Search Results

Search found 1687 results on 68 pages for 'packet sniffing'.

Page 32/68 | < Previous Page | 28 29 30 31 32 33 34 35 36 37 38 39  | Next Page >

  • How to Verify Signature, Loading PUBLIC KEY From PEM file?

    - by bbirtle
    I'm posting this in the hope it saves somebody else the hours I lost on this really stupid problem involving converting formats of public keys. If anybody sees a simpler solution or a problem, please let me know! The eCommerce system I'm using sends me some data along with a signature. They also give me their public key in .pem format. The .pem file looks like this: -----BEGIN PUBLIC KEY----- MIGfMA0GCSqGSIb3DQEBAQUAA4GNADCBiQKBgQDe+hkicNP7ROHUssGNtHwiT2Ew HFrSk/qwrcq8v5metRtTTFPE/nmzSkRnTs3GMpi57rBdxBBJW5W9cpNyGUh0jNXc VrOSClpD5Ri2hER/GcNrxVRP7RlWOqB1C03q4QYmwjHZ+zlM4OUhCCAtSWflB4wC Ka1g88CjFwRw/PB9kwIDAQAB -----END PUBLIC KEY----- Here's the magic code to turn the above into an "RSACryptoServiceProvider" which is capable of verifying the signature. Uses the BouncyCastle library, since .NET apparently (and appallingly cannot do it without some major headaches involving certificate files): RSACryptoServiceProvider thingee; using (var reader = File.OpenText(@"c:\pemfile.pem")) { var x = new PemReader(reader); var y = (RsaKeyParameters)x.ReadObject(); thingee = (RSACryptoServiceProvider)RSACryptoServiceProvider.Create(); var pa = new RSAParameters(); pa.Modulus = y.Modulus.ToByteArray(); pa.Exponent = y.Exponent.ToByteArray(); thingee.ImportParameters(pa); } And then the code to actually verify the signature: var signature = ... //reads from the packet sent by the eCommerce system var data = ... //reads from the packet sent by the eCommerce system var sha = new SHA1CryptoServiceProvider(); byte[] hash = sha.ComputeHash(Encoding.ASCII.GetBytes(data)); byte[] bSignature = Convert.FromBase64String(signature); ///Verify signature, FINALLY: var hasValidSig = thingee.VerifyHash(hash, CryptoConfig.MapNameToOID("SHA1"), bSignature);

    Read the article

  • MSMQ on Win2008 R2 won’t receive messages from older clients

    - by Graffen
    Hi all I'm battling a really weird problem here. I have a Windows 2008 R2 server with Message Queueing installed. On another machine, running Windows 2003 is a service that is set up to send messages to a public queue on the 2008 server. However, messages never show up on the server. I've written a small console app that just sends a "Hello World" message to a test queue on the 2008 machine. Running this app on XP or 2003 results in absolutely nothing. However, when I try running the app on my Windows 7 machine, a message is delivered just fine. I've been through all sorts of security settings, disabled firewalls on all machines etc. The event log shows nothing of interest, and no exceptions are being thrown on the clients. Running a packet sniffer (WireShark) on the server reveals only a little. When trying to send a message from XP or 2003 I only see an ICMP error "Port Unreachable" on port 3527 (which I gather is an MQPing packet?). After that, silence. Wireshark shows a nice little stream of packets when I try from my Win7 client (as expected - messages get delivered just fine from Win7). I've enabled MSMQ End2End logging on the server, but only entries from the messages sent from my Win7 machine are appearing in the log. So somehow it seems that messages are being dropped silently somewhere along the route from XP or 2003 to my 2008 server. Does anyone have any clues as to what might be causing this mysterious behaviour? -- Jesper

    Read the article

  • Game login authentication and security.

    - by Charles
    First off I will say I am completely new to security in coding. I am currently helping a friend develop a small game (in Python) which will have a login server. I don't have much knowledge regarding security, but I know many games do have issues with this. Everything from 3rd party applications (bots) to WPE packet manipulation. Considering how small this game will be and the limited user base, I doubt we will have serious issues, but would like to try our best to limit problems. I am not sure where to start or what methods I should use, or what's worth it. For example, sending data to the server such as login name and password. I was told his information should be encrypted when sending, so in-case someone was viewing it (with whatever means), that they couldn't get into the account. However, if someone is able to capture the encrypted string, wouldn't this string always work since it's decrypted server side? In other words, someone could just capture the packet, reuse it, and still gain access to the account? The main goal I am really looking for is to make sure the players are logging into the game with the client we provide, and to make sure it's 'secure' (broad, I know). I have looked around at different methods such as Public and Private Key encryption, which I am sure any hex editor could eventually find. There are many other methods that seem way over my head at the moment and leave the impression of overkill. I realize nothing is 100% secure. I am just looking for any input or reading material (links) to accomplish the main goal stated above. Would appreciate any help, thanks.

