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  • How do I generate a connection reset programatically?

    - by Brock Adams
    Hi, I'm sure you've seen the "the connection was reset" message displayed when trying to browse web pages. (The text is from Firefox, other browsers differ.) I need to generate that message/error/condition on demand, to test workarounds. So, how do I generate that condition programmatically? (How to generate a TCP RST from PHP -- or one of the other web-app languages?) Caveats and Conditions: It cannot be a general IP block. The test client must still be able to see the test server when not triggering the condition. Ideally, it would be done at the web-application level (Python, PHP, Coldfusion, Javascript, etc.). Access to routers is problematic. Access to Apache config is a pain. Ideally, it would be triggered by fetching a specific web-page. Bonus if it works on a standard, commercial web host. Update: Sending RST is not enough to cause this condition. See my partial answer, below. I've a solution that works on a local machine, Now need to get it working on a remote host.

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  • How to establish a socket connection from iPhone to a Apache server and communicate via PHP?

    - by candoyo
    Hi, I am working on an iPhone game which is depended on a LAMP server. I want to create a "event" based system where the apache server sends an event to the iphone. For this, I am thinking of using "CFStreamCreatePairWithSocketToHost" to connect to port 80 of the apache server. I am able to successfully connect to the server and open a read and write stream via the iPhone, but I am not sure how to send data to the iphone using PHP running from the LAMP server to the iPhone. I think I can use fsockopen in php to open a socket connection and write data to that socket. I tired running this code $fp = fsockopen("tcp://localhost", 80, $errno, $errstr); if (!$fp) { echo "ERROR: $errno - $errstr<br />\n"; } else { echo"writing to socket "; fwrite($fp, "wwqeqweqw eqwe qwe \n"); //echo fread($fp, 26); fclose($fp); echo "done"; } But, I dont see anything being read on the iphone.. Any idea what's going on, or how to accomplish this? Thanks!

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  • smbclient timing out

    - by Sam Lee
    I am trying to set up a Samba share on a Centos machine. I want to connect to this server using smbclient on OS X. Here is what happens: > smbclient -L X.X.X.X timeout connecting to X.X.X.X:445 timeout connecting to X.X.X.X:139 Error connecting to X.X.X.X (Operation already in progress) Connection to X.X.X.X failed What could be going wrong? Here is my iptables dump on the Centos machine (the server): > iptables -L -n Chain INPUT (policy ACCEPT) target prot opt source destination ACCEPT all -- 0.0.0.0/0 0.0.0.0/0 REJECT all -- 0.0.0.0/0 127.0.0.0/8 reject-with icmp-port-unreachable ACCEPT all -- 0.0.0.0/0 0.0.0.0/0 state RELATED,ESTABLISHED ACCEPT tcp -- 0.0.0.0/0 0.0.0.0/0 tcp dpt:445 ACCEPT tcp -- 0.0.0.0/0 0.0.0.0/0 tcp dpt:3000 ACCEPT tcp -- 0.0.0.0/0 0.0.0.0/0 tcp dpt:80 ACCEPT tcp -- 0.0.0.0/0 0.0.0.0/0 tcp dpt:443 ACCEPT tcp -- 0.0.0.0/0 0.0.0.0/0 state NEW tcp dpt:22 ACCEPT icmp -- 0.0.0.0/0 0.0.0.0/0 icmp type 8 REJECT all -- 0.0.0.0/0 0.0.0.0/0 reject-with icmp-port-unreachable ACCEPT tcp -- 0.0.0.0/0 0.0.0.0/0 tcp dpt:3000 And finally, my smb.conf: [global] workgroup = workgroup security = SHARE load printers = No default service = global path = /home available = No encrypt passwords = yes [share] writeable = yes admin users = myusername path = /home/myhome/ force user = root valid users = myusername public = yes available = yes

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  • Using Active Directory through a Firewall

    - by Adam Brand
    I had kind of a weird setup today where I wanted to enable Windows Firewall on a Windows 2003 R2 SP2 computer that would act as an Active Directory Domain Controller. I didn't see one resource on the Internet that listed what would be required to do this, so I thought I'd list them here and see if anyone has anything to add/sees something that isn't necessary. Ports to Open with "subnet" scope: 42 | TCP | WINS (if you use it) 53 | TCP | DNS 53 | UDP | DNS 88 | TCP | Kerberos 88 | UDP | Kerberos 123 | UDP | NTP 135 | TCP | RPC 135 | UDP | RPC 137 | UDP | NetBIOS 138 | UDP | NetBIOS 139 | TCP | NetBIOS 389 | TCP | LDAP 389 | UDP | LDAP 445 | TCP | SMB 445 | UDP | SMB 636 | TCP | LDAPS 3268 | TCP | GC LDAP 3269 | TCP | GC LDAP Ports to Open with "Any" Scope (for DHCP) 67 | UDP | DHCP 2535 | UDP | DHCP ALSO You need to restrict RPC to use fixed ports instead of everything 1024. For that, you need to add two registry keys: HKEY_LOCAL_MACHINE\SYSTEM\CurrentControlSet\Services\NTDS\Parameters Registry value: TCP/IP Port Value type: REG_DWORD Value data: <-- pick a port like 1600 and put it here HKEY_LOCAL_MACHINE\SYSTEM\CurrentControlSet\Services\Netlogon\Parameters Registry value: DCTcpipPort Value type: REG_DWORD Value data: <-- pick another port like 1650 and put it here ...don't forget to add entries in the firewall to allow those in (TCP, Subnet scope). After doing all that, I was able to add a client computer to the AD domain (behind Windows Firewall) and log in successfully.

