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  • How can private IPV4 addresses get past iptables NAT (tcp RST,FIN)

    - by gscott
    I've got a router performing simple NAT translation using iptables iptables -t nat -o -j MASQUERADE This works fine almost all of the time except for one particular case where some TCP RST and FIN packets are leaving the router un-NAT'd. In this scenario I setup 1 or 2 client computers streaming Flash video (eg www.nasa.gov/ntv) At the router I then tear down and re-establish the public interface (which is a modem) As expected the Flash streams stall out. After the connection is re-established and I try to refresh the Flash pages, I see some TCP RST and [FIN,ACK] packets leaving the public interface (I assume as Flash attempts to recover its stream). I don't know how these packets can leave the router non-NAT'd

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  • terminal tools and logs for debugging TCP issues

    - by kellogs
    I have a server which I am testing for functionality (not load, not stress) with tsung. 50 users / second, 100 total users. Judging from tsung (tsung is the testing framework) graphs, there TCP connections (red line) drops to 0 while the commenced user sessions (green line) does not. Server logs show nothing to be gripping onto, so I am speculating some kind of TCP issue. Should this be the case ? Where would I look further on the server, any logs / tools to be looking at ? Only SSH available, no GUI. > root@XMPP:~# cat /etc/lsb-release > DISTRIB_ID=Ubuntu > DISTRIB_RELEASE=11.10 > DISTRIB_CODENAME=oneiric > DISTRIB_DESCRIPTION="Ubuntu 11.10" Thank you

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  • TCP connection stuck in SYN_RECV state despite ACK received, Linux 2.6.18, embedded, ARM

    - by waynix
    My client cannot connect to my protocol port (TCP) after some network glitches, even though all other protocols (telnet/HTTP/FTP) work fine. netstat shows that my server is listening and tcpdump on the server shows all 3 packets are exchanged: 18:29:16.578964 IP 10.9.59.10.3355 10.9.43.131.5084: S 2602965897:2602965897(0) win 65535 <mss 1460,nop,nop,sackOK> 18:29:16.579107 IP 10.9.43.131.5084 10.9.59.10.3355: S 3464857909:3464857909(0) ack 2602965898 win 5840 <mss 1460,nop,nop,sackOK> 18:29:16.579284 IP 10.9.59.10.3355 10.9.43.131.5084: . ack 1 win 65535 But somehow netstat -t shows the connection still in SYN_RECV, as if the ack is not seen by the TCP state machine. I have to restart my server to get it to work. syncookie is not enabled, and I know from client code behavior and tcpdump that there is no SYN flooding. Help much appreciated.

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  • Windows Server 2008 constantly spamming external IP's on outbound TCP port 445

    - by RSXAdmin
    Hi Server Fault, I have a Windows Server 2008 box running as a Domain Controller. I have noticed in my Cisco ASA firewall logs that this box is continuously sending out (like a thousand requests a second) requests on TCP port 445 to external hosts. I have made an effort to deny this outbound traffic from getting on the internet (using the ASA), however I would like these requests to stop from even occurring at all. I have tried disabling TCP/IP over NetBIOS. I have even turned on Windows Advanced Firewall on the box itself to block outbound 445 but the ASA still detects this particular traffic hitting it. I have other DC's and similar type boxes which are not behaving the same way as this box. Is this normal? Is there a way to stop this spamming? Have I been infected? Thank you universe.

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  • TCP 30 small packets per second flood connection with server

    - by Denis Ermolin
    I'm testing connection with flash client and cloud server(boost::asio for software) over TCP connection. My connection with server already is really poor - 120 ms ping in average. I found when i start to send packets with 2 bytes size (without tcp header) with speed 30 packets/s - ping grow to 170-200 average. I think that it's really bad and my bad connection and bad cloud provider is reason for this high ping without any load. What do you think? (I tested my software - it can compute about 50k small packets/s so software is not a problem). I measure my ping through flash client - send packet with timestamp and immediatly send from server to client.

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  • Tcp window size won't go above 130048

    - by Roger
    I have 2 servers set up with about 80ms latency between them. Both are centos 6 and run a java app that transfers data from on location to another. Both are on 1gbps connections. I have been trying different sysctl settings and different send & receive buffer settings in java but no matter what I set them to, I cannot get the tcp window size to go above 130048 in the tcp dumps. This equates to roughly 13mbps which is the actual throughput I am getting.

