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  • Does it make sense to have several UDP ports ready? Will packets be dropped?

    - by Gubatron
    I'm coding a networking application on Android. I'm thinking of having a single UDP port and Datagram socket that receives all the datagrams that are sent to it and then have different processing queues for these messages. I'm doubting if I should have a second or third UDP socket on standby. Some messages will be very short (100bytes or so), but others will have to transfer files. My concern is, will the Android kernel drop the small messages if it's too busy handling the bigger ones?

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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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  • Packet loss rate with iperf and tcpdump

    - by stefita
    I tested a line for its link quality with iperf. The measured speed (UDP port 9005) was 96Mbps, which is fine, because both servers are connected with 100Mbps to the internet. On the other hand the datagram loss rate was shown to be 3.3-3.7%, which I found a little too much. Using a high-speed transfer protocol I recorded the packets on both sides with tcpdump. Than I calculated the packet loss - average 0.25%. Have anyone an explanation, where this big difference may be coming from? What is an acceptable packet loss in your opinion?

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  • Where is the actual content in a TCP segment

    - by packetloss
    When I email something or download a program, or do anything else over a network, where in the segment is the actual content? If I am emailing a 20KB word document, and the maximum data field size in a segment is 1500 bytes, does that mean it takes about 14 segments to mail my document wherever it is going? I get, I think, the OSI model and I have a decent grasp of the IP protocol. I think I understand the concept of header wrapping of each successive layer in the protocol stack. What I can't get a definitive answer to is where does the actual content go in a TCP segment? Is that the datagram? Maybe the fact I am asking proves I have no clue... Many thanks.

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  • Creating a secure multicast tunnel with socat

    - by ams
    How we can create secure multicast tunnels ith socat? Assume that we have a list of IP address, CIDR network addresses that we want to create secure tunnel to them. I found this: socat STDIO UDP4-DATAGRAM:224.1.0.1:6666,range=192.168.10.0/24 but I want a secure tunnel and different adds with net addrs I want to create script that give the IPs and net addresses and creates a secure tunnel ./myscript IP1 NetAdd1 IP2 NetAdd2 .... How can I send these parameters to socat? Does socat multicast have any limits?

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  • how're routing tables populated?

    - by Robbie Mckennie
    i've been reading "tcp/ip illustrated" and i started reading about ip forwarding. all about how you can receive a datagram and work out where to send it next based on the desination ip and your routing table. but what confused me is how (in a home network setting) the table itself is populated. is there a lower layer protocol at work here? does it come along with dhcp? or is it simply based on the ip address and netmask of each interface? i do know (from other books) that in the early days of ethernet one had to set up routing tables by hand, but i know i didn't do that.

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  • creating secure multicast with socat

    - by arash
    How we can create secure tunnels multicast with socat? Assume we have a list of ip address, CIDR network addresses that we want to create secure tunnel to them. I found this socat STDIO UDP4-DATAGRAM:224.1.0.1:6666,range=192.168.10.0/24 but I want a secure tunnel and different adds with net addrs I want to create script that give the IPs and net addresses and create secure tunnel ./myscript IP1 NetAdd1 IP2 NetAdd2 .... how can i send this parametersw to socat? Socat multicast hasn't any limits? Thanks for your help

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  • Can a network interface be configured to have a default gateway for UDP packets?

    - by Vaibhav
    It is quite possible that my question may not make a lot of sense. I apologize, but I am not a networking guy, and that's my excuse. To elaborate, WikiPedia defines "Default Gateway" as a node on a "TCP/IP" network. And the way it works is that if a network interface is sending a packet to an IP address not present on its subnet, it sends it out to the default gateway (which then knows what to do with that packet). Is this true if a UDP packet (datagram) is involved? I mean, if my network interface is sending a UDP packet to an IP address that is not present on its subnet, would it automatically send it to the Default Gateway as well?