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  • Adjust iptables

    - by madunix
    cat /etc/sysconfig/iptables: # Firewall configuration written by system-config-securitylevel # Manual customization of this file is not recommended. *filter :INPUT ACCEPT [0:0] :FORWARD ACCEPT [0:0] :OUTPUT ACCEPT [0:0] :RH-Firewall-1-INPUT - [0:0] -A INPUT -j RH-Firewall-1-INPUT -A FORWARD -j RH-Firewall-1-INPUT -A RH-Firewall-1-INPUT -i lo -j ACCEPT -A RH-Firewall-1-INPUT -p icmp --icmp-type any -j ACCEPT -A RH-Firewall-1-INPUT -p 50 -j ACCEPT -A RH-Firewall-1-INPUT -p 51 -j ACCEPT -A RH-Firewall-1-INPUT -p udp --dport 5353 -d X.0.0.Y -j ACCEPT -A RH-Firewall-1-INPUT -p udp -m udp --dport 631 -j ACCEPT -A RH-Firewall-1-INPUT -m state --state ESTABLISHED,RELATED -j ACCEPT -A RH-Firewall-1-INPUT -m state --state NEW -m tcp -p tcp --dport 443 -j ACCEPT -A RH-Firewall-1-INPUT -p tcp -m tcp -s X.Y.Z.W --dport 3306 -j ACCEPT -A RH-Firewall-1-INPUT -m state --state NEW -m tcp -p tcp -s M.M.M.M --dport 3306 -j ACCEPT -A RH-Firewall-1-INPUT -m state --state NEW -m tcp -p tcp --dport 22 -j ACCEPT -A RH-Firewall-1-INPUT -m state --state NEW -m tcp -p tcp --dport 80 -j ACCEPT -A RH-Firewall-1-INPUT -m state --state NEW -m tcp -p tcp --dport 21 -j ACCEPT -A RH-Firewall-1-INPUT -j REJECT --reject-with icmp-host-prohibited COMMIT I have the above following IPtables on my linux web server(Apache/MySQL), I want to have the following: Block any traffic from multiple IP's to my web server IP1:1.2.3.4.5, IP2:6.7.8.9 ..etc Limiting one host to 20 connections to 80 port, which should not affect non-malicious user, but would render slowloris unusable from one host. Limit MYSQL port 3306 access on my server only to the following IP range A.B.C.D/255.255.255.240 Block any ICMP traffic.

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  • Feedback on Optimizing C# NET Code Block

    - by Brett Powell
    I just spent quite a few hours reading up on TCP servers and my desired protocol I was trying to implement, and finally got everything working great. I noticed the code looks like absolute bollocks (is the the correct usage? Im not a brit) and would like some feedback on optimizing it, mostly for reuse and readability. The packet formats are always int, int, int, string, string. try { BinaryReader reader = new BinaryReader(clientStream); int packetsize = reader.ReadInt32(); int requestid = reader.ReadInt32(); int serverdata = reader.ReadInt32(); Console.WriteLine("Packet Size: {0} RequestID: {1} ServerData: {2}", packetsize, requestid, serverdata); List<byte> str = new List<byte>(); byte nextByte = reader.ReadByte(); while (nextByte != 0) { str.Add(nextByte); nextByte = reader.ReadByte(); } // Password Sent to be Authenticated string string1 = Encoding.UTF8.GetString(str.ToArray()); str.Clear(); nextByte = reader.ReadByte(); while (nextByte != 0) { str.Add(nextByte); nextByte = reader.ReadByte(); } // NULL string string string2 = Encoding.UTF8.GetString(str.ToArray()); Console.WriteLine("String1: {0} String2: {1}", string1, string2); // Reply to Authentication Request MemoryStream stream = new MemoryStream(); BinaryWriter writer = new BinaryWriter(stream); writer.Write((int)(1)); // Packet Size writer.Write((int)(requestid)); // Mirror RequestID if Authenticated, -1 if Failed byte[] buffer = stream.ToArray(); clientStream.Write(buffer, 0, buffer.Length); clientStream.Flush(); } I am going to be dealing with other packet types as well that are formatted the same (int/int/int/str/str), but different values. I could probably create a packet class, but this is a bit outside my scope of knowledge for how to apply it to this scenario. If it makes any difference, this is the Protocol I am implementing. http://developer.valvesoftware.com/wiki/Source_RCON_Protocol

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  • Unable to write data to the transport connection: An existing connection was forcibly closed by the remote host

    - by xnoor
    i have an update server that sends client updates through TCP port 12000, the sending of a single file is successful only the first time, but after that i get an error message on the server "Unable to write data to the transport connection: An existing connection was forcibly closed by the remote host", if i restart the update service on the server, it works again only one, i have normal multithreaded windows service SERVER CODE namespace WSTSAU { public partial class ApplicationUpdater : ServiceBase { private Logger logger = LogManager.GetCurrentClassLogger(); private int _listeningPort; private int _ApplicationReceivingPort; private string _setupFilename; private string _startupPath; public ApplicationUpdater() { InitializeComponent(); } protected override void OnStart(string[] args) { init(); logger.Info("after init"); Thread ListnerThread = new Thread(new ThreadStart(StartListener)); ListnerThread.IsBackground = true; ListnerThread.Start(); logger.Info("after thread start"); } private void init() { _listeningPort = Convert.ToInt16(ConfigurationSettings.AppSettings["ListeningPort"]); _setupFilename = ConfigurationSettings.AppSettings["SetupFilename"]; _startupPath = System.IO.Path.GetDirectoryName(System.Reflection.Assembly.GetExecutingAssembly().GetName().CodeBase).Substring(6); } private void StartListener() { try { logger.Info("Listening Started"); ThreadPool.SetMinThreads(50, 50); TcpListener listener = new TcpListener(_listeningPort); listener.Start(); while (true) { TcpClient c = listener.AcceptTcpClient(); ThreadPool.QueueUserWorkItem(ProcessReceivedMessage, c); } } catch (Exception ex) { logger.Error(ex.Message); } } void ProcessReceivedMessage(object c) { try { TcpClient tcpClient = c as TcpClient; NetworkStream Networkstream = tcpClient.GetStream(); byte[] _data = new byte[1024]; int _bytesRead = 0; _bytesRead = Networkstream.Read(_data, 0, _data.Length); MessageContainer messageContainer = new MessageContainer(); messageContainer = SerializationManager.XmlFormatterByteArrayToObject(_data, messageContainer) as MessageContainer; switch (messageContainer.messageType) { case MessageType.ApplicationUpdateMessage: ApplicationUpdateMessage appUpdateMessage = new ApplicationUpdateMessage(); appUpdateMessage = SerializationManager.XmlFormatterByteArrayToObject(messageContainer.messageContnet, appUpdateMessage) as ApplicationUpdateMessage; Func<ApplicationUpdateMessage, bool> HandleUpdateRequestMethod = HandleUpdateRequest; IAsyncResult cookie = HandleUpdateRequestMethod.BeginInvoke(appUpdateMessage, null, null); bool WorkerThread = HandleUpdateRequestMethod.EndInvoke(cookie); break; } } catch (Exception ex) { logger.Error(ex.Message); } } private bool HandleUpdateRequest(ApplicationUpdateMessage appUpdateMessage) { try { TcpClient tcpClient = new TcpClient(); NetworkStream networkStream; FileStream fileStream = null; tcpClient.Connect(appUpdateMessage.receiverIpAddress, appUpdateMessage.receiverPortNumber); networkStream = tcpClient.GetStream(); fileStream = new FileStream(_startupPath + "\\" + _setupFilename, FileMode.Open, FileAccess.Read); FileInfo fi = new FileInfo(_startupPath + "\\" + _setupFilename); BinaryReader binFile = new BinaryReader(fileStream); FileUpdateMessage fileUpdateMessage = new FileUpdateMessage(); fileUpdateMessage.fileName = fi.Name; fileUpdateMessage.fileSize = fi.Length; MessageContainer messageContainer = new MessageContainer(); messageContainer.messageType = MessageType.FileProperties; messageContainer.messageContnet = SerializationManager.XmlFormatterObjectToByteArray(fileUpdateMessage); byte[] messageByte = SerializationManager.XmlFormatterObjectToByteArray(messageContainer); networkStream.Write(messageByte, 0, messageByte.Length); int bytesSize = 0; byte[] downBuffer = new byte[2048]; while ((bytesSize = fileStream.Read(downBuffer, 0, downBuffer.Length)) > 0) { networkStream.Write(downBuffer, 0, bytesSize); } fileStream.Close(); tcpClient.Close(); networkStream.Close(); return true; } catch (Exception ex) { logger.Info(ex.Message); return false; } finally { } } protected override void OnStop() { } } i have to note something that my windows service (server) is multithreaded.. i hope anyone can help with this

