Re: Please reply error in Transmitting the rtp packet

Did you try openning a MJPEG video file?
" static int MJPEG_TYPE = 26; //RTP payload type for MJPEG video"

SilvioReis, please check the date of the last post before joining in a discussion. There's no need to try to answer year old questions...

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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...

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?

Please reply.Error in invoking target 'install' of ..../ins_ctx.mk'

Hi,
I am trying to install Oracle 10.2.0.1(64 bit) on RHEL 4.0(2.6.9-5.ELsmp). When installation was on linking stage i got three error messages which are given below.
(1)Error in invoking target 'install' of makefile '/oracle/app/oracle/product/10.2.0/db_2/ctx/lib/ins_ctx.mk'.
(2)Error in invoking target 'agent nmo nmb' of makefile '/oracle/app/oracle/product/10.2.0/db_2/sysman/lib/ins_sysman.mk'.
(3)Error in invoking target 'all_no_orcl' of makefile '/oracle/app/oracle/product/10.2.0/db_2/rdbms/lib/ins_rdbms.mk'
Due to this errors, installation of Oracle was not successfull. Can you please tell me what is causing this errors?
I have followed all steps given on following link eventhough it is throwing these errors.
http://www.oracle.com/technology/pub/articles/smiley_10gdb_install.html#rhel4
Please help in this matter as soon as possible.
Regards,
RJ.
Message was edited by:
RJiv

Make sure you have followed all steps according to the platform's install guide and you have met all prerequisites (verify all required install packages and in case you have done so, don't skip the prerequisites warning). It could also be an issue with the 64 bit library path.
~ Madrid

Sir please reply me my lap is not spport the recovery disc and error report this 0xc00000eg error

please reply me urgently

We need the laptop model and also a little explanation as to why you need to recover. Odds are it is a bad DVD drive or bad hard drive causing the error.

RNIF adapter error - Transmitting the message to endpoint failed

Hi
I have developed one interface SAPR/3 - PI - RNIF (Vendor system). the communication happens from PI to Vendor via HTTPS protocol, the SSL certificates has been installed on PI J2EE engine, and configured in RNIF receiver communication channel.
But when i perofrm end-end test and tryting to send messages from PI to Vendor system getting below error.
"Transmitting the message to endpoint https://**.*.*.:***/SeeburgerDT/HTTPController?HTTPInterface=rosettanet using connection RNIFAdapter failed, due to: com.sap.engine.interfaces.messaging.api.exception.MessagingException: Error transmitting the message over HTTP. Reason: java.lang.RuntimeException: Error while silently connecting: org.w3c.www.protocol.http.HttpException: Connect timed out."
The URL "https://**.*.*.:***/SeeburgerDT/HTTPController?HTTPInterface=rosettanet" provided by Vendor team. i have checkd with the Vendor team and they said the URL works fine from their side and able to receive XML messages from our PI server (we are working on PI7.1.).
Pls suggest why im getting Transmission error in PI RNIF adapter log.
Is it the problem with SSL certificates installed on PI or URL used in RNIF communication channel ?
Please help how to resolve this issue.. i would appreciate your help.
Regards,
Rajesh

Hi,
we are getting same message.
"Transmitting the message to endpoint https://**.*.*.:**/*********************** using connection RNIFAdapter failed, due to: com.sap.engine.interfaces.messaging.api.exception.MessagingException: Error transmitting the message over HTTP. Reason: java.lang.RuntimeException: Error while silently connecting: org.w3c.www.protocol.http.HttpException: Connect timed out."
SSL_get_state() returned 0x00001180 "SSLv3 read client certificate A"
[Thr 1085831488] *** ERROR during SecudeSSL_SessionStart() from SSL_accept()==SSL_ERROR_SSL
[Thr 1085831488] SecudeSSL_SessionStart: SSL_accept() failed --
[Thr 1085831488]   secude_error 536875074 (0x20001042) = "received a fatal SSLv3 bad certificate alert message from the peer"
[Thr 1085831488] >> -
Begin of Secude-SSL Errorstack -
>>
[Thr 1085831488] WARNING in ssl3_read_bytes: (536875074/0x20001042) received a fatal SSLv3 bad certificate alert message from the pe
[Thr 1085831488] << -
End of Secude-SSL Errorstack -
[Thr 1085831488]   SSL NI-sock: local=10.1.1.11:51101 peer=210.71.237.26:41004
[Thr 1085831488] <<- ERROR: SapSSLSessionStart(sssl_hdl=102b270)==SSSLERR_SSL_ACCEPT
[Thr 1085831488] *** ERROR => IcmConnInitServerSSL: SapSSLSessionStart returned (-56): SSSLERR_SSL_ACCEPT [icxxconn_mt. 1676]
[Thr 1088473408] Thu Oct 15 07:31:46 2009
[Thr 1088473408]   SSL_get_state() returned 0x00001180 "SSLv3 read client certificate A"
[Thr 1088473408] *** ERROR during SecudeSSL_SessionStart() from SSL_accept()==SSL_ERROR_SSL
[Thr 1088473408] SecudeSSL_SessionStart: SSL_accept() failed --
[Thr 1088473408]   secude_error 536875074 (0x20001042) = "received a fatal SSLv3 bad certificate alert message from the peer"
[Thr 1088473408] >> -
Begin of Secude-SSL Errorstack -
>>
[Thr 1088473408] WARNING in ssl3_read_bytes: (536875074/0x20001042) received a fatal SSLv3 bad certificate alert message from the pe
[Thr 1088473408] << -
End o
Could you please let us know how did you resolve the issue?
Thanks inadvance
Regards
Lshriva