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  • Announcing Unbreakable Enterprise Kernel Release 3 for Oracle Linux

    - by Lenz Grimmer
    We are excited to announce the general availability of the Unbreakable Enterprise Kernel Release 3 for Oracle Linux 6. The Unbreakable Enterprise Kernel Release 3 (UEK R3) is Oracle's third major supported release of its heavily tested and optimized Linux kernel for Oracle Linux 6 on the x86_64 architecture. UEK R3 is based on mainline Linux version 3.8.13. Some notable highlights of this release include: Inclusion of DTrace for Linux into the kernel (no longer a separate kernel image). DTrace for Linux now supports probes for user-space statically defined tracing (USDT) in programs that have been modified to include embedded static probe points Production support for Linux containers (LXC) which were previously released as a technology preview Btrfs file system improvements (subvolume-aware quota groups, cross-subvolume reflinks, btrfs send/receive to transfer file system snapshots or incremental differences, file hole punching, hot-replacing of failed disk devices, device statistics) Improved support for Control Groups (cgroups)  The ext4 file system can now store the content of a small file inside the inode (inline_data) TCP fast open (TFO) can speed up the opening of successive TCP connections between two endpoints FUSE file system performance improvements on NUMA systems Support for the Intel Ivy Bridge (IVB) processor family Integration of the OpenFabrics Enterprise Distribution (OFED) 2.0 stack, supporting a wide range of Infinband protocols including updates to Oracle's Reliable Datagram Sockets (RDS) Numerous driver updates in close coordination with our hardware partners UEK R3 uses the same versioning model as the mainline Linux kernel version. Unlike in UEK R2 (which identifies itself as version "2.6.39", even though it is based on mainline Linux 3.0.x), "uname" returns the actual version number (3.8.13). For further details on the new features, changes and any known issues, please consult the Release Notes. The Unbreakable Enterprise Kernel Release 3 and related packages can be installed using the yum package management tool on Oracle Linux 6 Update 4 or newer, both from the Unbreakable Linux Network (ULN) and our public yum server. Please follow the installation instructions in the Release Notes for a detailed description of the steps involved. The kernel source tree will also available via the git source code revision control system from https://oss.oracle.com/git/?p=linux-uek3-3.8.git If you would like to discuss your experiences with Oracle Linux and UEK R3, we look forward to your feedback on our public Oracle Linux Forum.

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  • Reading UDP Packets

    - by Thomas Mathiesen
    I am having some trouble dissecting a UDP packet. I am receiving the packets and storing the data and sender-address in variables 'data' and 'addr' with: data,addr = UDPSock.recvfrom(buf) This parses the data as a string, that I am now unable to turn into bytes. I know the structure of the datagram packet which is a total of 28 bytes, and that the data I am trying to get out is in bytes 17:28. I have tried doing this: mybytes = data[16:19] print struct.unpack('>I', mybytes) --> struct.error: unpack str size does not match format And this: response = (0, 0, data[16], data[17], 6) bytes = array('B', response[:-1]) print struct.unpack('>I', bytes) --> TypeError: Type not compatible with array type And this: print "\nData byte 17:", str.encode(data[17]) --> UnicodeEncodeError: 'ascii' codec can't encode character u'\xff' in position 0: ordinal not in range(128) And I am not sure what to try next. I am completely new to sockets and byte-conversions in Python, so any advice would be helpful :) Thanks, Thomas

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  • can I read exactly one UDP packet off a socket?

    - by Brian Palmer
    Using UNIX socket APIs on Linux, is there any way to guarantee that I read one UDP packet, and only one UDP packet? I'm currently reading packets off a non-blocking socket using recvmsg, with a buffer size a little larger than the MTU of our internal network. This should ensure that I can always receive the full UDP packet, but I'm not sure I can guarantee that I'll never receive more than one packet per recvmsg call, if the packets are small. The recvmsg man pages reference the MSG_WAITALL option, which attempts to wait until the buffer is filled. We're not using this, so does that imply that recvmsg will always return after one datagram is read? Is there any way to guarantee this? Ideally I'd like a cross-UNIX solution, but if that doesn't exist is there something Linux specific?

