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  • How to distinguish between two different UDP clients on the same IP address?

    - by Ricket
    I'm writing a UDP server, which is a first for me; I've only done a bit of TCP communications. And I'm having trouble figuring out exactly how to distinguish which user is which, since UDP deals only with packets rather than connections and I therefore cannot tell exactly who I'm communicating with. Here is pseudocode of my current server loop: DatagramPacket p; socket.receive(p); // now p contains the user's IP and port, and the data int key = getKey(p); if(key == 0) { // connection request key = makeKey(p); clients.add(key, p.ip); send(p.ip, p.port, key); // give the user his key } else { // user has a key // verify key belongs to that IP address // lookup the user's session data based on the key // react to the packet in the context of the session } When designing this, I kept in mind these points: Multiple users may exist on the same IP address, due to the presence of routers, therefore users must have a separate identification key. Packets can be spoofed, so the key should be checked against its original IP address and ignored if a different IP tries to use the key. The outbound port on the client side might change among packets. Is that third assumption correct, or can I simply assume that one user = one IP+port combination? Is this commonly done, or should I continue to create a special key like I am currently doing? I'm not completely clear on how TCP negotiates a connection so if you think I should model it off of TCP then please link me to a good tutorial or something on TCP's SYN/SYNACK/ACK mess. Also note, I do have a provision to resend a key, if an IP sends a 0 and that IP already has a pending key; I omitted it to keep the snippet simple. I understand that UDP is not guaranteed to arrive, and I plan to add reliability to the main packet handling code later as well.

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  • Django + Virtualenv: manage.py commands fail with ImportError of project name.

    - by Bartek
    This is blowing my mind because it's probably an easy solution, but I can't figure out what could be causing this. So I have a new dev box and am setting everything up. I installed virtualenv, created a new environment for my project under ~/.virtualenvs/projectname Then, I cloned my project from github into my projects directory. Nothing fancy here. There are no .pyc files sitting around so it's a clean slate of code. Then, I activated my virtualenv and installed Django via pip. All looks good so far. Then, I run python manage.py syncdb within my project dir. This is where I get confused: ImportError: No module named projectname So I figured I may have had some references of projectname within my code. So I grep (ack, actually) through my code base and I find nothing of the sorts. So now I'm at a loss, given this environment why am I getting an ImportError on a module named projectname that isn't referenced anywhere in my code? I look forward to a solution .. thanks guys!

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  • activemessaging with stomp and activemq.prefetchSize=1

    - by Clint Miller
    I have a situation where I have a single activemq broker with 2 queues, Q1 and Q2. I have two ruby-based consumers using activemessaging. Let's call them C1 and C2. Both consumers subscribe to each queue. I'm setting activemq.prefetchSize=1 when subscribing to each queue. I'm also setting ack=client. Consider the following sequence of events: 1) A message that triggers a long-running job is published to queue Q1. Call this M1. 2) M1 is dispatched to consumer C1, kicking off a long operation. 3) Two messages that trigger short jobs are published to queue Q2. Call these M2 and M3. 4) M2 is dispatched to C2 which quickly runs the short job. 5) M3 is dispatched to C1, even though C1 is still running M1. It's able to dispatch to C1 because prefetchSize=1 is set on the queue subscription, not on the connection. So the fact that a Q1 message has already been dispatched doesn't stop one Q2 message from being dispatched. Since activemessaging consumers are single-threaded, the net result is that M3 sits and waits on C1 for a long time until C1 finishes processing M1. So, M3 is not processed for a long time, despite the fact that consumer C2 is sitting idle (since it quickly finishes with message M2). Essentially, whenever a long Q1 job is run and then a whole bunch of short Q2 jobs are created, exactly one of the short Q2 jobs gets stuck on a consumer waiting for the long Q1 job to finish. Is there a way to set prefetchSize at the connection level rather than at the subscription level? I really don't want any messages dispatched to C1 while it is processing M1. The other alternative is that I could create a consumer dedicated to processing Q1 and then have other consumers dedicated to processing Q2. But, I'd rather not do that since Q1 messages are infrequent--Q1's dedicated consumers would sit idle most of the day tying up memory.

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  • Compiling scipy on Windows 32-bit