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  • C socket programming: select() is returning 0 despite messages sent from server

    - by Fantastic Fourier
    Hey all, I'm using select() to recv() messages from server, using TCP/IP. When I send() messages from the server, it returns a reasonable number of bytes, saying it's sent successful. And it does get to the client successfully when I use while loop to just recv(). Everything is fine and dandy. while(1) recv() // obviously pseudocode However, when I try to use select(), select() returns 0 from timeout (which is set to 1 second) and for the life of me I cannot figure out why it doesn't see the messages sent from the server. I should also mention that when the server disconnects, select() doesn't see that either, where as if I were to use recv(), it would return 0 to indicate that the connection using the socket has been closed. Any inputs or thoughts are deeply appreciated. #include <arpa/inet.h> #include <errno.h> #include <fcntl.h> #include <netdb.h> #include <netinet/in.h> #include <pthread.h> #include <stdio.h> #include <stdlib.h> #include <string.h> #include <strings.h> #include <sys/select.h> #include <sys/socket.h> #include <sys/time.h> #include <sys/types.h> #include <time.h> #include <unistd.h> #define SERVER_PORT 10000 #define MAX_CONNECTION 20 #define MAX_MSG 50 struct client { char c_name[MAX_MSG]; char g_name[MAX_MSG]; int csock; int host; // 0 = not host of a multicast group struct sockaddr_in client_address; struct client * next_host; struct client * next_client; }; struct fd_info { char c_name[MAX_MSG]; int socks_inuse[MAX_CONNECTION]; int sock_fd, max_fd; int exit; struct client * c_sys; struct sockaddr_in c_address[MAX_CONNECTION]; struct sockaddr_in server_address; struct sockaddr_in client_address; fd_set read_set; }; struct message { char c_name[MAX_MSG]; char g_name[MAX_MSG]; char _command[3][MAX_MSG]; char _payload[MAX_MSG]; struct sockaddr_in client_address; struct client peer; }; int main(int argc, char * argv[]) { char * host; char * temp; int i, sockfd; int msg_len, rv, ready; int connection, management, socketread; int sockfds[MAX_CONNECTION]; // for three threads that handle new connections, user inputs and select() for sockets pthread_t connection_handler, manager, socket_reader; struct sockaddr_in server_address, client_address; struct hostent * hserver, cserver; struct timeval timeout; struct message msg; struct fd_info info; info.exit = 0; // exit information: if exit = 1, threads quit info.c_sys = NULL; // looking up from the host database if (argc == 3) { host = argv[1]; // server address strncpy(info.c_name, argv[2], strlen(argv[2])); // client name } else { printf("plz read the manual, kthxbai\n"); exit(1); } printf("host is %s and hp is %p\n", host, hserver); hserver = gethostbyname(host); if (hserver) { printf("host found: %s\n", hserver->h_name ); } else { printf("host not found\n"); exit(1); } // setting up address and port structure information on serverside bzero((char * ) &server_address, sizeof(server_address)); // copy zeroes into string server_address.sin_family = AF_INET; memcpy(&server_address.sin_addr, hserver->h_addr, hserver->h_length); server_address.sin_port = htons(SERVER_PORT); bzero((char * ) &client_address, sizeof(client_address)); // copy zeroes into string client_address.sin_family = AF_INET; client_address.sin_addr.s_addr = htonl(INADDR_ANY); client_address.sin_port = htons(SERVER_PORT); // opening up socket sockfd = socket(AF_INET, SOCK_STREAM, 0); if (sockfd < 0) exit(1); else { printf("socket is opened: %i \n", sockfd); info.sock_fd = sockfd; } // sets up time out option for the bound socket timeout.tv_sec = 1; // seconds timeout.tv_usec = 0; // micro seconds ( 0.5 seconds) setsockopt(sockfd, SOL_SOCKET, SO_RCVTIMEO, &timeout, sizeof(struct timeval)); // binding socket to a port rv = bind(sockfd, (struct sockaddr *) &client_address, sizeof(client_address)); if (rv < 0) { printf("MAIN: ERROR bind() %i: %s\n", errno, strerror(errno)); exit(1); } else printf("socket is bound\n"); printf("MAIN: %li \n", client_address.sin_addr.s_addr); // connecting rv = connect(sockfd, (struct sockaddr *) &server_address, sizeof(server_address)); info.server_address = server_address; info.client_address = client_address; info.sock_fd = sockfd; info.max_fd = sockfd; printf("rv = %i\n", rv); if (rv < 0) { printf("MAIN: ERROR connect() %i: %s\n", errno, strerror(errno)); exit(1); } else printf("connected\n"); fd_set readset; FD_ZERO(&readset); FD_ZERO(&info.read_set); FD_SET(info.sock_fd, &info.read_set); while(1) { readset = info.read_set; printf("MAIN: %i \n", readset); ready = select((info.max_fd)+1, &readset, NULL, NULL, &timeout); if(ready == -1) { sleep(2); printf("TEST: MAIN: ready = -1. %s \n", strerror(errno)); } else if (ready == 0) { sleep(2); printf("TEST: MAIN: ready = 0. %s \n", strerror(errno)); } else if (ready > 0) { printf("TEST: MAIN: ready = %i. %s at socket %i \n", ready, strerror(errno), i); for(i = 0; i < ((info.max_fd)+1); i++) { if(FD_ISSET(i, &readset)) { rv = recv(sockfd, &msg, 500, 0); if(rv < 0) continue; else if(rv > 0) printf("MAIN: TEST: %s %s \n", msg._command[0], msg._payload); else if (rv == 0) { sleep(3); printf("MAIN: TEST: SOCKET CLOSEDDDDDD \n"); } FD_CLR(i, &readset); } } } info.read_set = readset; } // close connection close(sockfd); printf("socket closed. BYE! \n"); return(0); }