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  • TcpListener Socket still active after program exits.

    - by lnical
    I'm trying to stop a TCP Listener as my program is exiting. I do not care about any data that is currently active on the socket or any of the active client sockets. The socket clean up code is essentially: try { myServer.Server.Shutdown(SocketShutdown.Both) } catch (Exception ex) { LogException(ex) } myServer.Server.Close(0) myServer.Stop() myServer is a TCPListener On some occasions, Shutdown will thrown an exception System.Net.Sockets.SocketException: A request to send or receive data was disallowed because the socket is not connected and (when sending on a datagram socket using a sendto call) no address was supplied at System.Net.Sockets.Socket.Shutdown(SocketShutdown how) When this happens, the socket is never released. Even after the application exits netstat shows the socket is still in the listening state. I have not been able to create definitive reproduction scenerio, it happens at seemingly random times. Client Sockets are cleaned up independently. Do you have any suggestions to help me make this socket die?

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  • Using sys/socket.h functions on windows

    - by BSchlinker
    Hello, I'm attempting to utilize the socket.h functions within Windows. Essentially, I'm currently looking at the sample code at http://beej.us/guide/bgnet/output/html/multipage/clientserver.html#datagram. I understand that socket.h is a Unix function -- is there anyway I can easily emulate that environment while compiling this sample code? Does a different IDE / compiler change anything? Otherwise, I imagine that I need to utilize a virtualized Linux environment, which may be best anyways as the code will most likely be running in a UNIX environment. Thanks.

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  • UDP + total order, non-reliable

    - by disown
    I'm trying to find a version of UDP which just alleviates the restriction of a maximum size of the message sent. I don't care about reliability or partial retransmission, if all chunks arrive I want the message to be assembled from the chunks in sending order and delivered to the listening app. If one or more chunks are missing I would just like to discard the message. The goal is to have a low-latency notification mechanism about real time data, but with the added support for bigger messages than what would fit in an IP datagram. I would like the protocol to be one way only, and not have long connection setup times. An optional feature to be able to respond to a received message wouldn't hurt (a concept of an unreliable connection), but is not necessary.

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  • How send sms 2automatically 2 a paricular no: only?

    - by royalcasanova
    import net.rim.device.api.io.; import net.rim.device.api.system.; import javax.microedition.io.; import java.util.; import java.io.*; public class SendSms extends Application { private static final int MAX_PHONE_NUMBER_LENGTH = 32; private String addr = "15191112222"; private String msg = "This is a test message."; private DatagramConnection _dc = null; private static String _openString = "sms://"; public static void main(String[] args) { new SendSms().enterEventDispatcher(); } public SendSms() { try { _dc = (DatagramConnection)Connector.open(_openString); byte[] data = msg.getBytes(); Datagram d = _dc.newDatagram(_dc.getMaximumLength()); d.setAddress("//" + addr); _dc.send(d); } catch ( IOException e) {} System.exit(0); } }

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  • Interrupt a thread in DatagramSocket.receive

    - by SEK
    I'm building an application that listens on both TCP and UDP, and I've run into some trouble with my shutdown mechanism. When I call Thread.interrupt() on each of the listening threads, the TCP thread is interrupted from listening, whereas the UDP listener isn't. To be specific, the TCP thread uses Socket.accept(), which simply returns (without actually connecting). Whereas the UDP thread uses DatagramSocket.receive, and doesn't exit that method. Is this an issue in my JRE, my OS, or should I just switch to (Datagram)Socket.close()?