    - by Sridhar Ratnakumar
    Has anyone tried compiling SciPy on Windows using numpy-1.3.0 that was built with the pre-built ATLAS libraries (atlas3.6.0_WinNT_P4SSE2.zip) linked in the installation document. I get the following linker error, and have no ideas as to how to fix this issue. $ python setup.py config --compiler=mingw32 build --compiler=mingw32 install --root=i [...] creating build\temp.win32-2.6\Release creating build\temp.win32-2.6\Release\scipy creating build\temp.win32-2.6\Release\scipy\integrate compile options: '-DNO_ATLAS_INFO=2 -I"C:\Documents and Settings\apy\Application Data\Python\Python26\site-packages\numpy\core\inc lude" -IC:\Python26\include -IC:\Python26\PC -c' gcc -mno-cygwin -O2 -Wall -Wstrict-prototypes -DNO_ATLAS_INFO=2 -I"C:\Documents and Settings\apy\Application Data\Python\Python26\ site-packages\numpy\core\include" -IC:\Python26\include -IC:\Python26\PC -c scipy\integrate\_odepackmo dule.c -o build\temp.win32-2.6\Release\scipy\integrate\_odepackmodule.o C:\MinGW\bin\g77.exe -g -Wall -mno-cygwin -g -Wall -mno-cygwin -shared build\temp.win32-2.6\Release\scipy\integrate\_odepackmodule .o -LC:\atlas3.6.0_WinNT_P4SSE2 -LC:\MinGW\lib -LC:\MinGW\lib\gcc\mingw32\3.4.5 -LC:\Python26\libs -LC:\Act ivePython32Python26\PCbuild -Lbuild\temp.win32-2.6 -lodepack -llinpack_lite -lmach -latlas -lcblas -lf77blas -llapack -lpython26 - lg2c -o build\lib.win32-2.6\scipy\integrate\_odepack.pyd C:\atlas3.6.0_WinNT_P4SSE2/libf77blas.a(ATL_F77wrap_daxpy.o):ATL_F77wrap_axpy.c:(.text+0x3c): undefined reference to `ATL _daxpy' C:\atlas3.6.0_WinNT_P4SSE2/libf77blas.a(ATL_F77wrap_dscal.o):ATL_F77wrap_scal.c:(.text+0x26): undefined reference to `ATL _dscal' C:\atlas3.6.0_WinNT_P4SSE2/libf77blas.a(ATL_F77wrap_dcopy.o):ATL_F77wrap_copy.c:(.text+0x3d): undefined reference to `ATL _dcopy' C:\atlas3.6.0_WinNT_P4SSE2/libf77blas.a(ATL_F77wrap_idamax.o):ATL_F77wrap_amax.c:(.text+0x1e): undefined reference to `AT L_idamax' C:\atlas3.6.0_WinNT_P4SSE2/libf77blas.a(ATL_F77wrap_ddot.o):ATL_F77wrap_dot.c:(.text+0x36): undefined reference to `ATL_d dot' collect2: ld returned 1 exit status error: Command "C:\MinGW\bin\g77.exe -g -Wall -mno-cygwin -g -Wall -mno-cygwin -shared build\temp.win32-2.6\Release\scipy\integrat e\_odepackmodule.o -LC:\atlas3.6.0_WinNT_P4SSE2 -LC:\MinGW\lib -LC:\MinGW\lib\gcc\mingw32\3.4.5 -LC:\Python 26\libs -LC:\Python26\PCbuild -Lbuild\temp.win32-2.6 -lodepack -llinpack_lite -lmach -latlas -lcblas -lf77blas -llap ack -lpython26 -lg2c -o build\lib.win32-2.6\scipy\integrate\_odepack.pyd" failed with exit status 1 Does anyone know what could have gone wrong here?

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  • linux raw socket programming question

    - by user194420
    Hi all, I am trying to create a raw socket which send and receive message with ip/tcp header under linux. I can successfully binds to a port and receive tcp message(ie:syn) However, the message seems to be handled by the os, but not mine. I am just a reader of it(like wireshark). My raw socket binds to port 8888, and then i try to telnet to that port . In wireshark, it shows that the port 8888 reply a "rst ack" when it receive the "syn" request. In my program, it shows that it receive a new message and it doesnot reply with any message. Any way to actually binds to that port?(prevent os handle it) Here is part of my code, i try to cut those error checking for easy reading sockfd = socket(AF_INET, SOCK_RAW, IPPROTO_TCP); int tmp = 1; const int *val = &tmp; setsockopt (sockfd, IPPROTO_IP, IP_HDRINCL, val, sizeof (tmp)); servaddr.sin_family = AF_INET; servaddr.sin_addr.s_addr = htonl(INADDR_ANY); servaddr.sin_port = htons(8888); bind(sockfd, (struct sockaddr*)&servaddr, sizeof(servaddr)); //call recv in loop

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  • Add an item to the Finder/Save dialog sidebar

    - by Clinton Blackmore
    I'm working on a script where a user logs into a guest account on OS and is prompted for their network credentials in order to mount their network home folder (while they benefit from working on a local user folder). As the guest folder is deleted when users log out, I want to discourage them from saving anything there. I would like to replace the items on the Finder and Open/Save sidebar lists (such as "Desktop", username, "Documents", etc) with ones that would save into their network home folder. It is possible to do this using AppleScript or Cocoa APIs, or do I need to modify a plist and restart the Finder? [Ack. Looking into ~/Library/Preferences/com.apple.sidebars.plist, it isn't at all clear how I'd populate it.] Similar Questions: AppleScript: adding mounted folder to Finder Sidebar? suggests using fstab; this code will most likely run as a user and really, automounting at that point would be too late. How do you programmatically put folder icons on the Finder sidebar, given that you have to use a custom icon for the folder? Says there is no Cocoa API, but that you can use a carbon-style LSSharedFileList API that is only documented in a single header file. Does anyone know of some example code to add an item to the Finder sidebar?

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  • WYSIWYG with Qt - font size woes

    - by Rob
    I am creating a custom Qt widget that mimics an A4 printed page and am having problems getting fonts to render at the correct size. My widget uses QPainter::setViewport and QPainter::setWindow to mimic the A4 page, using units of 10ths of a millimetre which enables me to draw easily. However, attempting to create a font at a specific point size doesn't seem to work and using QFont:setPixelSize isn't accurate. Here is some code: View::View(QWidget *parent) : QWidget(parent), printer(new QPrinter) { printer->setPaperSize(QPrinter::A4); printer->setFullPage(true); } void View::paintEvent(QPaintEvent*) { QPainter painter(this); painter.setWindow(0, 0, 2100, 2970); painter.setViewport(0, 0, printer->width(), printer->height()); // Draw a rect at x = 1cm, y = 1cm, 6cm wide and 1 inch high painter.drawRect(100, 100, 600, 254); // Create a 72pt (1 inch) high font QFont font("Arial"); font.setPixelSize(254); painter.setFont(font); // Draw in the same box // The font is too large painter.drawText(QRect(100, 100, 600, 254), tr("Wg\u0102")); // Ack - the actual font size reported by the metrics is 283 pixels! const QFontMetrics fontMetrics = painter.fontMetrics(); qDebug() << "Font height = " << fontMetrics.height(); } So I'm asking for a 254 high font (1 inch, 72 pts) and it's too big and sure enough when I query for the font height via QFontMetrics it is 283 high. Does anyone else know how to use font sizes in points when using custom mapping modes like this? It must be possible. Note that I cannot see how to convert between logical/device points either (i.e. the Win32 DPtoLP/LPtoDP equivalents.)

