The RTP transmission of a message is slow?

Hi
I use JMF2.2.1e in windows, and there is the transmission of a message in RTP with a wav file of PCM.
However, it takes time for about 20 seconds when a DNS server of Network Neiborhood is set till a sound is played.
It is played whether you exclude setting of DNS immediately if I fill in /etc/hosts with a transmission of a message former address.
Because I do not get it, in specifications of a program making, as for the measures mentioned above, will not there be any good method?
Transmission of a message former appointment is as follows.
String url="rtp://" + addr + ":" + port + "/audio";
MediaLocator m = new MediaLocator(url);
datasink = Manager.createDatasink(ds,m);

The dialog you are referring to is a ''shell dialog''.
The most likely reason for this to take a long time is that the default location is taking a long time to resolve, as can happen if it's a network location or if the default folder doesn't exist.
To reset the download location, reset '''browser.download.dir''' in the advanced configuration interface:
# Select Firefox's location bar (press CTRL+L or click it)
# Type '''about:config''' and press enter
# Click '''I'll be careful, I promise!'''
# in the Filter box, type '''browser.download.dir'''
# Right click the keyword.url preference below and choose '''Reset'''

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RTP Transmission over HTTPS (Internet)

Hello people!
I am currently working on my undergraduate thesis. I need to transmit the feed from a video capture device (i.e. a webcam) and also from an audio capture device towards a client in a secure transmission/session. I am already able to capture the feed from the camera and from the microphone and I also am able to transmit it in a nonsecure transmission/session.
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Thanks a lot!!

I think you might have the wrong forum. This is a forum about Forte 4GL.
ka

Help! How to change the video frame size during RTP transmission

Hi there.
In a RTP transmission:
How can I change the size of the frames of the video while I have the processor in the server running?
i.e. while the server is running, i want to change a video from 400x200 to 200x100
Thanks

I am trying to implement the same functionality. I've tried to reset the format using the trackControl.setFormat method but this only appears to be supported while the processor is in a configured state. I'm still investigating, but if you happen to know a solution please let me know.
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keep getting "not responding" message, extremely slow, have tried rebuilding data base, tried removing and re-installing. Still the same

Do you use McAfee anti-virus software?
If yes, please read:
* https://wiki.mozilla.org/Thunderbird:Testing:Antivirus_Related_Performance_Issues#McAfee
Is this an Imap mail account?
if yes, you can choose to only synchronize the most recent of emails instead of a whole folder.
see info :
* https://support.mozilla.org/en-US/kb/imap-synchronization#w_configuring-synchronization-and-disk-space-usage
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RTP Transmission always lose the last packet

Hi,
I transmit a audio file through rtp by using the JMFStudio(V 2.1.1e), then capture the rtp packets on the sending port, but the last rtp packet is missing every time.
Several developers of my team tried on their own machine and got the same result.
Did anyone try this before or if anybody can tell me how to solve it?
Thanks a lot!

Several thoughts...
If the last "packet" isn't big enough, IE, a full payload worth of data, the underlaying transport mechanism may keep it trapped in the buffer forever. If it sends packets based on a BufferFull event of some sort, then the last packet won't ever be sent because it'll get trapped in the buffer. This is a big problem with TCP sockets, and in that instance, the buffer is flushed automaticlly when a close operation is called on the socket.
I have no idea how you're managing the sending of RTP packets, but that's a possible reason.
Another possible reason would be how you're handling the EndOfMedia event. If you were to, say, stop the entire process down before the final RTP packet got sent out, then you'd also not be getting the last packet sent out. Everything in JMF pretty much spawns a new execution thread, so it's possible the main thread is shutting those other threads down before they're done doing work with calls to whatever.Stop() and whatever.Close()...
Those are just 2 ideas off the top of my head to explain why a last packet may not be sent out...

Ello, I changed my Imac hard drive which no longer worked. I installed a hard drive Western Digital 500 GB 7200 rpm SATA3 WD5002AALX 32M Black. Sometimes I see the hard disk in the utility on the Starter CD, but access seems very slow. By cons when I want

Hello, I changed my Imac hard drive which no longer worked. I installed a hard drive
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Thanks for the help
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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.
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Use Photoshop menu Help>system Info... use its copy button and paste that info in an append here.
Supply pertinent information for quicker answers
The more information you supply about your situation, the better equipped other community members will be to answer. Consider including the following in your question:
Adobe product and version number
Operating system and version number
The full text of any error message(s)
What you were doing when the problem occurred
Screenshots of the problem
Computer hardware, such as CPU; GPU; amount of RAM; etc.