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  • ICMP Redirect Theory VS. Application

    - by joeqwerty
    I'm trying to watch ICMP redirects in a lab using Cisco Packet Tracer (version 5.3.2) and I'm not seeing them, which leads me to believe that either my lab configuration isn't correct or my understanding of ICMP redirects isn't correct or that Packet Tracer doesn't support/use ICMP redirects. Here's what I believe to be true regarding ICMP redirects: Routers send ICMP redirects when all of these conditions are met: The interface on which the packet comes into the router is the same interface on which the packet gets routed out. The subnet or network of the source IP address is on the same subnet or network of the next-hop IP address of the routed packet. The datagram is not source-routed. The router kernel is configured to send redirects. I have the lab set up in Cisco Packet Tracer as displayed in the image and would expect to see an ICMP redirect from Router1 when pinging from PC1 to PC3. I'm not seeing the ICMP redirect and it looks like Router1 is actually routing all of the packets via Router2. I have IP ICMP debugging enabled on Router1 (and Router2) and I'm not seeing any ICMP redirect activity in either console. I'm also not seeing a route to the PC3 network in the routing table on PC1, which I think confirms that the ICMP redirect is not occurring. I'm using only static routing on Routers 1 and 2. Is my understanding of ICMP redirects incorrect, or is there a problem with my lab configuration or does Packet Tracer not support/use ICMP redirects?

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  • SMB access from XP to Windows 2008 R2

    - by Pablo
    Here's the thing... I have a very slow file copy performance from Windows XP clients to Windows 2008R2 servers. Here are the facts: Windows XP to Windows 2K3: Fast Windows XP to Windows 2K8: Very Slow Windows 7 to Windows (any): Fast Despite the fact that the obvious solution would be to upgrade to Windows 7, well, we have 900 desktops so it's not an option in the short time. I have tried everything: Disabling SMB2.0, disabling security signatures, changing the TCP Window size, disabling the W2K8 auto tuning, upgraded the drivers, etc. We eliminated the network; both the server and the client are connected to the same core switch (no hops, no routers, same VLAN). Upon monitoring the network with a packet capture utility, we see that the SMB packets being exchanged between the W2K8 and the XP machines are very small packets (256 bytes); despite the fact that the MTUs are properly set (1500) and there is no fragmentation whatsoever. In fact, those SMB packets show, on the IP datagram, that the window is 65535 or close. The same trace, made using the same application but instead of using a W2K8 share uses a Windows XP share (and that goes FAST) shows SMB packets of 4096 bytes. I can post the traces if necessary. So, why does XP-W2K8 negotiation arrange for 24-bytes SMB payload, whereas the XP-XP negotiation arranges for 4096 SMB packets? Any ideas? I am running short of those...

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  • how i can identify which process is making UDP traffic on linux?

    - by boos
    my machine is continously making udp dns traffic request. what i need to know is the PID of the process generating this traffic. The normal way in TCP connection is to use netstat/lsof and get the process associated at the pid. Is UDP the connection is stateles, so, when i call netastat/lsof i can see it only if the UDP socket is opened and it's sending traffic. I have tried with lsof -i UDP and with nestat -anpue but i cant be able to find wich process is doing that request because i need to call lsof/netstat exactly when the udp traffic is sended, if i call lsof/netstat before/after the udp datagram is sended is impossible to view the opened UDP socket. call netstat/lsof exactly when 3/4 udp packet is sended is IMPOSSIBLE. how i can identify the infamous process ? I have already inspected the traffic to try to identify the sended PID from the content of the packet, but is not possible to identify it from the contect of the traffic. anyone can help me ? I'm root on this machine FEDORA 12 Linux noise.company.lan 2.6.32.16-141.fc12.x86_64 #1 SMP Wed Jul 7 04:49:59 UTC 2010 x86_64 x86_64 x86_64 GNU/Linux

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  • why can't i bind ipv6 socket to a linklocal address