    Read the article

  • Any suggestions for good automated web load testing tool?

    - by fmunkert
    What are some good automated tools for load testing (stress testing) web applications, that do not use record and replay of HTTP network packets? I am aware that there are numerous load testing tools on the market that record and replay HTTP network packets. But these are unsuitable for my purpose, because of this: The HTTP packet format changes very often in our application (e.g. when we optimize an AJAX call). We do not want to adapt all test scripts just because there is a slight change in HTTP packet format. Our test team shall not need to know any internals about our application to write their test scripts. A tool that replays HTTP packets, however, requires the team to know the format of HTTP requests and responses, such that they can adapt details of the replayed HTTP packets (e.g. user name). The automated load testing tool I am looking for should be able to let the test team write "black box" test scripts such as: Invoke web page at URL http://... . First, enter XXX into text field XXX. Then, press button XXX. Wait until response has been received from web server. Verify that text field XXX now contains the text XXX. The tool should be able to simulate up to several 1000 users, and it should be compatible with web applications using ASP.NET and AJAX.

    Read the article

  • how to continuously send data without blocking?

    - by Donal Rafferty
    I am trying to send rtp audio data from my Android application. I currently can send 1 RTP packet with the code below and I also have another class that extends Thread that listens to and receives RTP packets. My question is how do I continuously send my updated buffer through the packet payload without blocking the receiving thread? public void run() { isRecording = true; android.os.Process.setThreadPriority (android.os.Process.THREAD_PRIORITY_URGENT_AUDIO); int buffersize = AudioRecord.getMinBufferSize(8000, AudioFormat.CHANNEL_CONFIGURATION_MONO, AudioFormat.ENCODING_PCM_16BIT); Log.d("BUFFERSIZE","Buffer size = " + buffersize); arec = new AudioRecord(MediaRecorder.AudioSource.MIC, 8000, AudioFormat.CHANNEL_CONFIGURATION_MONO, AudioFormat.ENCODING_PCM_16BIT, buffersize); short[] readBuffer = new short[80]; byte[] buffer = new byte[160]; arec.startRecording(); while(arec.getRecordingState() == AudioRecord.RECORDSTATE_RECORDING){ int frames = arec.read(readBuffer, 0, 80); @SuppressWarnings("unused") int lenghtInBytes = codec.encode(readBuffer, 0, buffer, frames); RtpPacket rtpPacket = new RtpPacket(); rtpPacket.setV(2); rtpPacket.setX(0); rtpPacket.setM(0); rtpPacket.setPT(0); rtpPacket.setSSRC(123342345); rtpPacket.setPayload(buffer, 160); try { rtpSession2.sendRtpPacket(rtpPacket); } catch (UnknownHostException e) { // TODO Auto-generated catch block e.printStackTrace(); } catch (RtpException e) { // TODO Auto-generated catch block e.printStackTrace(); } catch (IOException e) { // TODO Auto-generated catch block e.printStackTrace(); } } } So when I send on one device and receive on another I get decent audio, but when I send and receive on both I get broken sound like its taking turns to send and receive audio. I have a feeling it could be to do with the while loop? it could be looping around in there and not letting anything else run?

    Read the article

  • Outgoing UDP sniffer in python?

    - by twneale
    I want to figure out whether my computer is somehow causing a UDP flood that is originating from my network. So that's my underlying problem, and what follows is simply my non-network-person attempt to hypothesize a solution using python. I'm extrapolating from recipe 13.1 ("Passing Messages with Socket Datagrams") from the python cookbook (also here). Would it possible/sensible/not insane to try somehow writing an outgoing UDP proxy in python, so that outgoing packets could be logged before being sent on their merry way? If so, how would one go about it? Based on my quick research, perhaps I could start a server process listening on suspect UDP ports and log anything that gets sent, then forward it on, such as: import socket s =socket.socket(socket.AF_INET, socket.SOCK_DGRAM) s.bind(("", MYPORT)) while True: packet = dict(zip('data', 'addr'), s.recvfrom(1,024)) log.info("Recieved {data} from {addr}.".format(**packet)) But what about doing this for a large number of ports simultaneously? Impractical? Are there drawbacks or other reasons not to bother with this? Is there a better way to solve this problem (please be gentle).

    Read the article

  • 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.

