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  • 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).

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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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  • 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?

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  • 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.

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  • 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)

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  • 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

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  • 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

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  • 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?

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  • 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 __.

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  • 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?

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  • 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" ???!

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  • 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.

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  • 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.

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  • 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!

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  • 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.

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  • 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!

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  • 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?

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  • Determining the health of a Cisco switch port?

    - by ewwhite
    I've been chasing a packet-loss and network stability issue for a handful of end-users on an internal network for the past few days... These issues surfaced recently, however, the location was struck by lightning six weeks ago. I was seeing 5-10% packet loss between a stack of four Cisco 2960's and several PC's and phones on the other side of a 77-meter run. The PC's were run inline with the phones over a trunked link. We were seeing dropped calls and interruptions in client-server applications and Microsoft Exchange connectivity. I tried the usual troubleshooting steps remotely, having a local technician do the following during breaks in user and production activity: change cables between the wall jack and device. change patch cables between the patch panel and switch port(s). try different switch ports within the 2960 stack. change end-user devices with known-good equipment (new phones, different PC's). clear switch port interface counters and monitor incrementing errors closely. (Pastebin output of sh int) Pored over the device logs and Observium RRD graphs. No link up/down issues from the switch side. change power strips on the end-user side. test cable runs from the Cisco 2960 using test cable-diagnostics tdr int Gi4/0/9 (clean)* test cable runs with a Tripp-Lite cable tester. (clean) run diagnostics on the switch stack members. (clean) In the end, it took three changes of switch ports to find a stable solution. The only logical conclusion is that a few Cisco 2960 switch ports are bad or flaky... Not dead, but not consistent in behavior either. I'm not used to seeing individual ports die in this manner. What else can I test or check to determine if these devices are bad? Is it common for single ports to have problems, rather than a contiguous bank of ports? BTW - show cable-diagnostics tdr int Gi4/0/14 is very cool... Interface Speed Local pair Pair length Remote pair Pair status --------- ----- ---------- ------------------ ----------- -------------------- Gi4/0/14 1000M Pair A 79 +/- 0 meters Pair B Normal Pair B 75 +/- 0 meters Pair A Normal Pair C 77 +/- 0 meters Pair D Normal Pair D 79 +/- 0 meters Pair C Normal

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  • IP failover with 2 nodes on different subnet: cannot ping virtual IP from second node?

