell the system how fast to play the file.
You need to go look through your code and find where the sample rate information is getting lost...

DVI/RTP packet decode

Hi,
I need to stream audio and/or video to a PDA device. There is a trick here which is:
The PDA must receive the stream from a multicast address. For this I have implemented a Bridge application which joins the multicast group on behalf of the PDA and receives the Multicast RTP packets (which are sent from JMStudio) and Unicasts them to the PDA.(HP iPAQ) I had no problem implementing this. The streaming is done using JMStudio player which encodes the streaming audio data into a number of encodings (DVI/RTP in my case). I choose DVI/RTP and stream a .wav audio file.
Now I have to accept the packets and play the stream on the PDA.
The j2me application receives all the RTP packets successfully and I can extract usefull information from the packets such as: Timestamp, sequence number, payload type. The payload type is 5 which means it is a DVI4 encoding.
I use the following method to decode the samples:
public int decode(Object state, byte[] input, int inp, int len, short[] output, int outp) {
int sign;
int delta;
int vpdiff;
//int valprev = audio.Convert.byte2short(input, inp);
//int index = input[inp + 2];
int valprev=0,index=0;
int inputbuffer = 0;
int bufferstep = 0;
valprev = input[0] <<8;
valprev |= input[1] &0xff;
index = input[2] &0xff;
if ( index < 0 ) index = 0;
else if ( index > 88 ) index = 88;
int step = stepsizeTable[index];
inp += 4;
len = (len - 4) * 2;
int count = len;
while(count-- > 0) {
if ( 0 == bufferstep ) {
inputbuffer = input[inp++];
delta = (inputbuffer >> 4) & 0xf;
bufferstep = 1;
} else {
delta = inputbuffer & 0xf;
bufferstep = 0;
index += indexTable[delta];
if ( index < 0 ) index = 0;
else if ( index > 88 ) index = 88;
sign = delta & 8;
delta = delta & 7;
vpdiff = step >> 1;
if ( (delta & 4) == 4 ) vpdiff += (step << 2);
if ( (delta & 2) == 2 ) vpdiff += (step << 1);
if ( (delta & 1) == 1 ) vpdiff += step;
vpdiff >>= 2;
if ( 0 != sign )
valprev -= vpdiff;
else
valprev += vpdiff;
if ( valprev > 32767 )
valprev = 32767;
else if ( valprev < -32768 )
valprev = -32768;
step = stepsizeTable[index];
output[outp++] = (short) valprev;
((AdpcmState)state).valprev = valprev;
((AdpcmState)state).index = index;
return len;
which stores the result into a short[] array.
I then convert this short[] array into a byte[] array with the following way:
s is the short[] array
adp is the byte array
for(int g=0,k=0;g<s.length;g++,k=k+2){
audio.Convert.short2byte(s[g],adp,k);
public static void short2byte(short ival, byte b[], int offset) {
int i;
int bits = 16;
for(i = 0; i >< 2; i++) {
bits -= 8;
b[offset + i] = (byte) ((ival >> bits) & 0xff);
The final result is loaded to the player as follows:
ByteArrayInputStream input1 = new ByteArrayInputStream(adp);
player = Manager.createPlayer(input1, "audio/x-wav");//create new player
player.addPlayerListener(this);
player.prefetch();
player.realize();
player.start();
The player begins to play but I only get horrible sounds instead of the original wave file
The player now initializes ok without any problem but I can only hear a meesed up sound rather than the original. So now I strongly believe that the problem is in the decoding of the samples of the DVI/RTP codec.

thesti wrote:
how JMF deal with RTP packet loss? since my application doesn't handle anything due to RTP packet loss, i believe that JMF has a mechanism to deal with it.It "deals" with it by having a blank spot in the rendering where that packet would have gone...