After I Repair The Permissions On My Mac Pro, Then I Verify The Permissions And They Have To Be Repaired Again And Again. Why Does This Happen All The Time. Please Reply, James

Hi,
After I Repair The Permissions On My Mac Pro, Then I Verify The Permissions And They Have To Be Repaired Again And Again.
Why Does This Happen All The Time.
Please Reply,
James

As long as the report ends up with 'Permissions repair complete' then, as far as permissions go, you are fine. You can ignore the various statements in the report:
Permissions you can ignore on 10.5 onwards:
http://support.apple.com/kb/TS1448
Using 'should be -rw-r--r-- , they are lrw-r--r--' as an example, you will see the that the permissions are not changed, but the | indicates a different location. This is because an update to Leopard onwards changed the location of a number of system components.
Poster rccharles has provided this description of what it all means:
drwxrwxrwx
d = directory
r = read
w = write
x = executeable program
drwxrwxrwx
| | |
| | all other users not in first two types
| |
| group
|
owner

Runtime error in transmitting message

In a scenario in XI using File sender to IDoc receiver Communication channel, I am getting an error in RWB Messg Monitoring (Details>Audit).
The file is picked up by the File adapter. But, moving ahead, we get this error.
"Transmitting the message to endpoint hhtp://blade3:8000/sap/xi/engine?type=entry using connection AFW failed, due to: com.sap.aii.af.ra.ms.api.RecoverableException: java.net.MalformedURLException: unknown protocol: hhtp."
Could any body please tell how to fix it? Why is the error being generated?

HI Niladri
) Check pipeline URL in the SLD in the business system of the Integration Server
For this go to SLD->Business System-><Integrtaion Server>->Pipline URL: It should be like this http://<host>:<port>/sap/xi/engine?type=entry
Where host is the host name of the Integration Server and port is the HTTP(8xxx) port.
2) RWB error - have a look into this SAP Note -768456
3) Check the ICM port also. Go to SMICM->Services here you should have http port
Cheers..
Vasu
<i>** Reward Points if found useful **</i>

Error while transmitting message.

All,
Newbie here. We are trying a simple file to file scenario - where xi picks up the file from a FTP location and maps the message and dumps it in another FTP location.
While we execute this example - we get the following error -
<b> "Transmitting the message to endpoint http://sandbox1.xxxx.com:54600/sap/xi/engine?type=entry using connection AFW failed, due to: com.sap.aii.af.ra.ms.api.RecoverableException: Received HTTP response code 404 : Not Found."</b>
This is what it shows on the message audit log,
2007-01-25 14:51:11     Success     Send binary file "profileinput.xml" from FTP server "166.151.310.200:/sap/work", size 476 bytes with QoS EO
2007-01-25 14:51:11     Success     Application attempting to send an XI message asynchronously using connection AFW.
2007-01-25 14:51:11     Success     Trying to put the message into the send queue.
2007-01-25 14:51:11     Success     Message successfully put into the queue.
2007-01-25 14:51:11     Success     The application sent the message asynchronously using connection AFW. Returning to application.
2007-01-25 14:51:11     Success     The message was successfully retrieved from the send queue.
2007-01-25 14:51:11     Success     Confirmation mode test found. File will be resent next time
2007-01-25 14:51:11     Success     The message status set to DLNG.
2007-01-25 14:51:11     Error     Transmitting the message to endpoint http://sandbox1.xxxx.com:54600/sap/xi/engine?type=entry using connection AFW failed, due to: com.sap.aii.af.ra.ms.api.RecoverableException: Received HTTP response code 404 : Not Found.
2007-01-25 14:51:11     Success     The asynchronous message was successfully scheduled to be delivered at Thu Jan 25 14:56:11 CST 2007.
What might be the issue? Any ideas? (im not able to see this message xml in the SXMB_MONI also.
Thanks.

Thanks for the replies. The problem is solved.
Can you tell me why we had to change the port from 54600 to 8046? The original IP was there on the SLD already - probably was configured by the installation team.
If you can give me some URLs or articles that explain this difference - that would be great!
Thanks.

How to synchronize audio track and video when transmitting by RTP

since the audio stream and video stream are transmitted separately on the network even if in a merged data, how to synchronize the audio stream and video stream at the receiving end ?

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.

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;
ds=null;
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()
on both datasources, but SPECIFICLLY on ds (as opposed to ds2) if you want it to release the port.

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...

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...

Packet sniffer only picks up UDP and no RTP packets when using JMF???

Hi,
I am developing a voice mail application to interface with asterisk. Here is the problem.
I am using ethereal packet sniffer to sniff the packets. When I connect two regular SIP phones and sniff , I can sniff the RTP packets.
But when I use JMF AVtransmit2.java and AVReceive2.java I sniff only UDP packets and no RTP packets.
I am very confused. What is going on? If JMF sends over RTP (that uses UDP underneath), then why cannot packet sniffers detect it.

Hi,
I am developing a voice mail application to interface with asterisk. Here is the problem.
I am using ethereal packet sniffer to sniff the packets. When I connect two regular SIP phones and sniff , I can sniff the RTP packets.
But when I use JMF AVtransmit2.java and AVReceive2.java I sniff only UDP packets and no RTP packets.
I am very confused. What is going on? If JMF sends over RTP (that uses UDP underneath), then why cannot packet sniffers detect it.

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
I'm not to sure about the RTP Header I'm working on that as well.. so if I find anything I will let you know.

Re: Please reply error in Transmitting the rtp packet

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