    - by Haiyuan Zhang
    #include <sys/types.h> #include <sys/socket.h> #include <netinet/in.h> #include <netdb.h> #include <stdio.h> void error(char *msg) { perror(msg); exit(0); } int main(int argc, char *argv[]) { int sock, length, fromlen, n; struct sockaddr_in6 server; struct sockaddr_in6 from; int portNr = 5555; char buf[1024]; length = sizeof (struct sockaddr_in6); sock=socket(AF_INET6, SOCK_DGRAM, 0); if (sock < 0) error("Opening socket"); bzero((char *)&server, length); server.sin6_family=AF_INET6; server.sin6_addr=in6addr_any; server.sin6_port=htons(portNr); inet_pton( AF_INET6, "fe80::21f:29ff:feed:2f7e", (void *)&server.sin6_addr.s6_addr); //inet_pton( AF_INET6, "::1", (void *)&server.sin6_addr.s6_addr); if (bind(sock,(struct sockaddr *)&server,length)<0) error("binding"); fromlen = sizeof(struct sockaddr_in6); while (1) { n = recvfrom(sock,buf,1024,0,(struct sockaddr *)&from,&fromlen); if (n < 0) error("recvfrom"); write(1,"Received a datagram: ",21); write(1,buf,n); n = sendto(sock,"Got your message\n",17, 0,(struct sockaddr *)&from,fromlen); if (n < 0) error("sendto"); } } when I compile and run the above code I got : binding: Invalid argument and if change to bind the ::1 and leave other thing unchanged in the source code, the code works! so could you tell me what's wrong with my code ? thanks in advance.

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  • Domain: Netlogon event sequence

    - by Bob
    I'm getting really confused, reading tutorials from SAMBA howto, which is hell of a mess. Could you write step-by-step, what events happen upon NetLogon? Or in particular, I can't get these things: I really can't get the mechanism of action of LDAP and its role. Should I think of Active Directory LDS as of its superset? What're the other roles of AD and why this term is nearly a synonym of term "domain"? What's the role of LDAP in the remote login sequence? Does it store roaming user profiles? Does it store anything else? How it is called (are there any upper-level or lower-level services that use it in the course of NetLogon)? How do I join a domain. On the client machine I just use the Domain Controller admin credentials, but how do I prepare the Domain Controller for a new machine to join it. What's that deal of Machine trust accounts? How it is used? Suppose, I've just configured a machine to join a domain, created its machine trust, added its data to the domain controller. How would that machine find WINS server to query it for Domain Controller NetBIOS name? Does any computer name, ending with <1C type, correspond to domain controller? In what cases Kerberos and LM/NTLM are used for authentication? Where are password hashes stored in, say, Windows2000 domain controller? Right in the registry? What is SAM - is it a service, responsible for authentication and sending/storing those passwords and accompanying information, such as groups policies etc.? Who calls it? Does it use Active Directory? What's the role of NetBIOS except by name service? Can you exemplify a scenario of its usage as a "datagram distribution service for connectionless communication" or "session service for connection-oriented communication"? (quoted taken from http://en.wikipedia.org/wiki/NetBIOS_Frames_protocol description of NetBIOS roles) Thanks and sorry for many questions.

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  • Problems with MGCP proxy creation