How to edit RTP Packets

Hi,
I want to add some information in RTP Packet, (in Extensions Header). How can I do that in JMF ?
Thanks in Advance,
Karthikeyan R

OK, thank you. So specifically -- if I want to prioritize all of the RTP traffic flowing out through the router, can I do it ALL with just COS and not set any QoS, profile binding etc?
So far I have enabled the COS Queue, left the default settings (where COS Priorities 6 and 7 are set to highest), then on the COS to DSCP page I have entered the value 46 into the Priority 6 and 7 boxes. All the rest I left at 0.
Unfortunately this didn't seem to solve the issue. The way I have been testing is to call our PBX from an outside line, then put myself on hold so I can hear the hold music (effectively an audio stream from the PBX server). Then I listen carefully while I run a bandwidth test from speedtest.net.
During the download test the audio (music on hold) is pretty smooth. But during the upload test (lots of data flowing outbound) the audio gets very choppy. The COS settings I've tried don't seem to improve or even change that
I assume I'm doing something wrong and/or need to involve QoS somehow?
- Keith

How to prioritize RTP Packets for VOIP Audio on RV180

Hi There,
I'm a relative newbie to more advanced networking but have managed to get our small office IP PBX running over a SIP Trunk. The only real problem we are having is choppy outgoing audio when there is other heavy outgoing traffic on the network.
My understanding is that I need to set some QoS parameters, which I have played with but it didn't seem to help much. I mostly dealt with allocating bandwidth. I now think I need to somehow prioritize the outgoing RTP packets from our PBX (which runs on a PC on our LAN) to help avoid the choppy audio. My research shows this can maybe be done with something called DSCP 46 and my router does support that -- I'm just a little confused on how to exactly set the configuration.
Our router is a Cisco rv180w. I'm thinking it should be pretty straightforward, but any guidance would be appreciated (and feel free to let me know if I'm barking up the entirely wrong tree, too!)
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OK, thank you. So specifically -- if I want to prioritize all of the RTP traffic flowing out through the router, can I do it ALL with just COS and not set any QoS, profile binding etc?
So far I have enabled the COS Queue, left the default settings (where COS Priorities 6 and 7 are set to highest), then on the COS to DSCP page I have entered the value 46 into the Priority 6 and 7 boxes. All the rest I left at 0.
Unfortunately this didn't seem to solve the issue. The way I have been testing is to call our PBX from an outside line, then put myself on hold so I can hear the hold music (effectively an audio stream from the PBX server). Then I listen carefully while I run a bandwidth test from speedtest.net.
During the download test the audio (music on hold) is pretty smooth. But during the upload test (lots of data flowing outbound) the audio gets very choppy. The COS settings I've tried don't seem to improve or even change that
I assume I'm doing something wrong and/or need to involve QoS somehow?
- Keith

How to capture rtp packet??

Hi,
can someone plz tell me how to capture rtp packets. And also can u tell me how to remove the rtp header. I want to add another header to the rtp packet. I will be glad if u can also send me some code samples too.
Thank you in advance.
bye
R.Ravi Kiran

To Capture the RTP Packet all you need to do is listen for a UDP packet on the destination address and you will be able to receive the data. The Data field is by default 256 bytes long (just to let you know).
So you will send the RTP Packet to Localhost port 4444
To capture the packet you need to be listening for a UDP packet on that Address
it would look something like this:
byte[] buf = new byte[256];
DatagramSocket socket = new DatagramSocket(4444);
DatagramPacket packet = new DatagramPacket(buf, buf.length);
socket.receive(packet);
then you just have to do what you want with the buf array
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Create one player to play RTP packets from many clients