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  • Poor TCP loopback throughput on Windows

    - by Yodan Tauber
    I measured the throughput of a locally bound TCP socket connection on my computer (Intel Q9550, 64 GB RAM, Windows XP 64 bit) using iperf. I got dissatisfying results (around 1.6 Gbit/s) each time, no matter how I tweaked the TCP settings (buffer length, window size, max segment size, no delay). I got similar results when I tried netperf. Now, I understand (from sources like these) that the average throughput of a loopback connection should be around 5 Gbit/s. What could be the reasons for such poor performance?

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  • DTracing TCP congestion control

    - by user12820842
    In a previous post, I showed how we can use DTrace to probe TCP receive and send window events. TCP receive and send windows are in effect both about flow-controlling how much data can be received - the receive window reflects how much data the local TCP is prepared to receive, while the send window simply reflects the size of the receive window of the peer TCP. Both then represent flow control as imposed by the receiver. However, consider that without the sender imposing flow control, and a slow link to a peer, TCP will simply fill up it's window with sent segments. Dealing with multiple TCP implementations filling their peer TCP's receive windows in this manner, busy intermediate routers may drop some of these segments, leading to timeout and retransmission, which may again lead to drops. This is termed congestion, and TCP has multiple congestion control strategies. We can see that in this example, we need to have some way of adjusting how much data we send depending on how quickly we receive acknowledgement - if we get ACKs quickly, we can safely send more segments, but if acknowledgements come slowly, we should proceed with more caution. More generally, we need to implement flow control on the send side also. Slow Start and Congestion Avoidance From RFC2581, let's examine the relevant variables: "The congestion window (cwnd) is a sender-side limit on the amount of data the sender can transmit into the network before receiving an acknowledgment (ACK). Another state variable, the slow start threshold (ssthresh), is used to determine whether the slow start or congestion avoidance algorithm is used to control data transmission" Slow start is used to probe the network's ability to handle transmission bursts both when a connection is first created and when retransmission timers fire. The latter case is important, as the fact that we have effectively lost TCP data acts as a motivator for re-probing how much data the network can handle from the sending TCP. The congestion window (cwnd) is initialized to a relatively small value, generally a low multiple of the sending maximum segment size. When slow start kicks in, we will only send that number of bytes before waiting for acknowledgement. When acknowledgements are received, the congestion window is increased in size until cwnd reaches the slow start threshold ssthresh value. For most congestion control algorithms the window increases exponentially under slow start, assuming we receive acknowledgements. We send 1 segment, receive an ACK, increase the cwnd by 1 MSS to 2*MSS, send 2 segments, receive 2 ACKs, increase the cwnd by 2*MSS to 4*MSS, send 4 segments etc. When the congestion window exceeds the slow start threshold, congestion avoidance is used instead of slow start. During congestion avoidance, the congestion window is generally updated by one MSS for each round-trip-time as opposed to each ACK, and so cwnd growth is linear instead of exponential (we may receive multiple ACKs within a single RTT). This continues until congestion is detected. If a retransmit timer fires, congestion is assumed and the ssthresh value is reset. It is reset to a fraction of the number of bytes outstanding (unacknowledged) in the network. At the same time the congestion window is reset to a single max segment size. Thus, we initiate slow start until we start receiving acknowledgements again, at which point we can eventually flip over to congestion avoidance when cwnd ssthresh. Congestion control algorithms differ most in how they handle the other indication of congestion - duplicate ACKs. A duplicate ACK is a strong indication that data has been lost, since they often come from a receiver explicitly asking for a retransmission. In some cases, a duplicate ACK may be generated at the receiver as a result of packets arriving out-of-order, so it is sensible to wait for multiple duplicate ACKs before assuming packet loss rather than out-of-order delivery. This is termed fast retransmit (i.e. retransmit without waiting for the retransmission timer to expire). Note that on Oracle Solaris 11, the congestion control method used can be customized. See here for more details. In general, 3 or more duplicate ACKs indicate packet loss and should trigger fast retransmit . It's best not to revert to slow start in this case, as the fact that the receiver knew it was missing data suggests it has received data with a higher sequence number, so we know traffic is still flowing. Falling back to slow start would be excessive therefore, so fast recovery is used instead. Observing slow start and congestion avoidance The following script counts TCP segments sent when under slow start (cwnd ssthresh). #!/usr/sbin/dtrace -s #pragma D option quiet tcp:::connect-request / start[args[1]-cs_cid] == 0/ { start[args[1]-cs_cid] = 1; } tcp:::send / start[args[1]-cs_cid] == 1 && args[3]-tcps_cwnd tcps_cwnd_ssthresh / { @c["Slow start", args[2]-ip_daddr, args[4]-tcp_dport] = count(); } tcp:::send / start[args[1]-cs_cid] == 1 && args[3]-tcps_cwnd args[3]-tcps_cwnd_ssthresh / { @c["Congestion avoidance", args[2]-ip_daddr, args[4]-tcp_dport] = count(); } As we can see the script only works on connections initiated since it is started (using the start[] associative array with the connection ID as index to set whether it's a new connection (start[cid] = 1). From there we simply differentiate send events where cwnd ssthresh (congestion avoidance). Here's the output taken when I accessed a YouTube video (where rport is 80) and from an FTP session where I put a large file onto a remote system. # dtrace -s tcp_slow_start.d ^C ALGORITHM RADDR RPORT #SEG Slow start 10.153.125.222 20 6 Slow start 138.3.237.7 80 14 Slow start 10.153.125.222 21 18 Congestion avoidance 10.153.125.222 20 1164 We see that in the case of the YouTube video, slow start was exclusively used. Most of the segments we sent in that case were likely ACKs. Compare this case - where 14 segments were sent using slow start - to the FTP case, where only 6 segments were sent before we switched to congestion avoidance for 1164 segments. In the case of the FTP session, the FTP data on port 20 was predominantly sent with congestion avoidance in operation, while the FTP session relied exclusively on slow start. For the default congestion control algorithm - "newreno" - on Solaris 11, slow start will increase the cwnd by 1 MSS for every acknowledgement received, and by 1 MSS for each RTT in congestion avoidance mode. Different pluggable congestion control algorithms operate slightly differently. For example "highspeed" will update the slow start cwnd by the number of bytes ACKed rather than the MSS. And to finish, here's a neat oneliner to visually display the distribution of congestion window values for all TCP connections to a given remote port using a quantization. In this example, only port 80 is in use and we see the majority of cwnd values for that port are in the 4096-8191 range. # dtrace -n 'tcp:::send { @q[args[4]-tcp_dport] = quantize(args[3]-tcps_cwnd); }' dtrace: description 'tcp:::send ' matched 10 probes ^C 80 value ------------- Distribution ------------- count -1 | 0 0 |@@@@@@ 5 1 | 0 2 | 0 4 | 0 8 | 0 16 | 0 32 | 0 64 | 0 128 | 0 256 | 0 512 | 0 1024 | 0 2048 |@@@@@@@@@ 8 4096 |@@@@@@@@@@@@@@@@@@@@@@@@@@ 23 8192 | 0