    Read the article

  • NetworkStream.Read delay .Net

    - by Gilbes
    I have a class that inherits from TcpClient. In that class I have a method to process responses. In that method I call I get the NetworkStream with MyBase.GetStream and call Read on it. This works fine, excpet the first call to read blocks too long. And by too long I mean that the socket has recieved plenty of data, but won't read it until some arbitrary limit is reached. I can see that it has recieved plenty of data using the packet sniffer WireShark. I have set the recieve buffer to small amounts, and very small amounts (like just a few bytes) to no avail. I have done the same with the buffer byte array I pass to the read method, and it still delays. Or to put it another way. I am download 600k. The download takes 5 seconds (at a little over 100k/second connection to the server which makes sense). The initial Read call takes 2-3 seconds and tells me only 256 bytes are availble (256 is the Recieve buffer and the size of the array I read in to). Then magically, the other few hundred thousand bytes can be read in 256 byte chunks in only a few process ticks each. Using a packet sniffer, I know that during those initial 2-3 seconds, the socket has recieved much more than just 256 bytes. My connection wasn't .25k/second for 3 seconds and then 400k for 2 seconds. How do I get the bytes from a socket as they come in?

    Read the article

  • Winsock tcp/ip Socket listening but connection refused, race condition?

    - by Wayne
    Hello folks. This involves two automated unit tests which each start up a tcp/ip server that creates a non-blocking socket then bind()s and listen()s in a loop on select() for a client that connects and downloads some data. The catch is that they work perfectly when run separately but when run as a test suite, the second test client will fail to connect with WSACONNREFUSED... UNLESS there is a Thread.Sleep() of several seconds between them??!!! Interestingly, there is retry loop every 1 second for connecting after any failure. So the second test loops for a while until timeout after 10 minutes. During that time, netstat -na shows the correct port number is in the LISTEN state for the server socket. So if it is in the listen state? Why won't it accept the connection? In the code, there are log messages that show the select NEVER even gets a socket ready to read (which means ready to accept a connection when it applies to a listening socket). Obviously the problem must be related to some race condition between finishing one test which means close() and shutdown() on each end of the socket, and the start up of the next. This wouldn't be so bad if the retry logic allowed it to connect eventually after a couple of seconds. However it seems to get "gummed up" and won't even retry. However, for some strange reason the listening socket SAYS it's in the LISTEN state even through keeps refusing connections. So that means it's the Windoze O/S which is actually catching the SYN packet and returning a RST packet (which means "Connection Refused"). The only other time I ever saw this error was when the code had a problem that caused hundreds of sockets to get stuck in TIME_WAIT state. But that's not the case here. netstat shows only about a dozen sockets with only 1 or 2 in TIME_WAIT at any given moment. Please help.

    Read the article

  • getaddrinfo appears to return different results between Windows and Ubuntu?