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  • No client internet access when setting up these iptables rules

    - by Siriss
    I have read many other posts but cannot figure this out. eth0 is my external connected to a Comcast modem. The server has internet access with no issues. eth1 is internal and running DHCP for the clients. I have DHCP working just fine, all my clients can get an IP and ping the server but they cannot access the internet. I am using ISC-DHCP-SERVER and have set /etc/default/isc-dhcp-server to INTERFACE="eht1" Here is my dhcpd.conf file located in /etc/dhcp/dhcpd.conf ddns-update-style interim; ignore client-updates; subnet 10.0.10.0 netmask 255.255.255.0 { range 10.0.10.10 10.0.10.200; option routers 10.0.10.2; option subnet-mask 255.255.255.0; option domain-name-servers 208.67.222.222, 208.67.220.220; #OpenDNS # option domain-name "example.com"; default-lease-time 21600; max-lease-time 43200; authoritative; } I have made the *net.ipv4.ip_forward=1* change in /etc/sysctl.conf here is my interfaces file: auto lo iface lo inet loopback auto eth0 iface eth0 inet dhcp iface eth1 inet static address 10.0.10.2 netmask 255.255.255.0 network 10.0.10.0 auto eth1 And finally- here is my iptables.conf file: # Firewall configuration written by system-config-firewall # Manual customization of this file is not recommended. *nat :PREROUTING ACCEPT [0:0] :OUTPUT ACCEPT [0:0] :POSTROUTING ACCEPT [0:0] -A POSTROUTING -s 10.0.10.0/24 -o eth0 -j MASQUERADE #-A PREROUTING -i eth0 -p tcp --dport 59668 -j DNAT --to-destination 10.0.10.2:59668 COMMIT *filter :INPUT ACCEPT [0:0] :FORWARD ACCEPT [0:0] :OUTPUT ACCEPT [0:0] -A INPUT -m state --state ESTABLISHED,RELATED -j ACCEPT -A INPUT -p icmp -j ACCEPT -A INPUT -i lo -j ACCEPT -A INPUT -i eth1 -j ACCEPT -A INPUT -m state --state NEW -m tcp -p tcp --dport 22 -j ACCEPT -A INPUT -m state --state NEW -m tcp -p tcp --dport 80 -j ACCEPT -A INPUT -m state --state NEW -m tcp -p tcp --dport 443 -j ACCEPT -A INPUT -m state --state NEW -m tcp -p tcp --dport 53 -j ACCEPT -A INPUT -m state --state NEW -m udp -p udp --dport 53 -j ACCEPT -A FORWARD -s 10.0.10.0/24 -o eth0 -j ACCEPT -A FORWARD -d 10.0.10.0/24 -m state --state ESTABLISHED,RELATED -i eth0 -j ACCEPT -A FORWARD -p icmp -j ACCEPT -A FORWARD -i lo -j ACCEPT -A FORWARD -i eth1 -j ACCEPT #-A FORWARD -i eth0 -m state --state NEW -m tcp -p tcp -d 10.0.10.2 --dport 59668 -j ACCEPT -A INPUT -j REJECT --reject-with icmp-host-prohibited -A FORWARD -j REJECT --reject-with icmp-host-prohibited COMMIT I am completely stuck. I cannot figure out why the clients cannot access the internet. Am I missing a service? Is a service not running? Any help would be greatly appreciated. I tried to be as thorough as possible but please let me know if I have missed something. Thank you!

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  • Why doesn't lsof show localhost TCP connections in 14.04?

    - by sfussenegger
    I have two servers running 12.04.4 and 14.04.1 respectively. Both have nginx (port 80) and a Java process (port 8080). As expected, the lsof output for the Java process on the 12.04 machine shows a couple of established connections for port 8080 (e.g. TCP 127.0.0.1:8080->127.0.0.1:58067 (ESTABLISHED)) The 14.04 machine however does not. It only shows the listening port (TCP *:8080 (LISTEN)). I'm sure there are active connections though (confirmed by access logs, Java process status output, etc). What has changed from 12.04 to result in this behavior? Can this change be the cause for the "Too many open files" errors I'm getting since moving from 12.04 to 14.04? 12.04: $ dpkg -l lsof linux-image-virtual openjdk-7-jre nginx ||/ Name Version +++-===========================================-=========================================== ii linux-image-virtual 3.2.0.59.70 ii lsof 4.81.dfsg.1-1build1 ii nginx 1.6.1-1~precise ii openjdk-7-jre 7u65-2.5.1-4ubuntu1~0.12.04.1 14.04: $ dpkg -l lsof linux-image-virtual openjdk-7-jre nginx-full ||/ Name Version Architecture Description +++-=====================================-=======================-======================= ii linux-image-virtual 3.13.0.32.38 amd64 ii lsof 4.86+dfsg-1ubuntu2 amd64 ii nginx-full 1.4.6-1ubuntu3 amd64 ii openjdk-7-jre:amd64 7u65-2.5.1-4ubuntu1~0.1 amd64