Hi,
Am a JMF newbie and I want to create one player to play packets from many clients.
So I wrote a small UDPserver thread within the app to receive rtp packets from the clients on the LAN which in turn forwards them to the player.
I instatiated two threads one to forward RTP packets and another to forward RTCP packets which listens on RTPPort+1
The reason why i do this is that i don't want the whole internet to bombard the player with anonymous voice transmissions.So the server thread is acting as a firewall. To filter out packets from from unknown ip addresses.
this is a snippet of the player.
MY_IPADDRESS =   InetAddress.getLocalHost().getHostAddress();+
url = "rtp://" + MY_IPADDRESS + ":" + RTPPlayer.PORT + "/audio/1";
MediaLocator mrl = new MediaLocator(url);
player = Manager.createPlayer(mrl);
More code which starts the server thread
if (player != null) {
       player.addControllerListener(this);
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player.start();When the server thread receives the packet it calls its forward method to forward the packet to the player by resetting the only the IP and PORT.
public void forward(DatagramPacket rtpPacket) {
         //print out packet info to view which packets are being received
         System.out.println("forwarding "+request.getAddress() + " -> " + MY_IPADDRESS+":"+portToSend);
         //set address of packet to MY_IPADDRESS
       rtpPacket.setAddress(
               InetAddress.getByName(RTPPlayer.MY_IPADDRESS));
          //set the port to the rtp port
       rtpPacket.setPort(RTPPlayer.PORT);
       datagramSocket.send(rtpPacket);
}This works fine for two clients.
When the clients become three(c1, c2 and c3),
two clients communicate well(c1 and c2) but c3's voice cannot be heard on any other pc(c1 or c2) though it plays voice from both c1 and c2.
But System.out.println("forwarding "+request.getAddress() + " -> " + MY_IPADDRESS+":"+portToSend);in the forward() method shows that packets from all clients on each pc are being received.
Does any one have an idea why this happens?
Are the packets so many that they overwhelm the player so it discards some or all?
Is this the best way of doing this?
Just to let u know all the mics are working fine.
Thx in advance
Edited by: noryak on Oct 29, 2008 10:29 AM

THAT IS MY MAIN PROBLEM. In the future, please do a little bit of research before you shout at people trying to help you. I'm so so sorry if you find my answer bothersom because it sheds some light on the fact that you have absolutely no idea what you're doing.
Your problem is that you obviously do not understand how JMF works...and you obviously havn't bothered to do any sort of research into it.
You also don't seem to understand the concept of streaming media, concurrency, politeness, good design, proper programming, audio interleaving, or common sense.
At least i have implemented a player playing packets from 2 different clients.Yeah, you implemented a player that plays packets from 2 different clients using a horrible workaround that doesn't treat the data correctly and manages to just drop data after scaling past 2 clients.
Oh yeah, you've definately found the holy grail there. At least.
You wanna know what your player is actually doing? It's playing a peice of data from A, and then a peice of data from B. It might sound like it's playing them both at the same time, but it's not. It's playing the data from one client in the gaps where there's no data, and once you've filled up the gaps in time by adding more nodes, you'll end up with data getting dropped (and that's the best case scenerio).
my issue is that i wouldn't like to create a player for each participant imagine they were people in a conference that makes it 10 players. Please understand that if you have 10 players, you'll receive 10 times as much data as you can play with one player. You end up either having to drop 90% of your data, or having to play the data at 1/10th the speed... because you're not mixing the audio data, you're interleaving it.
I just want to use one standard port on each client so that all clients send to the same port: The RTPManager class will allow you to receive as many streams as you want on a single port.
As a matter of fact, had you bothered to play with any of the source code readily available online, you'd realize there is a file that does exactly what you want.
[http://java.sun.com/javase/technologies/desktop/media/jmf/2.1.1/solutions/AVReceive2.java]
It handles receiving multiple RTP streams from a single port, and plays them all simultaniously using an array of player objects.
Does absolutely everything you want, out of the box.
That sounds like alot of threadsIf you're concerned that it's too many threads, well, maybe you should stick to hello world and other things less scary. Concurrent data processing requires threads...one per peice of concurrent data, as a matter of fact, and you're dealing with a lot of streams of concurrent data here.

Wireshark capture rtp packets on Cisco CUBE.