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  • Security header is not valid - using curl php

    - by toni
    Hi all, Im implementing the Express Checkout, Paypal API using PHP. I have no problem with the first step:SetExpressCheckout. I a have awk=success. But in method GetExpressCheckout I get "Security header is not valid". I try to figure out the problem and i think found out maybe it was the curl not working well.. What i did i copy the whole URL: https://api-3t.sandbox.paypal.com/nvp?USER=sanbox_1276609583_biz_api1.gmail.com&PWD=1276609589&SIGNATURE=AYVosblmD7khKkvvb.bNxvFT0OQ2A8GopwByEuC.CfMHt65VaUmvAEy-&VERSION=62.0&token=EC-3YG18670X88588437&METHOD=GetExpressCheckoutDetails and paste it to the browser. This will result to: TOKEN=EC%2d3YG18670X88588437&CHECKOUTSTATUS=PaymentActionNotInitiated&TIMESTAMP=2010%2d06%2d16T07%3a40%3a12Z&CORRELATIONID=e1a1e469bf066&ACK=Success&VERSION=62%2e0&BUILD=1356926... But when that url executed in the function I made it will not work. Below is my function: function mycurl($url,$querystr){ $ch = curl_init(); curl_setopt($ch, CURLOPT_URL, $url); curl_setopt($ch, CURLOPT_VERBOSE, 1); curl_setopt($ch, CURLOPT_SSL_VERIFYHOST, false); curl_setopt($ch, CURLOPT_SSL_VERIFYPEER, false); curl_setopt($ch, CURLOPT_POST, 1); curl_setopt($ch, CURLOPT_POSTFIELDS, $querystr); curl_setopt($ch, CURLOPT_RETURNTRANSFER,1); $response = curl_exec($ch); curl_close ($ch); return $response; } I hope somebody can help on this. thanks so much. Note: - I Used the sandbox for this. I created a sandbox account, I have a Business account to represent a merchant, and a Personal account to represent a buyer. And I used this: endpoint url: api-3t.sandbox.paypal.com/nvp sandbox url: www.sandbox.paypal.com/cgi-bin/webscr This should not be the issue.

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  • one two-directed tcp socket of two one-directed? (linux, high volume, low latency)

    - by osgx
    Hello I need to send (interchange) a high volume of data periodically with the lowest possible latency between 2 machines. The network is rather fast (e.g. 1Gbit or even 2G+). Os is linux. Is it be faster with using 1 tcp socket (for send and recv) or with using 2 uni-directed tcp sockets? The test for this task is very like NetPIPE network benchmark - measure latency and bandwidth for sizes from 2^1 up to 2^13 bytes, each size sent and received 3 times at least (in teal task the number of sends is greater. both processes will be sending and receiving, like ping-pong maybe). The benefit of 2 uni-directed connections come from linux: http://lxr.linux.no/linux+v2.6.18/net/ipv4/tcp_input.c#L3847 3847/* 3848 * TCP receive function for the ESTABLISHED state. 3849 * 3850 * It is split into a fast path and a slow path. The fast path is 3851 * disabled when: ... 3859 * - Data is sent in both directions. Fast path only supports pure senders 3860 * or pure receivers (this means either the sequence number or the ack 3861 * value must stay constant) ... 3863 * 3864 * When these conditions are not satisfied it drops into a standard 3865 * receive procedure patterned after RFC793 to handle all cases. 3866 * The first three cases are guaranteed by proper pred_flags setting, 3867 * the rest is checked inline. Fast processing is turned on in 3868 * tcp_data_queue when everything is OK. All other conditions for disabling fast path is false. And only not-unidirected socket stops kernel from fastpath in receive

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  • one two-directed tcp socket OR two one-directed? (linux, high volume, low latency)

    - by osgx
    Hello I need to send (interchange) a high volume of data periodically with the lowest possible latency between 2 machines. The network is rather fast (e.g. 1Gbit or even 2G+). Os is linux. Is it be faster with using 1 tcp socket (for send and recv) or with using 2 uni-directed tcp sockets? The test for this task is very like NetPIPE network benchmark - measure latency and bandwidth for sizes from 2^1 up to 2^13 bytes, each size sent and received 3 times at least (in teal task the number of sends is greater. both processes will be sending and receiving, like ping-pong maybe). The benefit of 2 uni-directed connections come from linux: http://lxr.linux.no/linux+v2.6.18/net/ipv4/tcp_input.c#L3847 3847/* 3848 * TCP receive function for the ESTABLISHED state. 3849 * 3850 * It is split into a fast path and a slow path. The fast path is 3851 * disabled when: ... 3859 * - Data is sent in both directions. Fast path only supports pure senders 3860 * or pure receivers (this means either the sequence number or the ack 3861 * value must stay constant) ... 3863 * 3864 * When these conditions are not satisfied it drops into a standard 3865 * receive procedure patterned after RFC793 to handle all cases. 3866 * The first three cases are guaranteed by proper pred_flags setting, 3867 * the rest is checked inline. Fast processing is turned on in 3868 * tcp_data_queue when everything is OK. All other conditions for disabling fast path is false. And only not-unidirected socket stops kernel from fastpath in receive

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  • iPhone: Which are the most useful techniques for faster Bluetooth?