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  • How to validate selects / inserts are hitting the right server with MySQL Master/Slave

    - by bwizzy
    I've got a rails app using the master_slave_adapter plugin (http://github.com/mauricio/master_slave_adapter/tree/master) to send all selects to a slave, and all other statements to the master. Replication is setup using Mysql master / slave. I'm trying to validate that all the SQL statements are indeed going to the right place. Selects to the slave (db2), inserts to the master (db1) but I'm not sure how to do it. I've tried using tcpdump on the webservers: sudo /usr/sbin/tcpdump -q -i eth0 dst port 3306 and this is the output for a page request with a ton of selects: 10:32:36.570930 IP web2.mydomain.com.57524 > db1.mydomain.com.mysql: tcp 0 10:32:36.576805 IP web2.mydomain.com.57524 > db1.mydomain.com.mysql: tcp 0 10:32:36.577201 IP web2.mydomain.com.57524 > db1.mydomain.com.mysql: tcp 0 10:32:36.577980 IP web2.mydomain.com.57524 > db1.mydomain.com.mysql: tcp 86 10:32:36.578186 IP web2.mydomain.com.57524 > db1.mydomain.com.mysql: tcp 21 10:32:36.578359 IP web2.mydomain.com.57524 > db1.mydomain.com.mysql: tcp 27 10:32:36.578522 IP web2.mydomain.com.57524 > db1.mydomain.com.mysql: tcp 5 10:32:36.578741 IP web2.mydomain.com.57524 > db1.mydomain.com.mysql: tcp 13 10:32:36.579611 IP web2.mydomain.com.57524 > db1.mydomain.com.mysql: tcp 29 10:32:36.588201 IP web2.mydomain.com.45978 > db2.mydomain.com.mysql: tcp 0 10:32:36.588323 IP web2.mydomain.com.45978 > db2.mydomain.com.mysql: tcp 0 10:32:36.588677 IP web2.mydomain.com.45978 > db2.mydomain.com.mysql: tcp 0 10:32:36.588784 IP web2.mydomain.com.45978 > db2.mydomain.com.mysql: tcp 86 It doesn't look like all the selects are going to the slave. Maybe this isn't the right way to test, anyone know a better way?

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  • Rewriting Live TCP/IP (Layer 4) (i.e. Socket Layer) Streams

    - by user213060
    I have a simple problem which I'm sure someone here has done before... I want to rewrite Layer 4 TCP/IP streams (Not lower layer individual packets or frames.) Ettercap's etterfilter command lets you perform simple live replacements of Layer 4 TCP/IP streams based on fixed strings or regexes. Example ettercap scripting code: if (ip.proto == TCP && tcp.dst == 80) { if (search(DATA.data, "gzip")) { replace("gzip", " "); msg("whited out gzip\n"); } } if (ip.proto == TCP && tcp.dst == 80) { if (search(DATA.data, "deflate")) { replace("deflate", " "); msg("whited out deflate\n"); } } http://ettercap.sourceforge.net/forum/viewtopic.php?t=2833 I would like to rewrite streams based on my own filter program instead of just simple string replacements. Anyone have an idea of how to do this? Is there anything other than Ettercap that can do live replacement like this, maybe as a plugin to a VPN software or something? I would like to have a configuration similar to ettercap's silent bridged sniffing configuration between two Ethernet interfaces. This way I can silently filter traffic coming from either direction with no NATing problems. Note that my filter is an application that acts as a pipe filter, similar to the design of unix command-line filters: >[eth0] <----------> [my filter] <----------> [eth1]< What I am already aware of, but are not suitable: Tun/Tap - Works at the lower packet layer, I need to work with the higher layer streams. Ettercap - I can't find any way to do replacements other than the restricted capabilities in the example above. Hooking into some VPN software? - I just can't figure out which or exactly how. libnetfilter_queue - Works with lower layer packets, not TCP/IP streams. Again, the rewriting should occur at the transport layer (Layer 4) as it does in this example, instead of a lower layer packet-based approach. Exact code will help immensely! Thanks!

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  • Sending and receiving a TMemoryStream using IdTCPClient and IdTCPServer

    - by Martin Melka
    I found Remy Lebeau's chat demo of IdTCP components in XE2 and I wanted to play with it a little bit. (It can be found here) I would like to send a picture using these components and the best approach seems to be using TMemoryStream. If I send strings, the connection works fine, the strings are transmitted successfully, however when I change it to Stream instead, it doesn't work. Here is the code: Server procedure TMainForm.IdTCPServerExecute(AContext: TIdContext); var rcvdMsg: string; ms:TMemoryStream; begin // This commented code is working, it receives and sends strings. // rcvdMsg:=AContext.Connection.IOHandler.ReadLn; // LogMessage('<ServerExec> '+rcvdMsg); // // TResponseSync.SendResponse(AContext, rcvdMsg); try ms:=TMemoryStream.Create; AContext.Connection.IOHandler.ReadStream(ms); ms.SaveToFile('c:\networked.bmp'); except LogMessage('Failed to receive',clred); end; end; Client procedure TfrmMain.Button1Click(Sender: TObject); var ms: TMemoryStream; bmp: TBitmap; pic: TPicture; s: string; begin // Again, this code is working for sending strings. // s:=edMsg.Text; // Client.IOHandler.WriteLn(s); ms:=TMemoryStream.Create; pic:=TPicture.Create; pic.LoadFromFile('c:\Back.png'); bmp:=TBitmap.Create; bmp.Width:=pic.Width; bmp.Height:=pic.Height; bmp.Canvas.Draw(0,0,pic.Graphic); bmp.SaveToStream(ms); ms.Position:=0; Client.IOHandler.Write(ms); ms.Free; end; When I try to send the stream from the client, nothing observable happens (breakpoint in the OnExecute doesn't fire). However, when closing the programs(after sending the MemoryStream), two things happen: If the Client is closed first, only then does the except part get processed (the log displays the 'Failed to receive' error. However, even if I place a breakpoint on the first line of the try-except block, it somehow gets skipped and only the error is displayed). If the Server is closed first, the IDE doesn't change back from debug, Client doesn't change its state to disconnected (as it normally does when server disconnects) and after the Client is closed as well, an Access Violation error from the Server app appears. I guess this means that there is a thread of the Server still running and maintaining the connection. But no matter how much time i give it, it never completes the task of receiving the MemoryStream. Note: The server uses IdSchedulerOfThreadDefault and IdAntiFreeze, if that matters. As I can't find any reliable source of help for the revamped Indy 10 (it all appears to apply for the older Indy 10, or even Indy 9), I hope you can tell me what is wrong. Thanks - ANSWER - SERVER procedure TMainForm.IdTCPServerExecute(AContext: TIdContext); var size: integer; ms:TMemoryStream; begin try ms:=TMemoryStream.Create; size:=AContext.Connection.IOHandler.ReadLongInt; AContext.Connection.IOHandler.ReadStream(ms, size); ms.SaveToFile('c:\networked.bmp'); except LogMessage('Failed to receive',clred); end; end; CLIENT procedure TfrmMain.Button1Click(Sender: TObject); var ms: TMemoryStream; bmp: TBitmap; pic: TPicture; begin ms:=TMemoryStream.Create; pic:=TPicture.Create; pic.LoadFromFile('c:\Back.png'); bmp:=TBitmap.Create; bmp.Width:=pic.Width; bmp.Height:=pic.Height; bmp.Canvas.Draw(0,0,pic.Graphic); bmp.SaveToStream(ms); ms.Position:=0; Client.IOHandler.Write(ms, 0, True); ms.Free; end;