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  • Sony PSM sdk and 2d Game engine

    - by Notbad
    I have started with Sony PSM sdk this week. I'm interested to create a little 2D game and have been reading through the web about a so called "2D game engine" integrated in psm. Some information I read suggested that it was going to be added on january 2012, but I have been going through the documentation and haven't been able to find any reference to it. Does anybody know if they finally introduced the 2D game engien for psm? Thanks in advance.

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  • Game engine development in C++ [closed]

    - by Chris Cochran
    I am arriving at completion on a multithreaded concurrency framework designed for high-performance computing. Though I am not a gamer, it has occurred to me that this stand-alone software core could be an ideal basis for a multiprocessor game engine (64-bit native C++, 5000+ entry points). Are there any websites I could visit to discuss this technology with programmers and developers who could really benefit from it?

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  • I need a very rapid game engine with as many presets as possible [closed]

    - by GLycan
    I have to make a game about the immune system (with as many grotesque simplifications as I want) in roughly two weeks. I'm thinking of something along the lines of bubble tanks, but I think I can roll with pretty much anything that fits the following: Working engine that handles mouvement, objects The fighting system can be done in any way that allows a bare minimum of configuration Scripting should be, preferably, Python Art should be customizable I'm hoping that there is some existing game that I could adapt. Any sggestions (aside from not leaving projects until later) ?