Hello all,
We have this call flow and we are having intermittent DTMF issue
CUCM 10.5--->CUBE(10.1.1.10--->AVAYA(10.1.1.11)--->PSTN
I am trying to capture RTP packets between CUBE and AVAYA, How can we capture RTP packets between(10.1.1.10 and 10.1.1.11)??
I followed below steps and I can see the traffic only from AVAYA to CUBE and that too only SIP and TCP not RTP.
Router(config)# access-list 140 permit ip host 32.55.55.32 any
Router(config)# access-list 140 permit ip any host 32.55.55.32
This ACL will capture all traffic to and from this IP address.
Next we need to enable the Cisco packet monitoring service:
Router# monitor capture buffer holdpackets
Now we can filter the monitored traffic by filtering it through our access-list:
Router# monitor capture buffer holdpackets filter access-list 140
Now we need to name our particular packet capture. I have called mine "testcap"
Router# monitor capture point ip cef testcap all both
Router# monitor capture point associate testcap holdpackets
Now we can start our capture!
Router# monitor capture point start testcap
Once you think you have acquired enough packets, to stop the capture, type:
Router# monitor capture point stop testcap
Now you can export your data to your tftp server by typing in the following command. You can then open the .pcap file in Wireshark for viewing
Router# monitor capture buffer holdpackets export tftp://10.0.0.55/testcap.pcap
Once uploaded you can clear your capture buffer by typing the following:
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But when i configure the destination as USB0 my pendrive, it fails.
Could be a bug but I wouldn't recommend configuring the destination as your USB drive because no one has the same luxury as you to have the USB sit there all the time.
Store to the flash and transfer to USB is probably the best solution.

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Hi!
I have a stream of raw RTP packets comming from another program.
How can I use JMF to decode all raw RTP packets and get the media stream using JMF?
/Tec

Hello,
Have you resolved your problem?
I've tried with Jmf, but finaly i receive each stream and save raw data in a file.
Then i convert it to an audio file format. (tritonus : RawAudioConverter.java)
If you have resolved with jmf i'm interested by your solution.
try{
rtpRcvSock.receive(rcvPkt);
data = rcvPkt.getData();
          boolean extensionHeader = ( (data[0] & 0x08) == 0x08) ? true : false;
     int payloadType = data[1] & 0x7F;
     int cc = data[0] & 0x0F;
     int sequenceNumber = ( (data[2] & 0xFF) << 8) | (data[3] & 0xFF);
     int mediaStart = 3 * 4 + cc * 4;
     int extLength = 0;
          System.out.println("PayLoad = "+payloadType);
          System.out.println("Seqnum = "+sequenceNumber);
          System.out.println("Data Length = "+ (data.length - mediaStart));
     if (extensionHeader) {
     extLength = (data[mediaStart + 3] & 0xFF)
     | ( (data[mediaStart + 2] & 0xFF) << 8);
     mediaStart += extLength;
     byte[] mediaData = new byte[data.length - mediaStart];
     System.arraycopy(data, mediaStart, mediaData, 0, mediaData.length);
          ByteArrayInputStream bb = new ByteArrayInputStream(mediaData);
               AudioInputStream aiStream = new AudioInputStream(bb,format,mediaData.length);
               //AudioSystem.write(aiStream,AudioFileFormat.Type.WAVE,file);
               file.write(mediaData);
Then as i told i use RawAudioDataConverter.java
Sam

Directly sending RTP Packets

Hello,
I have a media file divided in chunks on disk.
I want to stream the whole media file with RTP. How can I do that? Can you directly send RTP Packets?
For example:
for every chunk i
   for (every 1000 bytes b[] in i)
      sendRTPPacket(b[]) //this creates a new RTP Packet with the contents of b[] and sends itIs that possible?
If that is not possible, I have another idea:
If I open a different RTP session for each chunk (so I can just specify as dataSource the actual chunk file), can I synchronize that at the receiver side? how can I then reconstruct the whole file ?
How would you do that?
Thank you!