    - by MrDuk
    I have the following two sets of code: Windows #undef UNICODE #include <winsock2.h> #include <ws2tcpip.h> #include <stdio.h> // link with Ws2_32.lib #pragma comment (lib, "Ws2_32.lib") int __cdecl main(int argc, char **argv) { //----------------------------------------- // Declare and initialize variables WSADATA wsaData; int iResult; INT iRetval; DWORD dwRetval; argv[1] = "www.google.com"; argv[2] = "80"; int i = 1; struct addrinfo *result = NULL; struct addrinfo *ptr = NULL; struct addrinfo hints; struct sockaddr_in *sockaddr_ipv4; // struct sockaddr_in6 *sockaddr_ipv6; LPSOCKADDR sockaddr_ip; char ipstringbuffer[46]; DWORD ipbufferlength = 46; /* // Validate the parameters if (argc != 3) { printf("usage: %s <hostname> <servicename>\n", argv[0]); printf("getaddrinfo provides protocol-independent translation\n"); printf(" from an ANSI host name to an IP address\n"); printf("%s example usage\n", argv[0]); printf(" %s www.contoso.com 0\n", argv[0]); return 1; } */ // Initialize Winsock iResult = WSAStartup(MAKEWORD(2, 2), &wsaData); if (iResult != 0) { printf("WSAStartup failed: %d\n", iResult); return 1; } //-------------------------------- // Setup the hints address info structure // which is passed to the getaddrinfo() function ZeroMemory( &hints, sizeof(hints) ); hints.ai_family = AF_UNSPEC; hints.ai_socktype = SOCK_STREAM; // hints.ai_protocol = IPPROTO_TCP; printf("Calling getaddrinfo with following parameters:\n"); printf("\tnodename = %s\n", argv[1]); printf("\tservname (or port) = %s\n\n", argv[2]); //-------------------------------- // Call getaddrinfo(). If the call succeeds, // the result variable will hold a linked list // of addrinfo structures containing response // information dwRetval = getaddrinfo(argv[1], argv[2], &hints, &result); if ( dwRetval != 0 ) { printf("getaddrinfo failed with error: %d\n", dwRetval); WSACleanup(); return 1; } printf("getaddrinfo returned success\n"); // Retrieve each address and print out the hex bytes for(ptr=result; ptr != NULL ;ptr=ptr->ai_next) { printf("getaddrinfo response %d\n", i++); printf("\tFlags: 0x%x\n", ptr->ai_flags); printf("\tFamily: "); switch (ptr->ai_family) { case AF_UNSPEC: printf("Unspecified\n"); break; case AF_INET: printf("AF_INET (IPv4)\n"); sockaddr_ipv4 = (struct sockaddr_in *) ptr->ai_addr; printf("\tIPv4 address %s\n", inet_ntoa(sockaddr_ipv4->sin_addr) ); break; case AF_INET6: printf("AF_INET6 (IPv6)\n"); // the InetNtop function is available on Windows Vista and later // sockaddr_ipv6 = (struct sockaddr_in6 *) ptr->ai_addr; // printf("\tIPv6 address %s\n", // InetNtop(AF_INET6, &sockaddr_ipv6->sin6_addr, ipstringbuffer, 46) ); // We use WSAAddressToString since it is supported on Windows XP and later sockaddr_ip = (LPSOCKADDR) ptr->ai_addr; // The buffer length is changed by each call to WSAAddresstoString // So we need to set it for each iteration through the loop for safety ipbufferlength = 46; iRetval = WSAAddressToString(sockaddr_ip, (DWORD) ptr->ai_addrlen, NULL, ipstringbuffer, &ipbufferlength ); if (iRetval) printf("WSAAddressToString failed with %u\n", WSAGetLastError() ); else printf("\tIPv6 address %s\n", ipstringbuffer); break; case AF_NETBIOS: printf("AF_NETBIOS (NetBIOS)\n"); break; default: printf("Other %ld\n", ptr->ai_family); break; } printf("\tSocket type: "); switch (ptr->ai_socktype) { case 0: printf("Unspecified\n"); break; case SOCK_STREAM: printf("SOCK_STREAM (stream)\n"); break; case SOCK_DGRAM: printf("SOCK_DGRAM (datagram) \n"); break; case SOCK_RAW: printf("SOCK_RAW (raw) \n"); break; case SOCK_RDM: printf("SOCK_RDM (reliable message datagram)\n"); break; case SOCK_SEQPACKET: printf("SOCK_SEQPACKET (pseudo-stream packet)\n"); break; default: printf("Other %ld\n", ptr->ai_socktype); break; } printf("\tProtocol: "); switch (ptr->ai_protocol) { case 0: printf("Unspecified\n"); break; case IPPROTO_TCP: printf("IPPROTO_TCP (TCP)\n"); break; case IPPROTO_UDP: printf("IPPROTO_UDP (UDP) \n"); break; default: printf("Other %ld\n", ptr->ai_protocol); break; } printf("\tLength of this sockaddr: %d\n", ptr->ai_addrlen); printf("\tCanonical name: %s\n", ptr->ai_canonname); } freeaddrinfo(result); WSACleanup(); return 0; } Ubuntu /* ** listener.c -- a datagram sockets "server" demo */ #include <stdio.h> #include <stdlib.h> #include <unistd.h> #include <errno.h> #include <string.h> #include <sys/types.h> #include <sys/socket.h> #include <netinet/in.h> #include <arpa/inet.h> #include <netdb.h> #define MYPORT "4950" // the port users will be connecting to #define MAXBUFLEN 100 // get sockaddr, IPv4 or IPv6: void *get_in_addr(struct sockaddr *sa) { if (sa->sa_family == AF_INET) { return &(((struct sockaddr_in*)sa)->sin_addr); } return &(((struct sockaddr_in6*)sa)->sin6_addr); } int main(void) { int sockfd; struct addrinfo hints, *servinfo, *p; int rv; int numbytes; struct sockaddr_storage their_addr; char buf[MAXBUFLEN]; socklen_t addr_len; char s[INET6_ADDRSTRLEN]; memset(&hints, 0, sizeof hints); hints.ai_family = AF_UNSPEC; // set to AF_INET to force IPv4 hints.ai_socktype = SOCK_DGRAM; hints.ai_flags = AI_PASSIVE; // use my IP if ((rv = getaddrinfo(NULL, MYPORT, &hints, &servinfo)) != 0) { fprintf(stderr, "getaddrinfo: %s\n", gai_strerror(rv)); return 1; } // loop through all the results and bind to the first we can for(p = servinfo; p != NULL; p = p->ai_next) { if ((sockfd = socket(p->ai_family, p->ai_socktype, p->ai_protocol)) == -1) { perror("listener: socket"); continue; } if (bind(sockfd, p->ai_addr, p->ai_addrlen) == -1) { close(sockfd); perror("listener: bind"); continue; } break; } if (p == NULL) { fprintf(stderr, "listener: failed to bind socket\n"); return 2; } freeaddrinfo(servinfo); printf("listener: waiting to recvfrom...\n"); addr_len = sizeof their_addr; if ((numbytes = recvfrom(sockfd, buf, MAXBUFLEN-1 , 0, (struct sockaddr *)&their_addr, &addr_len)) == -1) { perror("recvfrom"); exit(1); } printf("listener: got packet from %s\n", inet_ntop(their_addr.ss_family, get_in_addr((struct sockaddr *)&their_addr), s, sizeof s)); printf("listener: packet is %d bytes long\n", numbytes); buf[numbytes] = '\0'; printf("listener: packet contains \"%s\"\n", buf); close(sockfd); return 0; } When I attempt www.google.com, I don't get the ipv6 socket returned on Windows - why is this? Outputs: (ubuntu) caleb@ub1:~/Documents/dev/cs438/mp0/MP0$ ./a.out www.google.com IP addresses for www.google.com: IPv4: 74.125.228.115 IPv4: 74.125.228.116 IPv4: 74.125.228.112 IPv4: 74.125.228.113 IPv4: 74.125.228.114 IPv6: 2607:f8b0:4004:803::1010 Outputs: (win) Calling getaddrinfo with following parameters: nodename = www.google.com servname (or port) = 80 getaddrinfo returned success getaddrinfo response 1 Flags: 0x0 Family: AF_INET (IPv4) IPv4 address 74.125.228.114 Socket type: SOCK_STREAM (stream) Protocol: Unspecified Length of this sockaddr: 16 Canonical name: (null) getaddrinfo response 2 Flags: 0x0 Family: AF_INET (IPv4) IPv4 address 74.125.228.115 Socket type: SOCK_STREAM (stream) Protocol: Unspecified Length of this sockaddr: 16 Canonical name: (null) getaddrinfo response 3 Flags: 0x0 Family: AF_INET (IPv4) IPv4 address 74.125.228.116 Socket type: SOCK_STREAM (stream) Protocol: Unspecified Length of this sockaddr: 16 Canonical name: (null) getaddrinfo response 4 Flags: 0x0 Family: AF_INET (IPv4) IPv4 address 74.125.228.112 Socket type: SOCK_STREAM (stream) Protocol: Unspecified Length of this sockaddr: 16 Canonical name: (null) getaddrinfo response 5 Flags: 0x0 Family: AF_INET (IPv4) IPv4 address 74.125.228.113 Socket type: SOCK_STREAM (stream) Protocol: Unspecified Length of this sockaddr: 16 Canonical name: (null)