    - by Popof
    Hi, I'm trying to bypass my ISP router with my FreeBSD server (I've an optical connection so I've a RJ45 used to connect the box to WAN) Internet and TV are working fine (Using igmpproxy to forward TV stream) but I've a problem with phone. ISP's box is connected to the server which gives it a LAN address. The problem is that when the box builds MGCP packets (and especially SDP ones) it uses its LAN address. So I've think of writing an UDP proxy to handle MGCP and SDP packets in order to replace LAN address with server WAN address and then forward packet to WAN. Before starting coding I've captured stream packets using my server as a bridge between WAN connection and the ISP's box. And, in order to see if my solution is viable, I've tried to send those packets to the box using nemesis. I tried to send a packet (found in capture) containing an endpoint audit: AUEP 1447 aaln/[email protected] MGCP 1.0 F: A In the wireshark capture the box replied: 200 1447 OK A: a:PCMU;PCMA;G726-16;G726-24;G726-32;G726-40;G.723.1-5.3;G.723.1-6.3;G729;TELEPHONE-EVENT, fmtp:"TELEPHONE-EVENT 0-15,144,149,159", p:10-30, b:4-40, e:on, t:00, s:on, v:L;M;G;D, m:sendonly;recvonly;sendrecv;inactive;confrnce;replcate;netwtest;netwloop, dq-gi But when I use nemesis, I got an ICMP error: Port unreachable (Type 3, Code 3). To build this packet, WAN source address of the capture is replaced with my server LAN address, using the mgcp-callagent port (2727) and the packet is sent to the LAN address of the box at mgcp-gateway port (2427). The command I use is nemesis udp -S 192.168.2.1 -D 192.168.2.2 -x 2727 -y 2427 -P packet_to_send. I also tried an UDP scan to the box on callagent and gateway port: PORT STATE SERVICE 2727/udp open|filtered unknown 2427/udp closed unknown I found those results a little bit strange because it should be the 2427 port opened, as it was in capture. Internet Protocol, Src: <ISP MGCP Server>, Dst: <My WAN Address> User Datagram Protocol, Src Port: mgcp-callagent (2727), Dst Port: mgcp-gateway (2427) Does someone has any idea about how having my box responding to my requests ? Thanks in advance and sorry for my english.

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  • Can I avoid a threaded UDP socket in Python dropping data?

    - by 666craig
    First off, I'm new to Python and learning on the job, so be gentle! I'm trying to write a threaded Python app for Windows that reads data from a UDP socket (thread-1), writes it to file (thread-2), and displays the live data (thread-3) to a widget (gtk.Image using a gtk.gdk.pixbuf). I'm using queues for communicating data between threads. My problem is that if I start only threads 1 and 3 (so skip the file writing for now), it seems that I lose some data after the first few samples. After this drop it looks fine. Even by letting thread 1 complete before running thread 3, this apparent drop is still there. Apologies for the length of code snippet (I've removed the thread that writes to file), but I felt removing code would just prompt questions. Hope someone can shed some light :-) import socket import threading import Queue import numpy import gtk gtk.gdk.threads_init() import gtk.glade import pygtk class readFromUDPSocket(threading.Thread): def __init__(self, socketUDP, readDataQueue, packetSize, numScans): threading.Thread.__init__(self) self.socketUDP = socketUDP self.readDataQueue = readDataQueue self.packetSize = packetSize self.numScans = numScans def run(self): for scan in range(1, self.numScans + 1): buffer = self.socketUDP.recv(self.packetSize) self.readDataQueue.put(buffer) self.socketUDP.close() print 'myServer finished!' class displayWithGTK(threading.Thread): def __init__(self, displayDataQueue, image, viewArea): threading.Thread.__init__(self) self.displayDataQueue = displayDataQueue self.image = image self.viewWidth = viewArea[0] self.viewHeight = viewArea[1] self.displayData = numpy.zeros((self.viewHeight, self.viewWidth, 3), dtype=numpy.uint16) def run(self): scan = 0 try: while True: if not scan % self.viewWidth: scan = 0 buffer = self.displayDataQueue.get(timeout=0.1) self.displayData[:, scan, 0] = numpy.fromstring(buffer, dtype=numpy.uint16) self.displayData[:, scan, 1] = numpy.fromstring(buffer, dtype=numpy.uint16) self.displayData[:, scan, 2] = numpy.fromstring(buffer, dtype=numpy.uint16) gtk.gdk.threads_enter() self.myPixbuf = gtk.gdk.pixbuf_new_from_data(self.displayData.tostring(), gtk.gdk.COLORSPACE_RGB, False, 8, self.viewWidth, self.viewHeight, self.viewWidth * 3) self.image.set_from_pixbuf(self.myPixbuf) self.image.show() gtk.gdk.threads_leave() scan += 1 except Queue.Empty: print 'myDisplay finished!' pass def quitGUI(obj): print 'Currently active threads: %s' % threading.enumerate() gtk.main_quit() if __name__ == '__main__': # Create socket (IPv4 protocol, datagram (UDP)) and bind to address socketUDP = socket.socket(socket.AF_INET, socket.SOCK_DGRAM) host = '192.168.1.5' port = 1024 socketUDP.bind((host, port)) # Data parameters samplesPerScan = 256 packetsPerSecond = 1200 packetSize = 512 duration = 1 # For now, set a fixed duration to log data numScans = int(packetsPerSecond * duration) # Create array to store data data = numpy.zeros((samplesPerScan, numScans), dtype=numpy.uint16) # Create queue for displaying from readDataQueue = Queue.Queue(numScans) # Build GUI from Glade XML file builder = gtk.Builder() builder.add_from_file('GroundVue.glade') window = builder.get_object('mainwindow') window.connect('destroy', quitGUI) view = builder.get_object('viewport') image = gtk.Image() view.add(image) viewArea = (1200, samplesPerScan) # Instantiate & start threads myServer = readFromUDPSocket(socketUDP, readDataQueue, packetSize, numScans) myDisplay = displayWithGTK(readDataQueue, image, viewArea) myServer.start() myDisplay.start() gtk.gdk.threads_enter() gtk.main() gtk.gdk.threads_leave() print 'gtk.main finished!'