    - by Mike Howard
    Hi. I'm adding peer-to-peer bluetooth using GameKit to an iPhone shoot-em-up, so speed is vital. I'm sending about 40 messages a second each way, most of them with the faster GKSendDataUnreliable, all serializing with NSCoding. In testing between a 3G and 3GS, this is slowing the 3G down a lot more than I'd like. I'm wondering where I should concentrate my efforts to speed it up. How much slower is GKSendDataReliable? For the few packets that have to get there, would it be faster to send a GKSendDataUnreliable and have the peer send an acknowledgement so I can send again if I don't get the Ack within, say, 100ms? How much faster would it be to create the NSData instance using a regular C array rather than archiving with the NSCoding protocol? Is this serialization process (for about a dozen floats) just as slow as you'd expect from an object creation/deallocation overhead, or is something particularly slow happening? I heard that (for example) sending four seperate sets of data is much, much slower, than sending one piece of data four times the size. Would I make a significant saving by sending separate packets of data that wouldn't always go together in the same packet when they happen at the same time? Are there any other bluetooth performance secrets I've missed? Thanks for your help.

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  • Which are the most useful techniques for faster Bluetooth?

    - by Mike Howard
    Hi. I'm adding peer-to-peer bluetooth using GameKit to an iPhone shoot-em-up, so speed is vital. I'm sending about 40 messages a second each way, most of them with the faster GKSendDataUnreliable, all serializing with NSCoding. In testing between a 3G and 3GS, this is slowing the 3G down a lot more than I'd like. I'm wondering where I should concentrate my efforts to speed it up. How much slower is GKSendDataReliable? For the few packets that have to get there, would it be faster to send a GKSendDataUnreliable and have the peer send an acknowledgement so I can send again if I don't get the Ack within, say, 100ms? How much faster would it be to create the NSData instance using a regular C array rather than archiving with the NSCoding protocol? Is this serialization process (for about a dozen floats) just as slow as you'd expect from an object creation/deallocation overhead, or is something particularly slow happening? I heard that (for example) sending four seperate sets of data is much, much slower, than sending one piece of data four times the size. Would I make a significant saving by sending separate packets of data that wouldn't always go together in the same packet when they happen at the same time? Are there any other bluetooth performance secrets I've missed? Thanks for your help.

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  • TCP/IP RST being sent differently in different browsers.

    - by Brian
    On Mac OS X (10.6), if I start a YouTube video download and pull the Ethernet cable for 5 or so seconds, then plug it back in, I get varying results depending on the browser. With Opera and Chrome, after I plug the cable back in the video continues to load. But with Safari and Firefox, it never does. Using Wireshark to look at the traffic, I found that Opera and Chrome simply ACK the first packet from YouTube after the cable has been plugged back in, but Safari and Firefox set the RST flag (0x4) in the TCP header and no more traffic follows. I can put a HUB in between the machine and the internet connection, the problem goes away and all four browsers continue loading the video when the cable is plugged back into the HUB. Again, looking at the Wireshark logs, it's evident that the machine doesn't see the Mulitcast connection close and there is simply a delay in the packets flowing through. So it seems that if Safari and Firefox sees a Multicast connection close, and then later see data on that same connection, they will send a RST. My question is why? What is the correct course of action, and why are 2/4 browsers doing it one way, while the other 2/4 are doing it another way? Is there somewhere in the code that I can see where this is happening in Firefox, for instance? Thank you very much.

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  • Salesforce/PHP - outbound messages (SOAP) - memory limit issue? DOMDocument::loadXML() issue?

    - by Phill Pafford
    I'm using Salesforce to send outbound messages (via SOAP) to another server. The server can process about 8 messages at a time, but will not send back the ACK file if the SOAP request contains more than 8 messages. SF can send up to 100 outbound messages in 1 SOAP request and I think this is causing a memory issue with PHP. If I process the outbound messages 1 by 1 they all go through fine, I can even do 8 at a time with no issues. But larger sets are not working. ERROR in SF: org.xml.sax.SAXParseException: Premature end of file Looking in the HTTP error logs I see that the incoming SOAP message looks to be getting cut of which throws a PHP warning stating: DOMDocument::loadXML() ... Premature end of data in tag ... PHP Fatal error: Call to a member function getAttribute() on a non-object This leads me to believe that PHP is having a memory issue and can not parse the incoming message due to it's size. I was thinking I could just set: ini_set('memory_limit', '64M'); // This has done nothing to fix the problem But would this be the correct approach? Is there a way I could set this to increase with the incoming SOAP request dynamically? UPDATE: Adding some code $data = fopen('php://input','rb'); $headers = getallheaders(); $content_length = $headers['Content-Length']; $buffer_length = 1000; $fread_length = $content_length + $buffer_length; $content = fread($data,$fread_length); /** * Parse values from soap string into DOM XML */ $dom = new DOMDocument(); $dom->loadXML($content); ....

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  • Unit Testing Interfaces in Python

    - by Nicholas Mancuso
    I am currently learning python in preperation for a class over the summer and have gotten started by implementing different types of heaps and priority based data structures. I began to write a unit test suite for the project but ran into difficulties into creating a generic unit test that only tests the interface and is oblivious of the actual implementation. I am wondering if it is possible to do something like this.. suite = HeapTestSuite(BinaryHeap()) suite.run() suite = HeapTestSuite(BinomialHeap()) suite.run() What I am currently doing just feels... wrong (multiple inheritance? ACK!).. class TestHeap: def reset_heap(self): self.heap = None def test_insert(self): self.reset_heap() #test that insert doesnt throw an exception... for x in self.inseq: self.heap.insert(x) def test_delete(self): #assert we get the first value we put in self.reset_heap() self.heap.insert(5) self.assertEquals(5, self.heap.delete_min()) #harder test. put in sequence in and check that it comes out right self.reset_heap() for x in self.inseq: self.heap.insert(x) for x in xrange(len(self.inseq)): val = self.heap.delete_min() self.assertEquals(val, x) class BinaryHeapTest(TestHeap, unittest.TestCase): def setUp(self): self.inseq = range(99, -1, -1) self.heap = BinaryHeap() def reset_heap(self): self.heap = BinaryHeap() class BinomialHeapTest(TestHeap, unittest.TestCase): def setUp(self): self.inseq = range(99, -1, -1) self.heap = BinomialHeap() def reset_heap(self): self.heap = BinomialHeap() if __name__ == '__main__': unittest.main()