    Read the article

  • Question about memory allocation when initializing char arrays in C/C++.

    - by Carlos Nunez
    Before anything, I apologize if this question has been asked before. I am programming a simple packet sniffer for a class project. For a little while, I ran into the issue where the source and destination of a packet appeared to be the same. For example, the source and destination of an Ethernet frame would be the same MAC address all of the time. I custom-made ether_ntoa(char *) because Windows does not seem to have ethernet.h like Linux does. Code snippet is below: char *ether_ntoa(u_char etheraddr[ETHER_ADDR_LEN]) { int i, j; char eout[32]; for(i = 0, j = 0; i < 5; i++) { eout[j++] = etheraddr[i] >> 4; eout[j++] = etheraddr[i] & 0xF; eout[j++] = ':'; } eout[j++] = etheraddr[i] >> 4; eout[j++] = etheraddr[i] & 0xF; eout[j++] = '\0'; for(i = 0; i < 17; i++) { if(eout[i] < 10) eout[i] += 0x30; else if(eout[i] < 16) eout[i] += 0x57; } return(eout); } I solved the problem by using malloc() to have the compiler assign memory (i.e. instead of char eout[32], I used char * eout; eout = (char *) malloc (32);). However, I thought that the compiler assigned different memory locations when one sized a char-array at compile time. Is this incorrect? Thanks! Carlos Nunez

    Read the article

  • How to change internal buffer size of DataInputStream

    - by Gaks
    I'm using this kind of code for my TCP/IP connection: sock = new Socket(host, port); sock.setKeepAlive(true); din = new DataInputStream(sock.getInputStream()); dout = new DataOutputStream(sock.getOutputStream()); Then, in separate thread I'm checking din.available() bytes to see if there are some incoming packets to read. The problem is, that if a packet bigger than 2048 bytes arrives, the din.available() returns 2048 anyway. Just like there was a 2048 internal buffer. I can't read those 2048 bytes when I know it's not the full packet my application is waiting for. If I don't read it however - it'll all stuck at 2048 bytes and never receive more. Can I enlarge the buffer size of DataInputStream somehow? Socket receive buffer is 16384 as returned by sock.getReceiveBufferSize() so it's not the socket limiting me to 2048 bytes. If there is no way to increase the DataInputStream buffer size - I guess the only way is to declare my own buffer and read everything from DataInputStream to that buffer? Regards