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  • XmlSerializer.Deserialize blocks over NetworkStream

    - by Luca
    I'm trying to sends XML serializable objects over a network stream. I've already used this on an UDP broadcast server, where it receive UDP messages from the local network. Here a snippet of the server side: while (mServiceStopFlag == false) { if (mSocket.Available > 0) { IPEndPoint ipEndPoint = new IPEndPoint(IPAddress.Any, DiscoveryPort); byte[] bData; // Receive discovery message bData = mSocket.Receive(ref ipEndPoint); // Handle discovery message HandleDiscoveryMessage(ipEndPoint.Address, bData); ... Instead this is the client side: IPEndPoint ipEndPoint = new IPEndPoint(IPAddress.Broadcast, DiscoveryPort); MemoryStream mStream = new MemoryStream(); byte[] bData; // Create broadcast UDP server mSocket = new UdpClient(); mSocket.EnableBroadcast = true; // Create datagram data foreach (NetService s in ctx.Services) XmlHelper.SerializeClass<NetService>(mStream, s); bData = mStream.GetBuffer(); // Notify the services while (mServiceStopFlag == false) { mSocket.Send(bData, (int)mStream.Length, ipEndPoint); Thread.Sleep(DefaultServiceLatency); } It works very fine. But now i'me trying to get the same result, but on a TcpClient socket, but the using directly an XMLSerializer instance: On server side: TcpClient sSocket = k.Key; ServiceContext sContext = k.Value; Message msg = new Message(); while (sSocket.Connected == true) { if (sSocket.Available > 0) { StreamReader tr = new StreamReader(sSocket.GetStream()); msg = (Message)mXmlSerialize.Deserialize(tr); // Handle message msg = sContext.Handler(msg); // Reply with another message if (msg != null) mXmlSerialize.Serialize(sSocket.GetStream(), msg); } else Thread.Sleep(40); } And on client side: NetworkStream mSocketStream; Message rMessage; // Network stream mSocketStream = mSocket.GetStream(); // Send the message mXmlSerialize.Serialize(mSocketStream, msg); // Receive the answer rMessage = (Message)mXmlSerialize.Deserialize(mSocketStream); return (rMessage); The data is sent (Available property is greater then 0), but the method XmlSerialize.Deserialize (which should deserialize the Message class) blocks. What am I missing?

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