    - by quanta
    I'm going to setup redundant failover Redmine: another instance was installed on the second server without problem MySQL (running on the same machine with Redmine) was configured as master-master replication Because they are in different subnet (192.168.3.x and 192.168.6.x), it seems that VIPArip is the only choice. /etc/ha.d/ha.cf on node1 logfacility none debug 1 debugfile /var/log/ha-debug logfile /var/log/ha-log autojoin none warntime 3 deadtime 6 initdead 60 udpport 694 ucast eth1 node2.ip keepalive 1 node node1 node node2 crm respawn /etc/ha.d/ha.cf on node2: logfacility none debug 1 debugfile /var/log/ha-debug logfile /var/log/ha-log autojoin none warntime 3 deadtime 6 initdead 60 udpport 694 ucast eth0 node1.ip keepalive 1 node node1 node node2 crm respawn crm configure show: node $id="6c27077e-d718-4c82-b307-7dccaa027a72" node1 node $id="740d0726-e91d-40ed-9dc0-2368214a1f56" node2 primitive VIPArip ocf:heartbeat:VIPArip \ params ip="192.168.6.8" nic="lo:0" \ op start interval="0" timeout="20s" \ op monitor interval="5s" timeout="20s" depth="0" \ op stop interval="0" timeout="20s" \ meta is-managed="true" property $id="cib-bootstrap-options" \ stonith-enabled="false" \ dc-version="1.0.12-unknown" \ cluster-infrastructure="Heartbeat" \ last-lrm-refresh="1338870303" crm_mon -1: ============ Last updated: Tue Jun 5 18:36:42 2012 Stack: Heartbeat Current DC: node2 (740d0726-e91d-40ed-9dc0-2368214a1f56) - partition with quorum Version: 1.0.12-unknown 2 Nodes configured, unknown expected votes 1 Resources configured. ============ Online: [ node1 node2 ] VIPArip (ocf::heartbeat:VIPArip): Started node1 ip addr show lo: 1: lo: <LOOPBACK,UP,LOWER_UP> mtu 16436 qdisc noqueue link/loopback 00:00:00:00:00:00 brd 00:00:00:00:00:00 inet 127.0.0.1/8 scope host lo inet 192.168.6.8/32 scope global lo inet6 ::1/128 scope host valid_lft forever preferred_lft forever I can ping 192.168.6.8 from node1 (192.168.3.x): # ping -c 4 192.168.6.8 PING 192.168.6.8 (192.168.6.8) 56(84) bytes of data. 64 bytes from 192.168.6.8: icmp_seq=1 ttl=64 time=0.062 ms 64 bytes from 192.168.6.8: icmp_seq=2 ttl=64 time=0.046 ms 64 bytes from 192.168.6.8: icmp_seq=3 ttl=64 time=0.059 ms 64 bytes from 192.168.6.8: icmp_seq=4 ttl=64 time=0.071 ms --- 192.168.6.8 ping statistics --- 4 packets transmitted, 4 received, 0% packet loss, time 3000ms rtt min/avg/max/mdev = 0.046/0.059/0.071/0.011 ms but cannot ping virtual IP from node2 (192.168.6.x) and outside. Did I miss something? PS: you probably want to set IP2UTIL=/sbin/ip in the /usr/lib/ocf/resource.d/heartbeat/VIPArip resource agent script if you get something like this: Jun 5 11:08:10 node1 lrmd: [19832]: info: RA output: (VIPArip:stop:stderr) 2012/06/05_11:08:10 ERROR: Invalid OCF_RESK EY_ip [192.168.6.8] http://www.clusterlabs.org/wiki/Debugging_Resource_Failures Reply to @DukeLion: Which router receives RIP updates? When I start the VIPArip resource, ripd was run with below configuration file (on node1): /var/run/resource-agents/VIPArip-ripd.conf: hostname ripd password zebra debug rip events debug rip packet debug rip zebra log file /var/log/quagga/quagga.log router rip !nic_tag no passive-interface lo:0 network lo:0 distribute-list private out lo:0 distribute-list private in lo:0 !metric_tag redistribute connected metric 3 !ip_tag access-list private permit 192.168.6.8/32 access-list private deny any

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  • WOL not working with two NIC's

    - by amazinghorse24
    We have recently moved from a windows 2003 server to an 08 R2. Running the program from the command line says that the magic packet has successfully been sent but it never actually reached the destination computer. We have two NIC's on the server. If I disable the NIC that isn't (currently) in use, WOL will work fine, but in the future we will need to use both NIC's. Also, we use MAC Addresses for WOL.

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  • OWA, Outlook Anywhere, RPCPing Inconsistencies