Hello,
after searchig the internet, I found this:
http://forums.sun.com/thread.jspa?threadID=5356475&tstart=1
I think creating a customized Push DataSource will work fine, what do you think? it would push data each time a new chunk is available.
I tried the example on http://java.sun.com/javase/technologies/desktop/media/jmf/2.1.1/solutions/LiveData.html but I'm not sure how to use it: I cannot register it.
I would like to see the results on a player, so I used a player that I have:
DataSource dataSource = new DataSource();
               //dataSource.connect();
               //dataSource.start();
               player = Manager.createPlayer(dataSource);But I get Error: DataSource not connected
If I call dataSource.connect() I get:
java.lang.ClassCastException: [B cannot be cast to [I
     at com.omnividea.media.renderer.video.Java2DRenderer.bufferToImage(Java2DRenderer.java:131)
     at com.omnividea.media.renderer.video.Java2DRenderer.process(Java2DRenderer.java:105)
     at com.sun.media.BasicRendererModule.processBuffer(BasicRendererModule.java:728)
     at com.sun.media.BasicRendererModule.scheduleBuffer(BasicRendererModule.java:499)
     at com.sun.media.BasicRendererModule.doProcess(BasicRendererModule.java:400)
     at com.sun.media.RenderThread.process(BasicRendererModule.java:1114)
     at com.sun.media.util.LoopThread.run(LoopThread.java:135)
Do you have a media player where this works? or can you tell me how to use the example, maybe without explicitly registering the new dataSource, to play media in a media player?
What do you think about the idea of creating a similar DataSource (push) in order to push the data chunks that we have already into it?
Thanks!
Edited by: thawxx on Mar 11, 2009 7:20 PM

Unable to play RTP packets

I am developing a J2ME Voip Client,
For testing purposes I am using a soft phone(Zoiper) on the other end.
I am able to receive RTP packets,encoded using G711-PCMA from the Soft phone in my J2ME client and I am able to extract useful data like payload type,SSRC etc.
But I am unable to play the RTP payload.I tried in WTK ,S60 emulator and N80
Here is part of the code.
RTPPacket rtpPacket = getRTPPacket(dg, dg.getData());
buffer = rtpPacket.getData();
form.append(rtpPacket + " with media length: " + buffer.length);
ByteArrayInputStream input1 = new ByteArrayInputStream(buffer);
form.append("\n Receiving RTP Packets...");
try {
form.append("\n Trying to create player...");
player= Manager.createPlayer(input1,"audio/x-wav");
form.append("\n Player created");
player.prefetch();
player.realize();
player.start();
form.append("\n Player started");
form.append("Player duration:"+player.getDuration());
} catch(Exception e) {
e.printStackTrace();
Thanks in advance.
Ramya

I am developing a J2ME Voip Client,
For testing purposes I am using a soft phone(Zoiper) on the other end.
I am able to receive RTP packets,encoded using G711-PCMA from the Soft phone in my J2ME client and I am able to extract useful data like payload type,SSRC etc.
But I am unable to play the RTP payload.I tried in WTK ,S60 emulator and N80
Here is part of the code.
RTPPacket rtpPacket = getRTPPacket(dg, dg.getData());
buffer = rtpPacket.getData();
form.append(rtpPacket + " with media length: " + buffer.length);
ByteArrayInputStream input1 = new ByteArrayInputStream(buffer);
form.append("\n Receiving RTP Packets...");
try {
form.append("\n Trying to create player...");
player= Manager.createPlayer(input1,"audio/x-wav");
form.append("\n Player created");
player.prefetch();
player.realize();
player.start();
form.append("\n Player started");
form.append("Player duration:"+player.getDuration());
} catch(Exception e) {
e.printStackTrace();
Thanks in advance.
Ramya

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Can you browse with your phone? If yes, your problem is alrealdy solved. Nokia PC Suite (any version) wont work with S40 phones like C3. You have to download Nokia Ovi Suite for PC. You can google it and download it free. It's about 87.something MB. It works perfectly.