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  • Low Throughput on Windows Named Pipe Over WAN

    - by MichaelB76
    I'm having problems with low performance using a Windows named pipe. The throughput drops off rapidly as the network latency increases. There is a roughly linear relationship between messages sent per second and round trip time. It seems that the client must ack each message before the server will send the next one. This leads to very poor performance, I can only send 5 (~100 byte) messages per second over a link with an RTT of 200 ms. The pipe is asynchronous, using multiple overlapped write operations (and multiple overlapped reads at the client end), but this is not improving throughput. Is it possible to send messages in parallel over a named pipe? The pipe is created using PIPE_TYPE_MESSAGE, would PIPE_READMODE_BYTE work better? Is there any other way I can improve performance? This is a deployed solution, so I can't simply replace the pipe with a socket connection (I've read that Windows named pipe aren't recommended for use over a WAN, and I'm wondering if this is why). I'd be grateful for any help with this matter.

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  • TCP/IP Implementation General Questions

    - by user2971023
    I've implemented the concepts shown here; http://wiki.unity3d.com/index.php/Simple_TCP/IP_Client_-_Server outside of unity and it works. (though i had to create the TCPIPServerApp from scratch as i could not find the base project anywhere). I have some general questions on how to use tcp/ip properly however. I've done some research on tcp/ip itself but I'm still a little confused. It seems like using the method above doesn't guarantee that I'll see the message (res). It just checks on every update to see if there is a different message in res. What if multiple messages are sent and the program lags or something, will i miss the earlier packet(s)? Should i instead do an array so it stores the last X messages? How do i know the data was received? Do I need to add a message id and build in my own ack into the data? Should i check to see if the port is in use before setting up a connection? Sorry for all the questions. This is all new to me but I enjoy this very much! ... Below already answered By Anton, Thanks It sounds like tcp uses its own packet numbering to ensure the packets end up in the right order on the other side. What if a packet is missed, are the subsequent packets thrown away? Or is this numbering and packet ordering, only for handling data that is broken out into multiple packets? TCP will automatically break the data into multiple packets if necessary right?

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  • Weird send() problem (with Wireshark log)

    - by Meta
    I had another question about this issue, but I didn't ask properly, so here I go again! I'm sending a file by sending it in chunks. Right now, I'm playing around with different numbers for the size of that chunk, to see what size is the most efficient. When testing on the localhost, any chunk size seems to work fine. But when I tested it over the network, it seems like the maximum chunk size is 8191 bytes. If I try anything higher, the transfer becomes extremely, painfully, slow. To show what happens, here are the first 100 lines of Wireshark logs when I use a chunk size of 8191 bytes, and when I use a chunk size of 8192 bytes: (the sender is 192.168.0.102, and the receiver is 192.168.0.100) 8191: http://pastebin.com/E7jFFY4p 8192: http://pastebin.com/9P2rYa1p Notice how in the 8192 log, on line 33, the receiver takes a long time to ACK the data. This happens again on line 103 and line 132. I believe this delay is the root of the problem. Note that I have not modified the SO_SNDBUF option nor the TCP_NODELAY option. So my question is, why am I getting delayed ACKs when sending files in chunks of 8192 bytes, when everything works fine when using chunks of 8191 bytes? Edit: As an experiment, I tried to do the file transfer in the other direction (from 192.168.0.100 to 192.168.0.102), and surprisingly, any number worked! (Although numbers around 8000 seemed to perform the smoothest). So then the problem is with my computer! But I'm really not sure what to check for. Edit 2: Here is the pseudocode I use to send and receive data.

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  • Forward all traffic through an ssh tunnel

    - by Eamorr
    I hope someone can follow this and I'll explain as best I can. I'm trying to forward all traffic from port 6999 on x.x.x.224, through an ssh tunnel, and onto port 7000 on x.x.x.218. Here is some ASCII art: |browser|-----|Squid on x.x.x.224|------|ssh tunnel|------<satellite link>-----|Squid on x.x.x.218|-----|www| 3128 6999 7000 80 When I remove the ssh tunnel, everything works fine. The idea is to turn off encryption on the ssh tunnel (to save bandwidth) and turn on maximum compression (to save more bandwidth). This is because it's a satellite link. Here's the ssh tunnel I've been using: ssh -C -f -C -o CompressionLevel=9 -o Cipher=none [email protected] -L 7000:172.16.1.224:6999 -N The trouble is, I don't know how to get data from Squid on x.x.x.224 into the ssh tunnel? Am I going about this the wrong way? Should I create an ssh tunnel on x.x.x.218? I use iptables to stop squid on x.x.x.224 from reading port 80, but to feed from port 6999 instead (i.e. via the ssh tunnel). Do I need another iptables rule? Any comments greatly appreciated. Many thanks in advance, Regarding Eduardo Ivanec's question, here is a netstat -i any port 7000 -nn dump from x.x.x.218: 14:42:15.386462 IP 172.16.1.224.40006 > 172.16.1.218.7000: Flags [S], seq 2804513708, win 14600, options [mss 1460,sackOK,TS val 86702647 ecr 0,nop,wscale 4], length 0 14:42:15.386690 IP 172.16.1.218.7000 > 172.16.1.224.40006: Flags [R.], seq 0, ack 2804513709, win 0, length 0 Update 2: When I run the second command, I get the following error in my browser: ERROR The requested URL could not be retrieved The following error was encountered while trying to retrieve the URL: http://109.123.109.205/index.php Zero Sized Reply Squid did not receive any data for this request. Your cache administrator is webmaster. Generated Fri, 01 Jul 2011 16:06:06 GMT by remote-site (squid/2.7.STABLE9) remote-site is 172.16.1.224 When I do a tcpdump -i any port 7000 -nn I get the following: root@remote-site:~# tcpdump -i any port 7000 -nn tcpdump: verbose output suppressed, use -v or -vv for full protocol decode listening on any, link-type LINUX_SLL (Linux cooked), capture size 65535 bytes channel 2: open failed: connect failed: Connection refused channel 2: open failed: connect failed: Connection refused channel 2: open failed: connect failed: Connection refused channel 2: open failed: connect failed: Connection refused channel 2: open failed: connect failed: Connection refused channel 2: open failed: connect failed: Connection refused channel 2: open failed: connect failed: Connection refused channel 2: open failed: connect failed: Connection refused channel 2: open failed: connect failed: Connection refused channel 2: open failed: connect failed: Connection refused channel 2: open failed: connect failed: Connection refused