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  • Maintaining State in Mud Engine

    - by Johnathon Sullinger
    I am currently working on a Mud Engine and have started implementing my state engine. One of the things that has me troubled is maintaining different states at once. For instance, lets say that the user has started a tutorial, which requires specific input. If the user types "help" I want to switch in to a help state, so they can get the help they need, then return them to the original state once exiting the help. my state system uses a State Manager to manage the state per user: public class StateManager { /// <summary> /// Gets the current state. /// </summary> public IState CurrentState { get; private set; } /// <summary> /// Gets the states available for use. /// </summary> /// <value> public List<IState> States { get; private set; } /// <summary> /// Gets the commands available. /// </summary> public List<ICommand> Commands { get; private set; } /// <summary> /// Gets the mob that this manager controls the state of. /// </summary> public IMob Mob { get; private set; } public void Initialize(IMob mob, IState initialState = null) { this.Mob = mob; if (initialState != null) { this.SwitchState(initialState); } } /// <summary> /// Performs the command. /// </summary> /// <param name="message">The message.</param> public void PerformCommand(IMessage message) { if (this.CurrentState != null) { ICommand command = this.CurrentState.GetCommand(message); if (command is NoOpCommand) { // NoOperation commands indicate that the current state is not finished yet. this.CurrentState.Render(this.Mob); } else if (command != null) { command.Execute(this.Mob); } else if (command == null) { new InvalidCommand().Execute(this.Mob); } } } /// <summary> /// Switches the state. /// </summary> /// <param name="state">The state.</param> public void SwitchState(IState state) { if (this.CurrentState != null) { this.CurrentState.Cleanup(); } this.CurrentState = state; if (state != null) { this.CurrentState.Render(this.Mob); } } } Each of the different states that the user can be in, is a Type implementing IState. public interface IState { /// <summary> /// Renders the current state to the players terminal. /// </summary> /// <param name="player">The player to render to</param> void Render(IMob mob); /// <summary> /// Gets the Command that the player entered and preps it for execution. /// </summary> /// <returns></returns> ICommand GetCommand(IMessage command); /// <summary> /// Cleanups this instance during a state change. /// </summary> void Cleanup(); } Example state: public class ConnectState : IState { /// <summary> /// The connected player /// </summary> private IMob connectedPlayer; public void Render(IMob mob) { if (!(mob is IPlayer)) { throw new NullReferenceException("ConnectState can only be used with a player object implementing IPlayer"); } //Store a reference for the GetCommand() method to use. this.connectedPlayer = mob as IPlayer; var server = mob.Game as IServer; var game = mob.Game as IGame; // It is not guaranteed that mob.Game will implement IServer. We are only guaranteed that it will implement IGame. if (server == null) { throw new NullReferenceException("LoginState can only be set to a player object that is part of a server."); } //Output the game information mob.Send(new InformationalMessage(game.Name)); mob.Send(new InformationalMessage(game.Description)); mob.Send(new InformationalMessage(string.Empty)); //blank line //Output the server MOTD information mob.Send(new InformationalMessage(string.Join("\n", server.MessageOfTheDay))); mob.Send(new InformationalMessage(string.Empty)); //blank line mob.StateManager.SwitchState(new LoginState()); } /// <summary> /// Gets the command. /// </summary> /// <param name="message">The message.</param> /// <returns>Returns no operation required.</returns> public Commands.ICommand GetCommand(IMessage message) { return new NoOpCommand(); } /// <summary> /// Cleanups this instance during a state change. /// </summary> public void Cleanup() { // We have nothing to clean up. return; } } With the way that I have my FSM set up at the moment, the user can only ever have one state at a time. I read a few different posts on here about state management but nothing regarding keeping a stack history. I thought about using a Stack collection, and just pushing new states on to the stack then popping them off as the user moves out from one. It seems like it would work, but I'm not sure if it is the best approach to take. I'm looking for recommendations on this. I'm currently swapping state from within the individual states themselves as well which I'm on the fence about if it makes sense to do there or not. The user enters a command, the StateManager passes the command to the current State and lets it determine if it needs it (like passing in a password after entering a user name), if the state doesn't need any further commands, it returns null. If it does need to continue doing work, it returns a No Operation to let the state manager know that the state still requires further input from the user. If null is returned, the state manager will then go find the appropriate state for the command entered by the user. Example state requiring additional input from the user public class LoginState : IState { /// <summary> /// The connected player /// </summary> private IPlayer connectedPlayer; private enum CurrentState { FetchUserName, FetchPassword, InvalidUser, } private CurrentState currentState; /// <summary> /// Renders the current state to the players terminal. /// </summary> /// <param name="mob"></param> /// <exception cref="System.NullReferenceException"> /// ConnectState can only be used with a player object implementing IPlayer /// or /// LoginState can only be set to a player object that is part of a server. /// </exception> public void Render(IMob mob) { if (!(mob is IPlayer)) { throw new NullReferenceException("ConnectState can only be used with a player object implementing IPlayer"); } //Store a reference for the GetCommand() method to use. this.connectedPlayer = mob as IPlayer; var server = mob.Game as IServer; // Register to receive new input from the user. mob.ReceivedMessage += connectedPlayer_ReceivedMessage; if (server == null) { throw new NullReferenceException("LoginState can only be set to a player object that is part of a server."); } this.currentState = CurrentState.FetchUserName; switch (this.currentState) { case CurrentState.FetchUserName: mob.Send(new InputMessage("Please enter your user name")); break; case CurrentState.FetchPassword: mob.Send(new InputMessage("Please enter your password")); break; case CurrentState.InvalidUser: mob.Send(new InformationalMessage("Invalid username/password specified.")); this.currentState = CurrentState.FetchUserName; mob.Send(new InputMessage("Please enter your user name")); break; } } /// <summary> /// Receives the players input. /// </summary> /// <param name="sender">The sender.</param> /// <param name="e">The e.</param> void connectedPlayer_ReceivedMessage(object sender, IMessage e) { // Be good memory citizens and clean ourself up after receiving a message. // Not doing this results in duplicate events being registered and memory leaks. this.connectedPlayer.ReceivedMessage -= connectedPlayer_ReceivedMessage; ICommand command = this.GetCommand(e); } /// <summary> /// Gets the Command that the player entered and preps it for execution. /// </summary> /// <param name="command"></param> /// <returns>Returns the ICommand specified.</returns> public Commands.ICommand GetCommand(IMessage command) { if (this.currentState == CurrentState.FetchUserName) { this.connectedPlayer.Name = command.Message; this.currentState = CurrentState.FetchPassword; } else if (this.currentState == CurrentState.FetchPassword) { // find user } return new NoOpCommand(); } /// <summary> /// Cleanups this instance during a state change. /// </summary> public void Cleanup() { // If we have a player instance, we clean up the registered event. if (this.connectedPlayer != null) { this.connectedPlayer.ReceivedMessage -= this.connectedPlayer_ReceivedMessage; } } Maybe my entire FSM isn't wired up in the best way, but I would appreciate input on what would be the best to maintain a stack of state in a MUD game engine, and if my states should be allowed to receive the input from the user or not to check what command was entered before allowing the state manager to switch states. Thanks in advance.