    - by pk.
    I'm troubleshooting an Outlook Anywhere issue with a new Exchange 2010 server. The server in question, MS2010, is behind a SonicWALL NSA 2400 device and works wonderfully except for Outlook Anywhere. Outlook Anywhere works internally and I've verified (through Ctrl-Right Click --> Connection Status) that I'm able to connect to MS2010 over HTTPS. When trying to connect to the server using HTTPS from outside the firewall, I'm unable to do so. A Wireshark trace shows 30 or so successful HTTPS packet transmissions, and then it fails with 3 straight transmissions to a destination port of 135. I have no idea why my computer is attempting to access anything on port 135 since I've setup my profile to use HTTPS on both slow and fast connections. I'm 99% certain that the firewall is configured correctly. I run Outlook Web Access (also HTTPS) on the same server and there are no issues with access. EDIT: My Autodiscover settings are correct (as far as I can tell). My server passes the Outlook Anywhere and Autodiscover tests at https://www.testexchangeconnectivity.com/. I've been using the RPCPing utility to troubleshoot and have come across the following results: Internally- >rpcping -t ncacn_http -s mail.mydomain.com -o RpcProxy=mail.mydomain.com -P "pk,mydomain,*" -I "pk,mydomain,*" -H 1 -u 10 -a connect -F 3 -v 3 -E -R none RPCPing v2.12. Copyright (C) Microsoft Corporation, 2002 OS Version is: 6.1, Service Pack 1 RPCPinging proxy server mail.mydomain.com with Echo Request Packet Sending ping to server Response from server received: 200 Pinging successfully completed in 93 ms Externally- >rpcping -t ncacn_http -s mail.mydomain.com -o RpcProxy=mail.mydomain.com -P "pk,mydomain,*" -I "pk,mydomain,*" -H 1 -u 10 -a connect -F 3 -v 3 -E -R none RPCPing v6.0. Copyright (C) Microsoft Corporation, 2002-2006 Enter password for RPC/HTTP proxy: RPCPing set Activity ID: {fc8411ba-2987-4175-b37b-801dc69d5ff9} RPCPinging proxy server mail.mydomain.com with Echo Request Packet Setting autologon policy to high WinHttpSetCredentials for target server called Error 87 : The parameter is incorrect. returned in WinHttpSetCredentials Ping failed What should I be checking in order to troubleshoot my Outlook Anywhere issues? I'm using Windows 7 SP1 for internal and external access. EDIT: Autodiscover.xml content <?xml version="1.0"?> <Autodiscover xmlns:xsd="http://www.w3.org/2001/XMLSchema" xmlns:xsi="http://www.w3.org/2001/XMLSchema-instance" xmlns="http://schemas.microsoft.com/exchange/autodiscover/responseschema/2006"> <Response xmlns="http://schemas.microsoft.com/exchange/autodiscover/outlook/responseschema/2006a"> <User> <DisplayName>John Doe</DisplayName> <LegacyDN>/o=MYDOMAIN/ou=Exchange Administrative Group (FYDIBOHF23SPDLT)/cn=Recipients/cn=pk</LegacyDN> <DeploymentId>d35170cc-f3a7-42c5-9427-1f554a469126</DeploymentId> </User> <Account> <AccountType>email</AccountType> <Action>settings</Action> <Protocol> <Type>EXCH</Type> <Server>MS2010.MYDOMAIN.local</Server> <ServerDN>/o=MYDOMAIN/ou=Exchange Administrative Group (FYDIBOHF23SPDLT)/cn=Configuration/cn=Servers/cn=MS2010</ServerDN> <ServerVersion>738180DA</ServerVersion> <MdbDN>/o=MYDOMAIN/ou=Exchange Administrative Group (FYDIBOHF23SPDLT)/cn=Configuration/cn=Servers/cn=MS2010/cn=Microsoft Private MDB</MdbDN> <ASUrl>https://MS2010.MYDOMAIN.local/EWS/Exchange.asmx</ASUrl> <OOFUrl>https://MS2010.MYDOMAIN.local/EWS/Exchange.asmx</OOFUrl> <OABUrl>http://MS2010.MYDOMAIN.local/OAB/2c34c9f5-5521-4c8c-b684-538df815052a/</OABUrl> <UMUrl>https://MS2010.MYDOMAIN.local/EWS/UM2007Legacy.asmx</UMUrl> <Port>0</Port> <DirectoryPort>0</DirectoryPort> <ReferralPort>0</ReferralPort> <PublicFolderServer>MS2007.MYDOMAIN.local</PublicFolderServer> <AD>dc1.MYDOMAIN.local</AD> <EwsUrl>https://MS2010.MYDOMAIN.local/EWS/Exchange.asmx</EwsUrl> <EcpUrl>https://MS2010.MYDOMAIN.local/ecp/</EcpUrl> <EcpUrl-um>?p=customize/voicemail.aspx&amp;exsvurl=1</EcpUrl-um> <EcpUrl-aggr>?p=personalsettings/EmailSubscriptions.slab&amp;exsvurl=1</EcpUrl-aggr> <EcpUrl-mt>PersonalSettings/DeliveryReport.aspx?exsvurl=1&amp;IsOWA=&lt;IsOWA&gt;&amp;MsgID=&lt;MsgID&gt;&amp;Mbx=&lt;Mbx&gt;</EcpUrl-mt> <EcpUrl-ret>?p=organize/retentionpolicytags.slab&amp;exsvurl=1</EcpUrl-ret> <EcpUrl-sms>?p=sms/textmessaging.slab&amp;exsvurl=1</EcpUrl-sms> </Protocol> <Protocol> <Type>EXPR</Type> <Server>mail.mycompany.com</Server> <ASUrl>https://mail.mycompany.com/ews/exchange.asmx</ASUrl> <OOFUrl>https://mail.mycompany.com/ews/exchange.asmx</OOFUrl> <OABUrl>https://mail.mycompany.com/OAB/2c34c9f5-5521-4c8c-b684-538df815052a/</OABUrl> <UMUrl>https://mail.mycompany.com/ews/UM2007Legacy.asmx</UMUrl> <Port>0</Port> <DirectoryPort>0</DirectoryPort> <ReferralPort>0</ReferralPort> <SSL>On</SSL> <AuthPackage>Basic</AuthPackage> <CertPrincipalName>msstd:mail.mycompany.com</CertPrincipalName> <EwsUrl>https://mail.mycompany.com/ews/exchange.asmx</EwsUrl> <EcpUrl>https://mail.mycompany.com/owa/</EcpUrl> <EcpUrl-um>?p=customize/voicemail.aspx&amp;exsvurl=1</EcpUrl-um> <EcpUrl-aggr>?p=personalsettings/EmailSubscriptions.slab&amp;exsvurl=1</EcpUrl-aggr> <EcpUrl-mt>PersonalSettings/DeliveryReport.aspx?exsvurl=1&amp;IsOWA=&lt;IsOWA&gt;&amp;MsgID=&lt;MsgID&gt;&amp;Mbx=&lt;Mbx&gt;</EcpUrl-mt> <EcpUrl-ret>?p=organize/retentionpolicytags.slab&amp;exsvurl=1</EcpUrl-ret> <EcpUrl-sms>?p=sms/textmessaging.slab&amp;exsvurl=1</EcpUrl-sms> </Protocol> <Protocol> <Type>WEB</Type> <Port>0</Port> <DirectoryPort>0</DirectoryPort> <ReferralPort>0</ReferralPort> <Internal> <OWAUrl AuthenticationMethod="Basic, Fba">https://MS2010.MYDOMAIN.local/owa/</OWAUrl> <Protocol> <Type>EXCH</Type> <ASUrl>https://MS2010.MYDOMAIN.local/EWS/Exchange.asmx</ASUrl> </Protocol> </Internal> <External> <OWAUrl AuthenticationMethod="Fba">https://mail.mycompany.com/owa/</OWAUrl> <Protocol> <Type>EXPR</Type> <ASUrl>https://mail.mycompany.com/ews/exchange.asmx</ASUrl> </Protocol> </External> </Protocol> </Account> </Response> </Autodiscover>