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  • CentOS 6 - Make system aware of custom lib paths and missing base links

    - by Mike Purcell
    I am trying to compile libmemcached (1.0.7) on CentOS6, and keep getting the following warning: ... checking for event.h... no configure: WARNING: Unable to find libevent ... I manually compiled libevent (2.0.19) and built it using the following configure line: OPTIONS="--prefix=/usr/local/_custom/app/libevent" Everything compiled and installed fine, but I couldn't figure out how to make the system aware that the lib files are in the custom /usr/local/_custom/app/libevent/libdir. I stumbled upon an article and read that I can make the system aware of custom lib paths by adding a custom file to /etc/ld.so.conf.d/ directory: # /etc/ld.so.conf.d/customApp.conf /usr/local/_custom/app/libevent/lib Then I issued the ldconfig command and was able to confirm that libevent was included by issuing this command: ldconfig -p | ack -i libevent Seeing that libevent was now included in the ldconfig output, I figured I would be able to compile libmemcached and satisfy the aforementioned warning. Unfortunately it did not. So I took another look at the ldconfig output and noticed this: libevent_pthreads-2.0.so.5 (libc6,x86-64) => /usr/local/_custom/app/libevent/lib/libevent_pthreads-2.0.so.5 libevent_openssl-2.0.so.5 (libc6,x86-64) => /usr/local/_custom/app/libevent/lib/libevent_openssl-2.0.so.5 libevent_extra-2.0.so.5 (libc6,x86-64) => /usr/local/_custom/app/libevent/lib/libevent_extra-2.0.so.5 libevent_core-2.0.so.5 (libc6,x86-64) => /usr/local/_custom/app/libevent/lib/libevent_core-2.0.so.5 libevent-2.0.so.5 (libc6,x86-64) => /usr/local/_custom/app/libevent/lib/libevent-2.0.so.5 There are no references to the base links, for example, I would expect to see links to these (ls -la /usr/local/_custom/app/libevent/lib): libevent.so -> libevent-2.0.so.5.1.7 libevent_openssl.so -> libevent_openssl-2.0.so.5.1.7 libevent_core.so -> libevent_core-2.0.so.5.1.7 So either I am doing something wrong, or the system still does not know where to look to find libevent.so. -- Update #1 -- I wasn't able to get libmemcached to compile without the warning notice, even after trying to compile using the following configure command: ./configure --prefix=/usr/local/_custom/app/libmemcached CFLAGS="-I/usr/local/_custom/app/libevent/include" LDFLAGS="-L/usr/local/_custom/app/libevent/lib" I thought for sure this would work because I am directly passing the include and lib directories to the configure command. But it did not.

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  • Forwarding RDP via a Linux machine using iptables: Not working

    - by Nimmy Lebby
    I have a Linux machine and a Windows machine behind a router that implements NAT (the diagram might be overkill, but was fun to make): I am forwarding RDP port (3389) on the router to the Linux machine because I want to audit RDP connections. For the Linux machine to forward RDP traffic, I wrote these iptables rules: iptables -t nat -A PREROUTING -p tcp --dport 3389 -j DNAT --to-destination win-box iptables -A FORWARD -p tcp --dport 3389 -j ACCEPT The port is listening on the Windows machine: C:\Users\nimmy>netstat -a Active Connections Proto Local Address Foreign Address State (..snip..) TCP 0.0.0.0:3389 WIN-BOX:0 LISTENING (..snip..) And the port is forwarding on the Linux machine: # tcpdump port 3389 tcpdump: verbose output suppressed, use -v or -vv for full protocol decode listening on eth0, link-type EN10MB (Ethernet), capture size 65535 bytes 01:33:11.451663 IP shieldsup.grc.com.56387 > linux-box.myapt.lan.ms-wbt-server: Flags [S], seq 94663035, win 8192, options [mss 1460], length 0 01:33:11.451846 IP shieldsup.grc.com.56387 > win-box.myapt.lan.ms-wbt-server: Flags [S], seq 94663035, win 8192, options [mss 1460], length 0 However, I am not getting any successful RDP connections from the outside. The port is not even responding: C:\Users\outside-nimmy>telnet example.com 3389 Connecting To example.com...Could not open connection to the host, on port 3389: Connect failed Any ideas? Update Per @Zhiqiang Ma, I looked at nf_conntrack proc file during a connection attempt and this is what I see (192.168.3.1 = linux-box, 192.168.3.5 = win-box): # cat /proc/net/nf_conntrack | grep 3389 ipv4 2 tcp 6 118 SYN_SENT src=4.79.142.206 dst=192.168.3.1 sport=43142 dport=3389 packets=6 bytes=264 [UNREPLIED] src=192.168.3.5 dst=4.79.142.206 sport=3389 dport=43142 packets=0 bytes=0 mark=0 secmark=0 zone=0 use=2 2nd update Got tcpdump on the router and it seems that win-box is sending an RST packet: 21:20:24.767792 IP shieldsup.grc.com.45349 > linux-box.myapt.lan.3389: S 19088743:19088743(0) win 8192 <mss 1460> 21:20:24.768038 IP shieldsup.grc.com.45349 > win-box.myapt.lan.3389: S 19088743:19088743(0) win 8192 <mss 1460> 21:20:24.770674 IP win-box.myapt.lan.3389 > shieldsup.grc.com.45349: R 721745706:721745706(0) ack 755785049 win 0 Why would Windows be doing this?