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  • TCP and UDP are using different OS Buffer?

    - by Jack
    HI all. Here is the scenario. I have port 8888 for my program to use. I build a TCP and a UDP listener on that port. (This can do, c# allows, because they are two different protocols) My question is If the network traffic is very busy, TCP sockets may refuse or signalling the other end to stop sending things, it is called congestion control, right? So if TCP is congestion controlling, other ends may not send more data, in this "TCP quiet period", UDP channel should have not that much of traffic, right? I want to figure out the TCP traffic will affect UDP traffic or not?

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  • Maximum number of bytes that can be sent on a TCP connection

    - by iamrohitbanga
    I initially assumed that since tcp has a sequence number field of 32 bits and each byte sent on a tcp connection is labeled with a unique number, maximum number of bytes that can be sent on a tcp connection is about 2^32-1 or 2^32-2 (which?). but now I feel that since TCP is a sliding window protocol, the wraparound of sequence numbers during the connection should not have an affect on the maximum number of bytes that can be sent over a tcp connection as long as the when wraparound occurs the old packet is no longer in the network (it is sent after 2*MSL). What is the correct answer?

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  • SSH stops at "using username" with IPTables in effect

    - by Rautamiekka
    We used UFW but couldn't make the Source Dedicated ports open, which was weird, so we purged UFW and switched to IPTables, using Webmin to configure. If the inbound chain is on DENY and SSH port open [judged from Webmin], PuTTY will say using username "root" and stops at that instead of asking for public key pw. Inbound chain on ACCEPT the pw is asked. This problem didn't happen with UFW. Picture of IPTables configuration in Webmin: http://s284544448.onlinehome.us/public/PlusLINE%20Dedicated%20Server,%20Webmin,%20IPTables,%200.jpgThe address is to the previous rautamiekka.org. iptables-save when on INPUT DENY: # Generated by iptables-save v1.4.8 on Wed Apr 11 16:09:20 2012 *mangle :PREROUTING ACCEPT [1430:156843] :INPUT ACCEPT [1430:156843] :FORWARD ACCEPT [0:0] :OUTPUT ACCEPT [1415:781598] :POSTROUTING ACCEPT [1415:781598] COMMIT # Completed on Wed Apr 11 16:09:20 2012 # Generated by iptables-save v1.4.8 on Wed Apr 11 16:09:20 2012 *nat :PREROUTING ACCEPT [2:104] :POSTROUTING ACCEPT [0:0] :OUTPUT ACCEPT [0:0] COMMIT # Completed on Wed Apr 11 16:09:20 2012 # Generated by iptables-save v1.4.8 on Wed Apr 11 16:09:20 2012 *filter :INPUT DROP [0:0] :FORWARD ACCEPT [0:0] :OUTPUT ACCEPT [1247:708906] -A INPUT -i lo -m comment --comment "Machine-within traffic - always allowed" -j ACCEPT -A INPUT -p tcp -m comment --comment "Services - TCP" -m tcp -m multiport --dports 22,80,443,10000,20,21 -m state --state NEW,ESTABLISHED -j ACCEPT -A INPUT -p tcp -m comment --comment "Minecraft - TCP" -m tcp --dport 25565 -j ACCEPT -A INPUT -p udp -m comment --comment "Minecraft - UDP" -m udp --dport 25565 -j ACCEPT -A INPUT -p tcp -m comment --comment "Source Dedicated - TCP" -m tcp --dport 27015 -j ACCEPT -A INPUT -p udp -m comment --comment "Source Dedicated - UDP" -m udp -m multiport --dports 4380,27000:27030 -j ACCEPT -A INPUT -p udp -m comment --comment "TS3 - UDP - main port" -m udp --dport 9987 -j ACCEPT -A INPUT -p tcp -m comment --comment "TS3 - TCP - ServerQuery" -m tcp --dport 10011 -j ACCEPT -A OUTPUT -o lo -m comment --comment "Machine-within traffic - always allowed" -j ACCEPT COMMIT # Completed on Wed Apr 11 16:09:20 2012 iptables --list when on INPUT DENY: Chain INPUT (policy DROP) target prot opt source destination ACCEPT all -- anywhere anywhere /* Machine-within traffic - always allowed */ ACCEPT tcp -- anywhere anywhere /* Services - TCP */ tcp multiport dports ssh,www,https,webmin,ftp-data,ftp state NEW,ESTABLISHED ACCEPT tcp -- anywhere anywhere /* Minecraft - TCP */ tcp dpt:25565 ACCEPT udp -- anywhere anywhere /* Minecraft - UDP */ udp dpt:25565 ACCEPT tcp -- anywhere anywhere /* Source Dedicated - TCP */ tcp dpt:27015 ACCEPT udp -- anywhere anywhere /* Source Dedicated - UDP */ udp multiport dports 4380,27000:27030 ACCEPT udp -- anywhere anywhere /* TS3 - UDP - main port */ udp dpt:9987 ACCEPT tcp -- anywhere anywhere /* TS3 - TCP - ServerQuery */ tcp dpt:10011 Chain FORWARD (policy ACCEPT) target prot opt source destination Chain OUTPUT (policy ACCEPT) target prot opt source destination ACCEPT all -- anywhere anywhere /* Machine-within traffic - always allowed */ The UFW rules prior to purging on INPUT DENY: 127.0.0.1 ALLOW IN 127.0.0.1 3306 DENY IN Anywhere 20,21/tcp ALLOW IN Anywhere 22/tcp (OpenSSH) ALLOW IN Anywhere 80/tcp ALLOW IN Anywhere 443/tcp ALLOW IN Anywhere 989 ALLOW IN Anywhere 990 ALLOW IN Anywhere 8075/tcp ALLOW IN Anywhere 9987/udp ALLOW IN Anywhere 10000/tcp ALLOW IN Anywhere 10011/tcp ALLOW IN Anywhere 25565/tcp ALLOW IN Anywhere 27000:27030/tcp ALLOW IN Anywhere 4380/udp ALLOW IN Anywhere 27014:27050/tcp ALLOW IN Anywhere 30033/tcp ALLOW IN Anywhere

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  • Finding cause of TCP retransmission within a LAN

    - by Surreal
    Hello denizens of Server Fault I have an irritating problem with a LAN of about 100 computers, 2 Windows domain servers, and 12 VoIP phones. Since their installation around a year ago, every week or so, we notice a VoIP phone resetting itself - occasionally in the middle of a call. Simultaneously there are often signs of temporary loss of connection on computers: freezes in explorer while accessing network shares, errors in our administration software due to loss of connection to the database server. I have been doing some Wireshark monitoring on the connection between the VoIP PBX and the rest of the network. Wireshark picks up a clump of retransmitted TCP packets at the times when we record phone restarts. The Wireshark log shows about 2 clusters of retransmissions a day ranging from 5 packets to hundreds. Those in each cluster are mainly between the PBX and some set of the VoIP phones, but not always the same set. Often retransmissions at the same time are to phones connected to the same switch, but sometimes retransmissions occur together to phones at opposite ends of the network. There are usually some coincident retransmissions in passing TCP traffic, for example between client machines and the file servers. The spikes in retransmissions and phone resets do not correlate well with when the network is heavily loaded. They seem to occur slightly more during the day, but most in the evening, when traffic should be decreasing. They occur reasonably often late at night when most computers are turned off and traffic should be lowest. Do you have any ideas that might help diagnose the cause of problems like this? One thing I have not yet tried, but should have, is updating the firmware of all the switches.