    Read the article

  • Can an application affect TCP retransmits

    - by sipwiz
    I'm troubleshooting some communications issues and in the network traces I am occasionally coming across TCP sequence errors. One example I've got is: Server to Client: Seq=3174, Len=50 Client to Server: Ack=3224 Server to Client: Seq=3224, Len=50 Client to Server: Ack=3224 Server to Client: Seq=3274, Len=10 Client to Server: Ack=3224, SLE=3274, SRE=3284 Packets 4 & 5 are recorded in the trace (which is from a router in between the client and server) at almost exactly the same time so they most likely crossed in transit. The TCP session has got out of sync with the client missing the last two transmissions from the server. Those two packets should have been retransmitted but they weren't, the next log in the trace is a RST packet from the Client 24 seconds after packet 6. My question is related to what could be responsible for the failure to retransmit the server data from packets 3 & 5? I would assume that the retransmit would be at the operating system level but is there anyway the application could influence it and stop it being sent? A thread blocking or put to sleep or something like that?

    Read the article

  • Decompress a GZipped response from the server (Socket)

    - by Lith
    Umm, ok, after sending some data to the server, noting this particular part: "Accept-Encoding: gzip,deflate\r\n" I am getting the following response: HTTP/1.1 200 OK Server: nginx Date: Fri, 09 Apr 2010 23:25:27 GMT Content-Type: text/html; charset=UTF-8 Transfer-Encoding: chunked Connection: keep-alive X-Powered-By: PHP/5.2.8 Expires: Mon, 26 Jul 1997 05:00:00 GMT Last-Modified: Fri, 09 Apr 2010 23:25:27 GMT Cache-Control: no-store, no-cache, must-revalidate Cache-Control: post-check=0, pre-check=0 Pragma: no-cache Content-Encoding: gzip Vary: Accept-Encoding 7aa ??U-?Rh?%?2?w??PM]??7?qZ?K?)???2?&??m???"q??/p9w?????x?[`tA!G???G?5z??????a>k????????Q ???N?? ('??f?,(??Y:5B???-?)?3x^0e:j?`,???**???F>G)?2????@???b??????A?k???Ar?n? But how do I decompress it? Note that I am using the Socket Class to do all the work. I know how to decompress it, but the problem here lies in the fact that I cannot separate the Packet from the GZipped data, psuedo-psuedocode (or whatever) on how I do it: Socket sends packet; Socket reads response from server, stores into a ByteArray; Create MemoryStream, use ByteArray; Create GZipStream, use Memorystream; now the problem occurs; I am getting the following Error: System.IO.InvalidDataException The magic number in GZip header is not correct. Make sure you are passing in a GZip stream. I hope the explanation is clear enough __.

    Read the article

  • node.js UDP data lost at high package rates

    - by koleto
    I am observing a significant data-lost on a UDP connection with node.js 0.6.18 and 0.8.0 . It appears at high packet rates about 1200 packet per second with frames about 1500 byte limit. Each data packages has a incrementing number so it easy to track the number of lost packages. var server = dgram.createSocket("udp4"); server.on("message", function (message, rinfo) { //~processData(message); //~ writeData(message, null, 5000); }).bind(10001); On the receiving callback I tested two cases I first saved 5000 packages in a file. The result ware no dropped packages. After I have included a data processing routine and got about 50% drop rate. What I expected was that the process data routine should be completely asynchronous and should not introduce dead time to the system, since it is a simple parser to process binary data in the package and to emits events to a further processing routine. It seems that the parsing routine introduce dead time in which the event handler is unable to handle each packets. At the low package rates (< 1200 packages/sec) there are no data lost observed! Is this a bug or I am doing something wrong?

    Read the article

  • no default constructor exists for class

    - by MixedCoder
    #include "Includes.h" enum BlowfishAlgorithm { ECB, CBC, CFB64, OFB64, }; class Blowfish { public: struct bf_key_st { unsigned long P[18]; unsigned long S[1024]; }; Blowfish(BlowfishAlgorithm algorithm); void Dispose(); void SetKey(unsigned char data[]); unsigned char Encrypt(unsigned char buffer[]); unsigned char Decrypt(unsigned char buffer[]); char EncryptIV(); char DecryptIV(); private: BlowfishAlgorithm _algorithm; unsigned char _encryptIv[200]; unsigned char _decryptIv[200]; int _encryptNum; int _decryptNum; }; class GameCryptography { public: Blowfish _blowfish; GameCryptography(unsigned char key[]); void Decrypt(unsigned char packet[]); void Encrypt(unsigned char packet[]); Blowfish Blowfish; void SetKey(unsigned char k[]); void SetIvs(unsigned char i1[],unsigned char i2[]); }; GameCryptography::GameCryptography(unsigned char key[]) { } Error:IntelliSense: no default constructor exists for class "Blowfish" ???!