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  • Is it possible to connect iPhones through NAT traversal techniques over 3G for peer 2 peer gaming

    - by Dougnukem
    Is it possible to connect an iPhone to another iPhone or any other network client talking that game's protocol. I was thinking that I could use the same NAT Traversal technique used in the XBox 360 to host games on users XBox's through Firewalls. For NAT traversal to work effectively I'd have to have a central server that would allow clients to "broadcast" that they were hosting so that another client could initiate the NAT connection. In the link provided this is called relaying. Is there any inherent limitation of the US AT&T 3G network that will block the NAT traversal techniques (would you be limited to wifi firewall NAT traversal)? NAT Traversal Projects: - PJSIP - providing NAT traversal functionalities by using standard based protocols such as STUN, TURN, and ICE. - Freecast using Apache MINA Java network library for NAT Traversal

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  • How much buffer does NetworkStream and TcpClient have?

    - by Earlz
    Hello, We are writing a TCPServer and Client program. How much space is there in the TcpClient buffer? Like, at what point will it begin to throw away data? We are trying to determine if the TcpClient can be blocking or if it should go into it's own background thread(so that the buffer can not get full)..

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  • Spikes in Socket Performance

    - by Harun Prasad
    We are facing random spikes in high throughput transaction processing system using sockets for IPC. Below is the setup used for the run: The client opens and closes new connection for every transaction, and there are 4 exchanges between the server and the client. We have disabled the TIME_WAIT, by setting the socket linger (SO_LINGER) option via getsockopt as we thought that the spikes were caused due to the sockets waiting in TIME_WAIT. There is no processing done for the transaction. Only messages are passed. OS used Centos 5.4 The average round trip time is around 3 milli seconds, but some times the round trip time ranges from 100 milli seconds to couple of seconds. Steps used for Execution and Measurement and output Starting the server $ python sockServerLinger.py /dev/null & Starting the client to post 1 million transactions to the server. And logs the time for a transaction in the client.log file. $ python sockClient.py 1000000 client.log Once the execution finishes the following command will show the execution time greater than 100 milliseconds in the format <line_number>:<execution_time>. $ grep -n "0.[1-9]" client.log | less Below is the example code for Server and Client. Server # File: sockServerLinger.py import socket, traceback,time import struct host = '' port = 9999 l_onoff = 1 l_linger = 0 lingeropt = struct.pack('ii', l_onoff, l_linger) s = socket.socket(socket.AF_INET, socket.SOCK_STREAM) s.setsockopt(socket.SOL_SOCKET, socket.SO_REUSEADDR, 1) s.setsockopt(socket.SOL_SOCKET, socket.SO_LINGER, lingeropt) s.bind((host, port)) s.listen(1) while 1: try: clientsock, clientaddr = s.accept() print "Got connection from", clientsock.getpeername() data = clientsock.recv(1024*1024*10) #print "asdasd",data numsent=clientsock.send(data) data1 = clientsock.recv(1024*1024*10) numsent=clientsock.send(data) ret = 1 while(ret>0): data1 = clientsock.recv(1024*1024*10) ret = len(data) clientsock.close() except KeyboardInterrupt: raise except: print traceback.print_exc() continue Client # File: sockClient.py import socket, traceback,sys import time i = 0 while 1: try: st = time.time() s = socket.socket(socket.AF_INET,socket.SOCK_STREAM) while (s.connect_ex(('127.0.0.1',9999)) != 0): continue numsent=s.send("asd"*1000) response = s.recv(6000) numsent=s.send("asd"*1000) response = s.recv(6000) i+=1 if i == int(sys.argv[1]): break except KeyboardInterrupt: raise except: print "in exec:::::::::::::",traceback.print_exc() continue print time.time() -st

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  • [python] voice communication for python help!

    - by Eric
    Hello! I'm currently trying to write a voicechat program in python. All tips/trick is welcome to do this. So far I found pyAudio to be a wrapper of PortAudio. So I played around with that and got an input stream from my microphone to be played back to my speakers. Only RAW of course. But I can't send RAW-data over the netowrk (due the size duh), so I'm looking for a way to encode it. And I searched around the 'net and stumbled over this speex-wrapper for python. It seems to good to be true, and believe me, it was. You see in pyAudio you can set the size of the chunks you want to take from your input audiobuffer, and in that sample code on the link, it's set to 320. Then when it's encoded, its like ~40 bytes of data per chunk, which is fairly acceptable I guess. And now for the problem. I start a sample program which just takes the input stream, encodes the chunks, decodes them and play them (not sending over the network due testing). If I just let my computer idle and run this program it works great, but as soon as I do something, i.e start Firefox or something, the audio input buffer gets all clogged up! It just grows and then it all crashes and gives me an overflow error on the buffer.. OK, so why am I just taking 320 bytes of the stream? I could just take like 1024 bytes or something and that will easy the pressure on the buffer. BUT. If I give speex 1024 bytes of data to encode/decode, it either crashes and says that thats too big for its buffer. OR it encodes/decodes it, but the sound is very noisy and "choppy" as if it only encoded a tiny bit of that 1024 chunk and the rest is static noise. So the sound sounds like a helicopter, lol. I did some research and it seems that speex only can convert 320 bytes of data at time, and well, 640 for wide-band. But that's the standard? How can I fix this problem? How should I construct my program to work with speex? I could use a middle-buffer tho that takes all available data to read from the buffer, then chunk this up in 320 bits and encode/decode them. But this takes a bit longer time and seems like a very bad solution of the problem.. Because as far as I know, there's no other encoder for python that encodes the audio so it can be sent over the network in acceptable small packages, or? I've been googling for three days now. Also there is this pyMedia library, I don't know if its good to convert to mp3/ogg for this kind of software. Thank in in advance for reading this, hope anyone can help me! (:

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  • Can't access a local site site on LAN

    - by Dilawar
    I have lighttpd setup on a machine (say ip is 10.107.105.13) with following details. inet addr : 10.107.105.13 Bcast : 10.107.111.255 Mask : 255.255.240.0 I can access my site on this computer by using firefox http://localhost/index.html. Now I am trying to access this site from another computer with following details inet addr : 10.14.42.7 Bcast : 10.14.42.255 Mask : 255.255.255.0 But it says 'access denied'. nmap 10.107.105.13 gives the following output. PORT STATE SERVICE 22/tcp open ssh 80/tcp open http 1234/tcp open hotline 3306/tcp open mysql 9418/tcp open git Following is the output of iptables -L -n -v on 10.107.105.13 141 11207 ACCEPT tcp -- * * 0.0.0.0/0 0.0.0.0/0 tcp dpt:80 0 0 ACCEPT tcp -- * * 0.0.0.0/0 0.0.0.0/0 tcp dpt:80 FORWARD and OUTPUT section empty. What is wrong with all this?

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  • LDAP over SSL with an EFI Fiery printer

    - by austinian
    I've got a printer with a Fiery running 8e Release 2. I can authenticate users against AD using the LDAP configuration, but I can only get it to work if I don't use SSL/TLS, and only if I use SIMPLE authentication. Right now, it's authenticating using a fairly low-impact user, but it's also the only system on our network that's not using LDAPS. I can get AD info fine over LDAPS using ldp.exe from my machine, our firewall, our mail filter, our linux boxes, etc. The only problem child is the Fiery. I've added the LDAP server certificate as a trusted cert to the Fiery, but after I check the box for Secure Communication and change the port to 636, pressing Validate results in a dialog box coming up saying: LDAP Validation Failed Server Name invalid or server is unavailable. I've tried changing the server name to use just the name, the FQDN, and the IP address, and changed it to another server, just to see if it was just this AD server that was fussy with the Fiery. EDIT: removed LDP output, added packet capture analysis from wireshark: The conversation seems pretty normal to me, up to the point where the Fiery terminates the connection after the server sends back a handshake response. Maybe they messed up their TLS implementation? I'm trying support, but it's been fairly useless so far. The cert is a SHA-2 (sha256RSA) 2048-bit certificate. Also, it looks like the Fiery is specifying TLS 1.0. Looking at http://msdn.microsoft.com/en-us/library/windows/desktop/aa374757(v=vs.85).aspx, I'm not seeing SHA256 and TLS 1.0 combination being supported by SChannel. headdesk perhaps that's why, after the DC changes the cipher spec, the connection is terminated by the Fiery? TLS 1.1 and 1.2 are enabled on the DC. Wireshark conversation: DC: 172.17.2.22, Fiery: 172.17.2.42 No. Time Source Source Port Destination Destination Port Protocol Length Info 1 0.000000000 172.17.2.42 48633 172.17.2.22 ldaps TCP 74 48633 > ldaps [SYN] Seq=0 Win=5840 Len=0 MSS=1460 SACK_PERM=1 TSval=3101761 TSecr=0 WS=4 2 0.000182000 Dell_5e:94:e3 Broadcast ARP 60 Who has 172.17.2.42? Tell 172.17.2.22 3 0.000369000 TyanComp_c9:0f:90 Dell_5e:94:e3 ARP 60 172.17.2.42 is at 00:e0:81:c9:0f:90 4 0.000370000 172.17.2.22 ldaps 172.17.2.42 48633 TCP 74 ldaps > 48633 [SYN, ACK] Seq=0 Ack=1 Win=8192 Len=0 MSS=1460 WS=256 SACK_PERM=1 TSval=67970573 TSecr=3101761 5 0.000548000 172.17.2.42 48633 172.17.2.22 ldaps TCP 66 48633 > ldaps [ACK] Seq=1 Ack=1 Win=5840 Len=0 TSval=3101761 TSecr=67970573 6 0.001000000 172.17.2.42 48633 172.17.2.22 ldaps TLSv1 147 Client Hello 7 0.001326000 172.17.2.22 ldaps 172.17.2.42 48633 TCP 1514 [TCP segment of a reassembled PDU] 8 0.001513000 172.17.2.22 ldaps 172.17.2.42 48633 TCP 1514 [TCP segment of a reassembled PDU] 9 0.001515000 172.17.2.42 48633 172.17.2.22 ldaps TCP 66 48633 > ldaps [ACK] Seq=82 Ack=1449 Win=8736 Len=0 TSval=3101761 TSecr=67970573 10 0.001516000 172.17.2.42 48633 172.17.2.22 ldaps TCP 66 48633 > ldaps [ACK] Seq=82 Ack=2897 Win=11632 Len=0 TSval=3101761 TSecr=67970573 11 0.001732000 172.17.2.22 ldaps 172.17.2.42 48633 TCP 1514 [TCP segment of a reassembled PDU] 12 0.001737000 172.17.2.22 ldaps 172.17.2.42 48633 TLSv1 1243 Server Hello, Certificate, Certificate Request, Server Hello Done 13 0.001738000 172.17.2.42 48633 172.17.2.22 ldaps TCP 66 48633 > ldaps [ACK] Seq=82 Ack=4345 Win=14528 Len=0 TSval=3101761 TSecr=67970573 14 0.001739000 172.17.2.42 48633 172.17.2.22 ldaps TCP 66 48633 > ldaps [ACK] Seq=82 Ack=5522 Win=17424 Len=0 TSval=3101761 TSecr=67970573 15 0.002906000 172.17.2.42 48633 172.17.2.22 ldaps TLSv1 78 Certificate 16 0.004155000 172.17.2.42 48633 172.17.2.22 ldaps TLSv1 333 Client Key Exchange 17 0.004338000 172.17.2.22 ldaps 172.17.2.42 48633 TCP 66 ldaps > 48633 [ACK] Seq=5522 Ack=361 Win=66304 Len=0 TSval=67970573 TSecr=3101762 18 0.004338000 172.17.2.42 48633 172.17.2.22 ldaps TLSv1 72 Change Cipher Spec 19 0.005481000 172.17.2.42 48633 172.17.2.22 ldaps TLSv1 327 Encrypted Handshake Message 20 0.005645000 172.17.2.22 ldaps 172.17.2.42 48633 TCP 66 ldaps > 48633 [ACK] Seq=5522 Ack=628 Win=66048 Len=0 TSval=67970574 TSecr=3101762 21 0.010247000 172.17.2.22 ldaps 172.17.2.42 48633 TLSv1 125 Change Cipher Spec, Encrypted Handshake Message 22 0.016451000 172.17.2.42 48633 172.17.2.22 ldaps TCP 66 48633 > ldaps [FIN, ACK] Seq=628 Ack=5581 Win=17424 Len=0 TSval=3101765 TSecr=67970574 23 0.016630000 172.17.2.22 ldaps 172.17.2.42 48633 TCP 66 ldaps > 48633 [ACK] Seq=5581 Ack=629 Win=66048 Len=0 TSval=67970575 TSecr=3101765 24 0.016811000 172.17.2.22 ldaps 172.17.2.42 48633 TCP 60 ldaps > 48633 [RST, ACK] Seq=5581 Ack=629 Win=0 Len=0