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  • Cannot start listening on a certain TCP port, but there's nothing currently listening on it

    - by John Rasch
    I have Windows Service that uses a WCF service host to listen for connections on TCP port 61000. When I try to start the service, I get the error: Service cannot be started. System.ServiceModel.AddressAlreadyInUseException: HTTP could not register URL http://+:61000/ because TCP port 61000 is being used by another application. ---> System.Net.HttpListenerException: The process cannot access the file because it is being used by another process at System.Net.HttpListener.AddAll() at System.Net.HttpListener.Start() at System.ServiceModel.Channels.SharedHttpTransportManager.OnOpen() --- End of inner exception stack trace --- at System.ServiceModel.Channels.SharedHttpTransportManager.OnOpen() at System.ServiceModel.Channels.TransportManager.Open(TransportChannelListener channelListener) at System.ServiceModel.Channels.TransportManagerContainer.Open(SelectTransportManagersCallback selectTransportManagerCallback) at System.ServiceModel.Channels.HttpChannelListener.OnOpen(TimeSpan timeout) at System.ServiceModel.Channels.CommunicationObject.Open(TimeSpan timeout) at System.ServiceModel.Dispatcher.ChannelDispatcher.OnOpen(TimeSpan timeout) at... A quick netstat -a shows there is nothing listening on port 61000. I've also found several posts online that mention reserving namespaces using netstat, but the account that the service runs under has administrator privileges so that shouldn't be necessary. Any other ideas as to why I'm getting this message? This service is running on 64-bit Windows Server 2008 R2 Standard.

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  • What characteristic of networking/TCP causes linear relation between TCP activity and latency?

    - by DeLongey
    The core of this problem is that our application uses websockets for real-time interfaces. We are testing our app in a new environment but strangely we're noticing an increasing delay in TCP websocket packets associated with an increase in websocket activity. For example, if one websocket event occurs without any other activity in a 1-minute period, the response from the server is instantaneous. However, if we slowly increase client activity the latency in server response increases with a linear relationship (each packet will take more time to reach the client with more activity). For those wondering this is NOT app-related since our logs show that our server is running and responding to requests in under 100ms as desired. The delay starts once the server processes the request and creates the TCP packet and sends it to the client (and not the other way around). Architecture This new environment runs with a Virtual IP address and uses keepalived on a load balancer to balance the traffic between instances. Two boxes sit behind the balancer and all traffic runs through it. Our host provider manages the balancer and we do not have control over that part of the architecture. Theory Could this somehow be related to something buffering the packets in the new environment? Thanks for your help.

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  • Finding cause of TCP retransmission within a LAN

    - by Surreal
    Hello denizens of Server Fault I have an irritating problem with a LAN of about 100 computers, 2 Windows domain servers, and 12 VoIP phones. Since their installation around a year ago, every week or so, we notice a VoIP phone resetting itself - occasionally in the middle of a call. Simultaneously there are often signs of temporary loss of connection on computers: freezes in explorer while accessing network shares, errors in our administration software due to loss of connection to the database server. I have been doing some Wireshark monitoring on the connection between the VoIP PBX and the rest of the network. Wireshark picks up a clump of retransmitted TCP packets at the times when we record phone restarts. The Wireshark log shows about 2 clusters of retransmissions a day ranging from 5 packets to hundreds. Those in each cluster are mainly between the PBX and some set of the VoIP phones, but not always the same set. Often retransmissions at the same time are to phones connected to the same switch, but sometimes retransmissions occur together to phones at opposite ends of the network. There are usually some coincident retransmissions in passing TCP traffic, for example between client machines and the file servers. The spikes in retransmissions and phone resets do not correlate well with when the network is heavily loaded. They seem to occur slightly more during the day, but most in the evening, when traffic should be decreasing. They occur reasonably often late at night when most computers are turned off and traffic should be lowest. Do you have any ideas that might help diagnose the cause of problems like this? One thing I have not yet tried, but should have, is updating the firmware of all the switches.