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Last night every thing was OK, I was able to connect to internet using my phone
but today I tried to connect but I get this msg
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I called the company and they sent me the configuration and I saved it but nothing changed .....
Any ideas ??

Straight Talk with AT&T USA (APN: tfdata) or T-Mobile US (APN: wap.tracfone)?
Lauraboop wrote:
how this all this could happen just from taking out the battery.
Well, you had to take out the battery while it was running. Then, a lot of fancy things can happen like a file-system corruption or just a SIM re-detection. The latter could load new configurations. The former would require the Nokia Software Recovery Tool on a Windows computer. Anyway, that is very, very rare – let us hope this is fixable by just changing the settings.

N80 & Orange packet data

I wonder if anyone else has had the same experience as I've had, I was just playing with my Orange N80 and when I looked in the log file I found the data counter had registered all sent data as 3.05MB and all received data as 4.72MB and all I'd done was look on Orange world a few times and checked out the sport on BBC mobile - which I know is very economical on data, however when I first got the phone I was trying out the camera and didn't realise that my pictures were being uploaded to Orange until I turned off the album feature but there weren't many photo's anyway but at £1. or so a MB this could get very expensive! I called Orange who, whilst being very polite, couldn't help very much other than to advise me to check my next bill and reset the data counter and compare the following bill! Anyway, I've since turned off just about everything as Orange seem to set all connections to a default setting of always on and I now get prompted for any connection other than for calls and sms which has the added bonus of increased battery life - not much mind but still better. Now call me cynical if you will but how many people will run up huge data bills before they realise they need to change the default settings to off which,imho, should the case anyway. Since turning off the packet data, online album etc though, I experience less problems with the handset too!

Just mention my findings on this
I too myself have seen a high amount of data being classed as transmitted via orangeinternet even though I have my accesspoint set to my home network, set the connection to only when needed and also changed the softkey menu so now says camera instead of home.
I rang orange about this and advised I was not happy the phone stated I had transfered 3mb of data in less than 2 days of using the phone, and that I don't use their webpages and if checked previous bills they would know this. I have been told that so far no data is being shown as being used, and when get my bill it will be looked into then if I am charged for anything.
Incase you may not know if you dont have the internet package you will be charged £10 per mb from orange.
I think I have now found out what the problem is, if you connect the data cable to the pc it will go into offline mode and you can transfer whatever you want. The minute you unplug the cable you go back into online mode and the orangeinternet automatically connects and whatever you have transfered by cable gets added to the orangeinternet connection.
Hopefully when get bill I should be able to dispute any charges for this, especially as I've phoned them about it.
regards
walkerx

"Subscribe to Packet Data First" (Bit of feedback)

I recently bought a Nokia 6021, which offers Bluetooth connectivity and E-mail function (send and retrieve). I got some setup advice Forum members.
Bluetooth was easy to set up, and I now use the phone in the car with a handsfree unit, which clips on the sun visor.
Very satisfactory.
E-mail was (much) more difficult to set up.
I kept getting an error: "Subscribe to Packet Data First".
It turned out that the fundamental problem lay in the fact that Vodafone Support had incorrectly given me the settings for a "Contract SIM card" when in fact I operate mine as a "Pay-As-You-Talk SIM card".
Specifically they got me to insert "Internet" as one of the Bearer Settings parameters - when it should actually be "pp.vodafone.co.uk". "pp" indicates PrePaid.
So watch that point, all you PAYT Users!
It wasn't a 6021 problem ... it was a question of identifying the right kind of SIM card.
Incidentally, when the 6021 retrieves e-mails, if the message is too big (I think above 100k), it fetches the message header, with sender's ID, etc., so that you can identify who has e-mailed you even if you can't read their message/file. Suits me perfectly.
Thanks for the advice I got from Forum members.
Best regards - Charles

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