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  • What is SocketOptionName.ReuseAddress used for?

    - by Hemant
    I used to think that using SocketOptionName.ReuseAddress, I can reuse a port that is in TIME_WAIT state. But I tried to experiment with it and it seems it has no effect. If I check sockets using netstat, and it shows the socket is in TIME_WAIT state and I immediately run the client again, I get the exception: Only one usage of each socket address (protocol/network address/port) is normally permitted 172.16.16.16:12345 I cannot make anything out of it. Please can you elaborate what SocketOptionName.ReuseAddress is good for?

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  • [Perl] Testing for EAGAIN / EWOULDBLOCK on a recv

    - by Robert S. Barnes
    I'm testing a socket to see if it's still open: my $dummy = ''; my $ret = recv($sock, $dummy, 1, MSG_DONTWAIT | MSG_PEEK); if (!defined $ret || (length($dummy) == 0 && $! != EAGAIN && $! != EWOULDBLOCK )) { logerr("Broken pipe? ".__LINE__." $!"); } else { # socket still connected, reuse logerr(__LINE__.": $!"); return $sock; } I'm passing this code a socket I know for certain is open and it's always going through the first branch and logging "Broken pipe? 149 Resource temporarily unavailable". I don't understand how this is happening since "Resource temporarily unavailable" is supposed to correspond to EAGAIN as far as I know. I'm sure there must be something simple I'm missing. And yes, I know this is not a full proof way to test and I account for that.

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  • How do the routers communicate with each other ?

    - by Berkay
    Let's say that i want make a request a to a web page which is hosted in Europe (i live in USA).My packets only consist the IP address of the web page, first the domain name to ip address transformation is done, then my packets start their journey through to europe. i assume that MAC addresses never used in this situation? are they? First, my packets deal with many routers on way how these routers communicate with each other?, are router addresses added to my packet headers ? Second, is there a specific path router to router comminication or which conditions affect this route? Third to cross the Atlantic Ocean, are cables used or... ?

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  • silverlight Socket: Unhandled Error in Silverlight Application An attempt was made to access a sock

    - by Yang
    I basically try to reproduce the Socket example from here: http://www.silverlightshow.net/items/Sockets-and-their-implementation-in-SL2-Beta-1-including-a-chat-like-example.aspx I only made a small change in the client side, i.e., String safeHost = "127.0.0.1"; int port = 4509; Then I got this permission error? Any idea why? Unhandled Error in Silverlight Application An attempt was made to access a socket in a way forbidden by its access permissions.

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  • Cocoa NSOutputStream send to a connection

    - by Chuck
    Hi, I am new to Cocoa, but managed to get a connection (to a FTP) up and running, and I've set up an eventhandler for the NSInputStream iStream to alert every response (which also works). What I manage to get is simply the hello message and a connection timeout 60 sec, closing control connection. After searching stackoverflow and finding a lot of NSOutputStream write problems (e.g. http://stackoverflow.com/questions/703729/how-to-use-nsoutputstreams-write-message) and a lot of confusion in my google hits, I figured I'd try to ask my own question: I've tried reading the developer.apple.com doc on OutputStream, but it seems almost impossible for me to send some data (in this case just a string) to the "connection" via the NSOutputStream oStream. - (IBAction) send_something: sender { const char *send_command_char = [@"USER foo" UTF8String]; send_command_buffer = [NSMutableData dataWithBytes:send_command_char length:strlen(send_command_char) + 1]; uint8_t *readBytes = (uint8_t *)[send_command_buffer mutableBytes]; NSInteger byteIndex = 0; readBytes += byteIndex; int data_len = [send_command_buffer length]; unsigned int len = ((data_len - byteIndex >= 1024) ? 1024 : (data_len-byteIndex)); uint8_t buf[len]; (void)memcpy(buf, readBytes, len); len = [oStream write:(const uint8_t *)buf maxLength:len]; byteIndex += len; } the above seems not to result in any useable events. typing it under NSStreamEventHasSpaceAvailable sometimes give a response if I spam the ftp by keep creating new connection instances and keep sending some command whenever oStream has free space. In other words, nothing "right" and so I'm still unclear how to properly send a command to the connection. Should I open - write - close every time i want to write to oStream (and thus to the ftp) and can I then expect a reply (hasBytesAvailable event on iStream)? For some reason I find it very difficult to find any clear tutorials on this matter. Seems like there are more than a few in the same position as me: unclear how to use oStream write? Any little bit that can help clear this up is greatly appreciated! If needed I can write the rest of the code. Chuck

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