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  • Ubuntu with KVM guest VM and broken bridges

    - by MadPsy
    I have an Ubuntu box with a KVM guest VM running. They use bridging so the guest VM attaches to the physical network of its host. The guest VM has 2 NICs in 2 different bridges. First NIC of the VM is tap5 and is in bridge br0 br0 8000.46720f5c572e no eth0.500 tap5 Second NIC of the VM is tap2 and is in bridge br100 br100 8000.76ad2fc96661 no eth0.100 eth0.101 eth0.103 eth0.104 eth0.105 tap2 On the host, br0 has an IP and br100 does not 21: br0: <BROADCAST,MULTICAST,UP,LOWER_UP> mtu 1500 qdisc noqueue state UP link/ether 46:72:0f:5c:57:2e brd ff:ff:ff:ff:ff:ff inet 192.168.100.4/24 brd 192.168.10.255 scope global br0 inet6 fe80::d6ae:52ff:febe:777/64 scope link valid_lft forever preferred_lft forever On the guest, its eth0 and eth1 interfaces both have IP addresses 2: eth0: <BROADCAST,MULTICAST,UP,LOWER_UP> mtu 1500 qdisc pfifo_fast state UP qlen 1000 link/ether 00:3e:61:fb:7a:da brd ff:ff:ff:ff:ff:ff inet 192.168.100.6/24 brd 192.168.100.255 scope global eth0 inet6 fe80::23e:61ff:fefb:7ada/64 scope link valid_lft forever preferred_lft forever 3: eth1: <BROADCAST,MULTICAST,UP,LOWER_UP> mtu 1500 qdisc pfifo_fast state UP qlen 1000 link/ether 00:3e:61:fb:7a:ea brd ff:ff:ff:ff:ff:ff inet 172.16.50.129/25 brd 172.16.50.255 scope global eth1 inet6 fe80::23e:61ff:fefb:7aea/64 scope link valid_lft forever preferred_lft forever On the guest VM, a tcpdump of its eth1 interface (tap2) shows traffic from its eth0 interface (tap5), as if the 2 bridges are themselves bridged. This means any interface on br100 is now bridged across to br0 - which is completely broken. root@chillispot:~# tcpdump -c 1 -n -v -i eth1 net 192.168.100.0/24 tcpdump: listening on eth1, link-type EN10MB (Ethernet), capture size 65535 bytes 16:31:24.175583 IP (tos 0x0, ttl 64, id 48054, offset 0, flags [DF], proto TCP (6), length 148) 192.168.100.6.22 > 192.168.100.4.59505: Flags [P.], cksum 0x6c2b (correct), seq 1056321648:1056321744, ack 398642983, win 1700, options [nop,nop,TS val 197473436 ecr 200655363], length 96 What could be bridging the 2 bridges, except the guest VM (which is a stock Ubuntu install)? I am at a complete loss! Thanks.

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  • Wireless traffic stops when downloading large files at high speed: packets lost (Linksys WRT120N router)

    - by Torious
    The problem Note: First I'd like to understand WHY this is happening. Ofcourse, a solution would be nice too. :) When downloading a large file over HTTP at high-speeds, my wireless traffic basically stops: I can't open webpages and the download itself pauses. It pauses pretty much immediately after starting it; sometimes at 800 KB, sometimes at a few MB. After some time, the download (and other traffic) resumes, but the problem keeps reoccurring during the same download. The problem does not occur when using a wired connection through the same router (Linskys WRT120N). Also note that the connection is not dropped when this happens. It's just that the traffic stops and I can't browse to web pages, etc. (SYN packets are sent but nothing is received, etc.) Inspection with Wireshark shows that the following happens: Server sends data packets which are acknowledged by client Server sends a packet, but SEQ indicates some packets were lost (6 packets in one occurrence). Server sends a few more packets and client acknowledges these using "selective acknowledgement" Server stops sending data for a while (since the lost packets were not acknowledged or the router stops forwarding them?) Eventually, server does a "retransmission" and traffic resumes as normal. This all seems normal behavior to me when packet loss occurs. It's the consistent packet loss throughout a large, high-speed download that puzzles me. What might cause this? My own idea is the following: My internet is pretty fast (100 mbps), so when starting a large-file download, the router buffers the incoming data (since wireless introduces some slight delay / lower speed, in part due to other networks), but the buffer overflows and the router drops packets to regulate traffic (and because it has no choice). But how could that happen? Doesn't the TCP window size limit the amount of data that can go unacknowledged? So how can the router's buffer overflow if there can only be like 64 KB waiting to be acknowledged? Note: I've disabled TCP window scaling and dynamic window size through netsh options, in an attempt to fix this, but it doesn't seem to matter. Also, Wireshark shows a pattern of the server sending 2 packets (of 1514 bytes) and the client sending an ACK, so does that rule out a possible buffer overflow? And a few more subsequent packets are received... I'm at a loss here. Thanks for any insights. Things that are (probably) NOT the cause / I have experimented with The browser Various TCP options in Windows 7 (netsh etc.) Router settings such as MTU, beacon interval, UPnP, ...