    Read the article

  • when to use Hibernate vs. Simple ResultSets for small application

    - by luke
    I just started working on upgrading a small component in a distributed java application. The main application is a rather complicated applet/servlet combo running on JBoss and it extensively uses Hibernate for its DataAccess. The component i am working on however is very a very straightforward data importing service. Basically the workflow is Listen for a network event Parse the data packet, extract a set of identifiers Map the identifier set to a primary key in our database Parse the rest of the packet and insert items in a related table using the foreign key found in step 3 Repeat in the previous version of this component it used a hibernate based DAL, that is no longer usable for a variety of reasons (in particular it is EOL), so I am in charge of replacing the Data Access layer for this component. So on the one hand I think i should use Hibernate because that's what the rest of the application does, but on the other i think i should just use regular java.sql.* classes because my requirements are really straightforward and aren't expected to change any time soon. So my question is (and i understand it is subjective) at what point do you think that the added complexity of using an ORM tool (in terms of configuration, dependencies...) is worth it? UPDATE due to the way the DataAccesLayer for the main application was written (weird dependencies) i cannot easily use it, i would have to implement it myself.

    Read the article

  • How to read/write from erlang to a named pipe ?

    - by cstar
    I need my erlang application to read and write through a named pipe. Opening the named pipe as a file will fail with eisdir. I wrote the following module, but it is fragile and feels wrong in many ways. Also it fails on reading after a while. Is there a way to make it more ... elegant ? -module(port_forwarder). -export([start/2, forwarder/2]). -include("logger.hrl"). start(From, To)-> spawn(fun() -> forwarder(From, To) end). forwarder(FromFile, ToFile) -> To = open_port({spawn,"/bin/cat > " ++ ToFifo}, [binary, out, eof,{packet, 4}]), From = open_port({spawn,"/bin/cat " ++ FromFifo}, [binary, in, eof, {packet, 4}]), forwarder(From, To, nil). forwarder(From, To, Pid) -> receive {Manager, {command, Bin}} -> ?ERROR("Sending : ~p", [Bin]), To ! {self(), {command, Bin}}, forwarder(From, To, Manager); {From ,{data,Data}} -> Pid ! {self(), {data, Data}}, forwarder(From, To, Pid); E -> ?ERROR("Quitting, first message not understood : ~p", [E]) end. As you may have noticed, it's mimicking the port format in what it accepts or returns. I want it to replace a C code that will be reading the other ends of the pipes and being launched from the debugger.

    Read the article

  • Using LINQ, need help splitting a byte array on data received from Silverlight sockets

    - by gcadmes
    The message packats received contains multiple messages deliniated by a header=0xFD and a footer=0xFE // sample message packet with three // different size messages List<byte> receiveBuffer = new List<byte>(); receiveBuffer.AddRange(new byte[] { 0xFD, 1, 2, 0xFE, 0xFD, 1, 2, 3, 4, 5, 6, 7, 8, 0xFE, 0xFD, 33, 65, 25, 44, 0xFE}); // note: this sample code is without synchronization, // statements, error handling...etc. while (receiveBuffer.Count > 0) { var bytesInRange = receiveBuffer.TakeWhile(n => n != 0xFE); foreach (var n in bytesInRange) Console.WriteLine(n); // process message.. // 1) remove bytes read from receive buffer // 2) construct message object... // 3) etc... receiveBuffer.RemoveRange(0, bytesInRange.Count()); } As you can see, (including header/footer) the first message in this message packet contains 4 bytes, and the 2nd message contains 10 bytes,a and the 3rd message contains 6 bytes. In the while loop, I was expecting the TakeWhile to add the bytes that did not equal the footer part of the message. Note: Since I am removing the bytes after reading them, the header can always be expected to be at position '0'. I searched examples for splitting byte arrays, but non demonstrated splitting on arrays of unknown and fluctuating sizes. Any help will be greatly appreciated. thanks much!