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  • OpenVPN, Great on Windows, VERY slow on Mac...

    - by Phsion
    Hello, I'm not really an IT Pro, but this seemed like the best place to ask this question... I have setup VPN networks in the past, for fun, and everything was great, but now I've set one up for my boss, and while my computers all work great, his Mac machines are almost too slow to work with. Its pretty much vanilla configs all around, anyone have any ideas? Its a TUN routing setup over UDP. Back Story: My boss travels a lot, and wants to be able to access all his files from the road, and is also pretty paranoid about security (even though knows almost nothing about computers). SO i figured a VPN would be the answer. I went with OpenVPN, but there are some other issues. The only ISP we can get in our area besides Dial-UP is a crappy Satellite provider, that doesn't offer public IPs unless your willing to pay, so while the computers and VPN setup are pretty vanilla, the routing and structure is strange to get around this limitation. Specs: Its OpenVPN2, and there are six machines using it (only three actually use it, the rest are my test machines), one Windows 7 laptop, two XP Desktops, one OS X 10.5 Desktop, one 10.6 Desktop, and one 10.6 Laptop. One XP Desktop sits at my house and acts as the server (6Mbs/2Mbs FIOS connection). One XP desktop sits at the office and hosts a webpage that will wake up the Main Mac Desktop from sleep, and also ping all the machines on the VPN and show their status. The main office mac (10.6) stays in sleep mode until it gets the Wake-On-Lan packet from the Office XP, and then it auto connects to the VPN and opens itself up. The reason for all this is the Satellite private IP crap means i cant directly access the office machines outside of the LAN, so everyone connects to my house first, then they talk to each other from there. The Wake On Lan weirdness is because my boss doesn't want to leave the main Mac on all the time, and making a quick and dirty webpage was the easiest way to send a Magic Packet from inside the LAN without confusing my boss. The VPN uses Client Config files to make static IPs for the client. The only thing i found in google was some changes to the VPN MTU settings (down to 1400) but no real help. Oh, and i forgot...all the windows machines just have OpenVPN start as a service. The Mac laptop uses tunnelblick (an OpenVPN GUI) and the Mac Desktops use OpenVPN in normal command line mode. Server Config: tun-mtu 1500 fragment 1450 mssfix 1450 management localhost #### port #### proto udp dev tun ca ####### cert ####### key ###### dh ###### server 10.8.0.0 255.255.255.0 ifconfig-pool-persist ipp.txt client-config-dir ccd route 10.8.0.0 255.255.255.252 client-to-client keepalive 10 120 comp-lzo persist-key persist-tun status openvpn-status log Client Configs (all are simple variations on this) tun-mtu 1500 fragment 1450 mssfix 1450 client dev tun proto udp remote ######## #### resolv-retry infinite nobind persist-key presist-tun ca ##### cert ##### key ##### ns-cert-type server comp-lzo verb 3