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  • Concatenation of a 2 second silence audio with a normal audio not working

    - by user1665130
    I have a code for concatenation of files using ffmpeg.Here silence.wav is a mute audio file with 2 seconds length. I need to prepend this mut audio file to REC00096_Jun-06-2014 16.47.28.wav. I tried the folowing code. ffmpeg -i D:\vishnu\silence.wav -i D:\vishnu\REC00096_Jun-06-2014 16.47.28.wav \-filter_complex '[0:0][1:0][2:0][3:0]concat=n=2:v=0:a=1[out]' \-map '[out]' output.wav Following is the error i am getting. D:\vishnu>ffmpeg -i silence.wav -i "D:\vishnu\REC00096_Jun-06-2014 16.47.28.wav" -filter_complex '[0:0][1:0][2:0][3:0]concat=n=2:v=0:a=1[out]' -map '[out]' outp ut.wav ffmpeg version N-59036-g5d8e4f6 Copyright (c) 2000-2013 the FFmpeg developers built on Dec 12 2013 22:01:01 with gcc 4.8.2 (GCC) configuration: --enable-gpl --enable-version3 --disable-w32threads --enable-av isynth --enable-bzlib --enable-fontconfig --enable-frei0r --enable-gnutls --enab le-iconv --enable-libass --enable-libbluray --enable-libcaca --enable-libfreetyp e --enable-libgsm --enable-libilbc --enable-libmodplug --enable-libmp3lame --ena ble-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-l ibopus --enable-librtmp --enable-libschroedinger --enable-libsoxr --enable-libsp eex --enable-libtheora --enable-libtwolame --enable-libvidstab --enable-libvo-aa cenc --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libwavp ack --enable-libx264 --enable-libxavs --enable-libxvid --enable-zlib libavutil 52. 58.100 / 52. 58.100 libavcodec 55. 45.101 / 55. 45.101 libavformat 55. 22.100 / 55. 22.100 libavdevice 55. 5.102 / 55. 5.102 libavfilter 3. 92.100 / 3. 92.100 libswscale 2. 5.101 / 2. 5.101 libswresample 0. 17.104 / 0. 17.104 libpostproc 52. 3.100 / 52. 3.100 Input #0, wav, from 'silence.wav': Metadata: encoder : Lavf55.22.100 Duration: 00:00:02.02, bitrate: 4234 kb/s Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 44100 Hz, 5.1, s16, 4 233 kb/s Guessed Channel Layout for Input Stream #1.0 : mono Input #1, wav, from 'D:\vishnu\REC00096_Jun-06-2014 16.47.28.wav': Duration: 00:00:08.04, bitrate: 384 kb/s Stream #1:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 24000 Hz, mono, s16, 384 kb/s [wav @ 036f5e40] Invalid stream specifier: '[out]'. Last message repeated 1 times Stream map ''[out]'' matches no streams. D:\vishnu>

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  • Conflicting ip routes with local table on attaching a virtual network interface

    - by user1071840
    I have an EC2 instance with these ip rules: $ sudo ip rule show 0: from all lookup local 32766: from all lookup main 32767: from all lookup default I can attach an elastic network interface to it with a private IP. Say the IP of my machine is 10.1.3.12 and the IP of the interface is 10.1.1.190. As soon as I attach the interface to my machine a new entry is added to the routing policy and local routing table: sudo ip rule show 0: from all lookup local 32765: from 10.1.1.190 lookup 10003 32766: from all lookup main 32767: from all lookup default $ sudo ip route show table local broadcast 10.1.1.0 dev eth3 proto kernel scope link src 10.1.1.190 local 10.1.1.190 dev eth3 proto kernel scope host src 10.1.1.190 broadcast 10.1.1.255 dev eth3 proto kernel scope link src 10.1.1.190 broadcast 10.1.3.0 dev eth0 proto kernel scope link src 10.1.3.12 local 10.1.3.12 dev eth0 proto kernel scope host src 10.1.3.12 broadcast 10.1.3.255 dev eth0 proto kernel scope link src 10.1.3.12 broadcast 127.0.0.0 dev lo proto kernel scope link src 127.0.0.1 local 127.0.0.0/8 dev lo proto kernel scope host src 127.0.0.1 local 127.0.0.1 dev lo proto kernel scope host src 127.0.0.1 broadcast 127.255.255.255 dev lo proto kernel scope link src 127.0.0.1 I can send traffic to this ENI directly from a host that can have the same IP as the host the ENI is attached to. This is where the problem starts. I ran tcpdump on the port in question and saw multiple SYNs going to the ENI with src '10.1.3.12' and destination '10.1.1.190' but didn't see even a single ACK. In my understanding if ACKs were being sent from the ENI they'd have destination as 10.1.3.12 i.e. the same as the local machine's IP and such packets will now be routed as local packets matching local routing policy: local 10.1.3.12 dev eth0 proto kernel scope host src 10.1.3.12 I'd like to send all the packets originating from 10.1.1.190 (my ENI) to go back on the same interface i.e. eth3 in this case. Contents of the nee table 10003 are: $ sudo ip route show table 10003 default via 10.1.1.1 dev eth3 I think I can do the following: I don't know if its possible but probably decrease the priority of local table so the packets match the table 10003. Use iptables to mangle these packets and update the local table route to include the mark information But I'm not sure if these are the right approaches.

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