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Moderators are here, I can assure you!
But we're here just to moderate - we're not able to help or offer advice. If you want feedback from Nokia, you should use the 'Contact Us' link at the top of the page.
Board Mod Team
READ THIS FIRST!
Click here if you want help from Nokia

After upgrade mac os to 10.8 the safari surve net is very very slow

Hi my imac using the Mac OS 10.7 and after upgrade to 10.8 the whole imac become very slow especially safari and apps features. May i know what happen to my iMac?

First, back up all data immediately, as your boot drive might be failing.
Take these steps when you notice the problem.
Step 1
Launch the Activity Monitor application in any of the following ways:
☞ Enter the first few letters of its name into a Spotlight search. Select it in the results (it should be at the top.)
☞ In the Finder, select Go ▹ Utilities from the menu bar, or press the key combination shift-command-U. The application is in the folder that opens.
☞ Open LaunchPad. Click Utilities, then Activity Monitor in the icon grid.
Select the CPU tab of the Activity Monitor window.
Select All Processes from the menu in the toolbar, if not already selected.
Click the heading of the % CPU column in the process table to sort the entries by CPU usage. You may have to click it twice to get the highest value at the top. What is it, and what is the process? Also post the values for % User, % System, and % Idle at the bottom of the window.
Select the System Memory tab. What values are shown in the bottom part of the window for Page outs and Swap used?
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Step 2
You must be logged in as an administrator to carry out this step.
Launch the Console application in the same way as above. Make sure the title of the Console window is All Messages. If it isn't, select All Messages from the SYSTEM LOG QUERIES menu on the left.
Post the 50 or so most recent messages in the log — the text, please, not a screenshot.
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And
[loaded]    com.starfield.backupservice.plist 3rd-Party support link
You can test by booting into Safe Mode. If it works reasonable well there, it is likely one of the startup or login items.
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JMF Does not release the RTP ports

Hi,
I am working on a program that simply listens RTP messages from the network and play them on a PC. My datasource is created as follows:
String url2= "rtp://127.0.0.1:10000/audio/1";
medialocator=new MediaLocator(url2);
ds=Manager.createDataSource(medialocator);
This ds will later be given to a player for the local playback. It has a flow like this: DS->Processor->DS->Player.
Everything works fine for the first execution. The stream is played. However when I stop the player, processor and even the ds, JMF does not realease the RTP ports. I still can see them in the netstat output. Heres how I try to release the resources:
public void releaseResources(){
try{
if(player!=null){
player.stop();
player.deallocate();
player=null;
if(ds2!=null){
ds2.stop();
ds2.disconnect();
ds2=null;
if(processor!=null){
processor.stop();
processor.removeControllerListener(this);
processor.deallocate();
medialocator=null;
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Because of this problem, when I try to do the same thing on the same url after sometime, it gives the error "Cannot create the RTP Session: Can't open local data port: 10000". The ports are released after I shutdown the JVM.
Does anyone have an idea on how to release this ports in the same JVM session?
Thanks a lot,
Murat

DataSource objects open a connection to their source, in your case, a connection to an RTP port. You need to call
ds.disconnect()
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RTP transmission in linux

I am using linux for JMF RTP transmission.But the RTP data is not being transmitted.Can anyone suggest me the problem behind it.In windows its working fine.Is there any special configuration to be done on linux side for RTP transmission? I am using linux performance pack of JMF in linux machine.I also tried with cross platform pack of JMF.Thanks in advance...

We finally did resolve this issue!!!!
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     MulticastSocket socket = new MulticastSocket(4446);
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Greetings!
Nico and Charly

Setting priority to the RTP packets

Hi,
I am developing a system which will use multicasting to trnamist RTP packets. To ensure the better transmission, I want to set priority to my transmitting RTP packets. My system has been developed based on AVTransmit3 and AVReceive3 provided by sun where RTPManager has been used.
However is it possible to add special information to the RTP packets so that router underlying router will allow my packet to be transmitted first?
A repply is badly needed...................

Well......according to the RFC2474 and RFC2475, diffserv header can map with the precedence field of the IP header, In IP header TOS field can be mapped with DSCP field of the Diffserv Header. Now jcparques, though IP header can be diffserv capable but is their no router availble in reality which is diifserv capable?Or if I use any simulator like NS2 or other simulator which can work with java, can't I implement diffserv in rtp packet?

The RTP transmission of a message is slow?

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