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  • Certain SFTP user cannot connect

    - by trobrock
    I have my Ubuntu Server set up so users with the group of sftponly can connect with sftp, but have a shell of /bin/false, and they connect to their home directories. This is working fine with three of the user accounts I have. But I added a new user account today the same way that I added the others and it will not successfully connect. sftp -vvv user@hostname debug1: Next authentication method: password user@hostname's password: debug3: packet_send2: adding 48 (len 73 padlen 7 extra_pad 64) debug2: we sent a password packet, wait for reply debug1: Authentication succeeded (password). debug2: fd 5 setting O_NONBLOCK debug3: fd 6 is O_NONBLOCK debug1: channel 0: new [client-session] debug3: ssh_session2_open: channel_new: 0 debug2: channel 0: send open debug1: Requesting [email protected] debug1: Entering interactive session. debug1: channel 0: free: client-session, nchannels 1 debug3: channel 0: status: The following connections are open: #0 client-session (t3 r-1 i0/0 o0/0 fd 5/6 cfd -1) debug3: channel 0: close_fds r 5 w 6 e 7 c -1 debug1: fd 0 clearing O_NONBLOCK debug3: fd 1 is not O_NONBLOCK Connection to hostname closed by remote host. Transferred: sent 2176, received 1848 bytes, in 0.0 seconds Bytes per second: sent 127453.3, received 108241.6 debug1: Exit status -1 Connection closed For a successful user: sftp -vvv good_user@hostname debug1: Next authentication method: password good_user@hostname's password: debug3: packet_send2: adding 48 (len 63 padlen 17 extra_pad 64) debug2: we sent a password packet, wait for reply debug1: Authentication succeeded (password). debug2: fd 5 setting O_NONBLOCK debug3: fd 6 is O_NONBLOCK debug1: channel 0: new [client-session] debug3: ssh_session2_open: channel_new: 0 debug2: channel 0: send open debug1: Requesting [email protected] debug1: Entering interactive session. debug2: callback start debug2: client_session2_setup: id 0 debug1: Sending subsystem: sftp debug2: channel 0: request subsystem confirm 1 debug2: fd 3 setting TCP_NODELAY debug2: callback done debug2: channel 0: open confirm rwindow 0 rmax 32768 debug2: channel 0: rcvd adjust 2097152 debug2: channel_input_status_confirm: type 99 id 0 debug2: subsystem request accepted on channel 0 debug2: Remote version: 3 debug2: Server supports extension "[email protected]" revision 1 debug2: Server supports extension "[email protected]" revision 2 debug2: Server supports extension "[email protected]" revision 2 debug3: Sent message fd 3 T:16 I:1 debug3: SSH_FXP_REALPATH . -> / sftp> I cannot figure out why one user will work and the other wont, I have restart the ssh service after adding the user. I have even removed the user and added them again to be sure I am adding it correctly.

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  • Optical multicast

    - by Randomblue
    I have a 10G XPF+ optical cable with market updates from a stock exchange. This cable goes into a switch, which then multicasts every packet to a couple of computers. The problem with using a switch for multicast is that there is latency overhead, even with a pass-through switch (~200ns). Are there "optical" solutions (I'm thinking of a beam splitter of some sort) which would allow for close to zero latency 10G multicast?

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