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  • Gigabit network limited to 25MB/s by CPU. How to make it faster?

    - by netvope
    I have a Acer Aspire R1600-U910H with a nForce gigabit network adapter. The maximum TCP throughput of it is about 25MB/s, and apparently it is limited by the single core Intel Atom 230; when the maximum throughput is reached, the CPU usage is about 50%-60%, which corresponds to full utilization considering this is a Hyper-threading enabled CPU. The same problem occurs on both Windows XP and on Ubuntu 8.04. On Windows, I have installed the latest nForce chipset driver, disabled power saving features, and enabled checksum offload. On Linux, the default driver has checksum offload enabled. There is no Linux driver available on Nvidia's website. ethtool -k eth0 shows that checksum offload is enabled: Offload parameters for eth0: rx-checksumming: on tx-checksumming: on scatter-gather: on tcp segmentation offload: on udp fragmentation offload: off generic segmentation offload: off The following is the output of powertop when the network is idle: Wakeups-from-idle per second : 61.9 interval: 10.0s no ACPI power usage estimate available Top causes for wakeups: 90.9% (101.3) <interrupt> : eth0 4.5% ( 5.0) iftop : schedule_timeout (process_timeout) 1.8% ( 2.0) <kernel core> : clocksource_register (clocksource_watchdog) 0.9% ( 1.0) dhcdbd : schedule_timeout (process_timeout) 0.5% ( 0.6) <kernel core> : neigh_table_init_no_netlink (neigh_periodic_timer) And when the maximum throughput of about 25MB/s is reached: Wakeups-from-idle per second : 11175.5 interval: 10.0s no ACPI power usage estimate available Top causes for wakeups: 99.9% (22097.4) <interrupt> : eth0 0.0% ( 5.0) iftop : schedule_timeout (process_timeout) 0.0% ( 2.0) <kernel core> : clocksource_register (clocksource_watchdog) 0.0% ( 1.0) dhcdbd : schedule_timeout (process_timeout) 0.0% ( 0.6) <kernel core> : neigh_table_init_no_netlink (neigh_periodic_timer) Notice the 20000 interrupts per second. Could this be the cause for the high CPU usage and low throughput? If so, how can I improve the situation? The other computers in the network can usually transfer at 50+MB/s without problems. And a minor question: How can I find out what is the driver in use for eth0?

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  • Game show game engine [closed]

    - by Red
    So, I am pretty new to the world of game development, so I am a bit fuzzy on what I require. Could someone suggest a game engine that I could use? I need it to be light weight (my game won't require that much power) and have networking functionality for multiplay or even an MMO aspect. The game I am making is like a game show, so it is your basic choose and answer hit the buzzer kind of game. Any suggestions? I would also like it to be open source or at the least free. I would like to support open source projects.

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  • Good, free isometric game engine?

    - by posfan12
    Any recommendations for a good isometric game engine that is also free? Should be possible to develop entirely using freely available tools (meaning: no Flash, and no I don't want to learn haXe...) Works-in-a-browser is a plus, but not required. Support for 32-bit images is required! Good performance. Excellent documentation. I have looked at FIFE but it is still too unfinished, and the documentation sucks! Thanks!

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  • Farseer Physics Engine and the Ms-PL License

    - by Stephen Tierney
    Am I able to produce code for a game which uses the Farseer engine and release my code under an open source license other than the Ms-PL? My concern is with the following section from the license: If you distribute any portion of the software in source code form, you may do so only under this license by including a complete copy of this license with your distribution. If you distribute any portion of the software in compiled or object code form, you may only do so under a license that complies with this license. If I do not include Farseer in my source code distribution does this give me an exemption from this clause as I am not distributing the software? My code merely uses its functions. No where in the license does it force you to provide source code for derivative works or linking works, it simply gives you the option of "if you distribute".

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  • 2D Physics Engine to Handle Shapes Composed of Multiple Densities XNA

    - by Stupac
    The game I'm working on involves shapes that might be composed of multiple materials in a variety of ways. Let's just take for example a wooden rod with and sizable tip of iron or say a block composed of a couple triangles of stone and aluminum and small nugget of gold. The shapes and compositions will change from time to time, so I was wondering what engine I should use and how I might implement this feature? I've looked at Farseer 3, but I'm still trying to decipher the library by reading the source and the samples and wasn't sure if I was barking up the wrong tree. Thanks!

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