    Read the article

  • Can't ping host from vmware guest using bridged networking

    - by user199421
    Host is Windows 7 Guest is Ubuntu 11.04 Network adapter is wireless I can ping other computers on the network but not the host. No firewall are involved. Sniffing the traffic with wireshark it looks like both the host and the guest are using the same MAC address. My guest simply doesn't receive a reply when asking for 192.168.1.101 (the host) My router has no problem giving both of them different IP addresses but maybe duplicate MAC address is the problem? It seems logical that both will have the same MAC address (from the host point of view) but it strange that there is no work around for this because otherwise I don't see how the host and guess are supposed to communicate.

    Read the article

  • How to allow local LAN access while connected to Cisco VPN?

    - by Ian Boyd
    How can I maintain local LAN access while connected to Cisco VPN? When connecting using Cisco VPN, the server has to ability to instruct the client to prevent local LAN access. Assuming this server-side option cannot be turned off, how can allow local LAN access while connected with a Cisco VPN client? I used to think it was simply a matter of routes being added that capture LAN traffic with a higher metric, for example: Network Destination Netmask Gateway Interface Metric 10.0.0.0 255.255.0.0 10.0.0.3 10.0.0.3 20 <--Local LAN 10.0.0.0 255.255.0.0 192.168.199.1 192.168.199.12 1 <--VPN Link And trying to delete the 10.0.x.x -> 192.168.199.12 route don't have any effect: >route delete 10.0.0.0 >route delete 10.0.0.0 mask 255.255.0.0 >route delete 10.0.0.0 mask 255.255.0.0 192.168.199.1 >route delete 10.0.0.0 mask 255.255.0.0 192.168.199.1 if 192.168.199.12 >route delete 10.0.0.0 mask 255.255.0.0 192.168.199.1 if 0x3 And while it still might simply be a routing issue, attempts to add or delete routes fail. At what level is Cisco VPN client driver doing what in the networking stack that takes overrides a local administrator's ability to administer their machine? The Cisco VPN client cannot be employing magic. It's still software running on my computer. What mechanism is it using to interfere with my machine's network? What happens when an IP/ICMP packet arrives on the network? Where in the networking stack is the packet getting eaten? See also No internet connection with Cisco VPN Cisco VPN Client interrupts connectivity to my LDAP server Cisco VPN stops Windows 7 Browsing How can I prohibit the creation of a route in Windows XP upon connection to Cisco VPN? Rerouting local LAN and Internet traffic when in VPN VPN Client "Allow local LAN Access" Allow Local LAN Access for VPN Clients on the VPN 3000 Concentrator Configuration Example LAN access gone when I connect to VPN Windows XP Documentation: Route Edit: Things I've not yet tried: >route delete 10.0.* Update: Since Cisco has abandoned their old client, in favor of AnyConnect (HTTP SSL based VPN), this question, unsolved, can be left as a relic of history. Going forward, we can try to solve the same problem with their new client.

    Read the article

  • How to troubleshoot a remote wmi query/access failure?

    - by Roman
    Hi I'm using Powershell to query a remote computer in a domain for a wmi object, eg: "gwmi -computer test -class win32_bios". I get this error message: Value does not fall within the expected range Executing the query local under the same user works fine. It seems to happen on both windows 2003 and also 2008 systems. The user that runs the shell has admin rights on the local and remote server. I checked wmi and dcom permissions as far as I know how to do this, they seem to be the same on a server where it works, and another where it does not. I think it is not a network issue, all ports are open that are needed, and it also happens within the same subnet. When sniffing the traffic we see the following errors: RPC: c/o Alter Cont Resp: Call=0x2 Assoc Grp=0x4E4E Xmit=0x16D0 Recv=0x16D0 Warning: GssAPIMechanism is not found, either caused by not reassembled, conversation off or filtering. And an errormessage from Kerberos: Kerberos: KRB_ERROR - KDC_ERR_BADOPTION (13) The option code in the packet is 0x40830000 Any idea what I should look into?

    Read the article

  • How can I filter packets from a port monitor?

    - by engineerchuan
    I have some data going from Point A to Point B. I have a SPAN monitor set up to a monitoring device C. To recreate some real world scenarios, I want to filter out all traffic which is a certain type (H.323 VoIP Signaling Packets) so that C sees a subset of the information that is flowing from A to B. What would the easiest way to do this be? I assume I would need a computer with 2 NIC cards and some software to examine each packet and chuck out the H.323 VoIP packets? Thanks!

    Read the article

  • checksum in raw sockets and pcap

    - by hero
    i am using pcap library to sniff some packets, change their tcp data , and then inject my packet on the network. my question is: if i changed in the tcp data, should i recalculate the length field in the tcp header? should i also change the checksum? i read in a page on how to create raw sockets that if you set the tcp_checksum to 0, the kernel will automatically calculate it and fill it, is this true for windows machines also?

    Read the article

< Previous Page | 28 29 30 31 32 33 34 35 36 37 38 39  | Next Page >