Search Results

Search found 1671 results on 67 pages for 'packets'.

Page 28/67 | < Previous Page | 24 25 26 27 28 29 30 31 32 33 34 35  | Next Page >

  • VPN still working after rebooting without client - DrayTek client shows "No Connection"

    - by HeavenCore
    My home network is a simple router + pc's setup, nothing fancy - the router has DHCP enabled for 192.168.0.X (255.255.255.0) and my PC picks up the address 192.168.0.82. There are no devices on my local lan in the 192.168.1.x range. On my pc i have the DrayTek VPN client, and a company i do some work for has a DrayTek Vigor router. The VPN client establishes a VPN to that remote company using an IPSec Tunnel (PreShared Key - no encryption) Last night i shut down my pc with the VPN tunnel still connected, when i turned my computer on this morning i accidentally clicked an RDP shortcut to 192.168.1.2 (a host in the remote company) and to my amazement it connected?!? I checked and the DrayTek VPN client isnt running, and when i did run it, it clearly shows "Status: No connection". confused as to how my machine can still talk to this remote machine i tried a trace: C:\Users\HeavenCore>tracert 192.168.1.2 Tracing route to C4SERVERII [192.168.1.2] over a maximum of 30 hops: 1 * * * Request timed out. 2 * * * Request timed out. 3 * * * Request timed out. 4 * * * Request timed out. 5 * * * Request timed out. 6 * * * Request timed out. 7 * * * Request timed out. 8 * * * Request timed out. 9 * * * Request timed out. 10 * * * Request timed out. 11 * * * Request timed out. 12 15 ms 21 ms 32 ms C4SERVERII [192.168.1.2] Trace complete. No indication there as to how it's getting from my network to the remote host. with my network mask being 255.255.255.0 with ip 192.168.0.1 i dont even see how packets are routing to 192.168.1.1 - unless there was a static route in place, so i checked the route table: IPv4 Route Table =========================================================================== Active Routes: Network Destination Netmask Gateway Interface Metric 0.0.0.0 0.0.0.0 192.168.0.1 192.168.0.82 266 127.0.0.0 255.0.0.0 On-link 127.0.0.1 306 127.0.0.1 255.255.255.255 On-link 127.0.0.1 306 127.255.255.255 255.255.255.255 On-link 127.0.0.1 306 192.168.0.0 255.255.255.0 On-link 192.168.0.82 266 192.168.0.82 255.255.255.255 On-link 192.168.0.82 266 192.168.0.255 255.255.255.255 On-link 192.168.0.82 266 224.0.0.0 240.0.0.0 On-link 127.0.0.1 306 224.0.0.0 240.0.0.0 On-link 192.168.0.82 266 255.255.255.255 255.255.255.255 On-link 127.0.0.1 306 255.255.255.255 255.255.255.255 On-link 192.168.0.82 266 =========================================================================== Persistent Routes: Network Address Netmask Gateway Address Metric 0.0.0.0 0.0.0.0 192.168.0.1 Default =========================================================================== As far as i can see, nothing indicating how my packets are getting to 192.168.1.2??? To confirm i was on a different subnet i did an ipconfig /all: Ethernet adapter Local Area Connection: Connection-specific DNS Suffix . : Description . . . . . . . . . . . : Marvell Yukon 88E8056 PCI-E Gigabit Ether net Controller Physical Address. . . . . . . . . : 00-23-54-F3-4E-BA DHCP Enabled. . . . . . . . . . . : No Autoconfiguration Enabled . . . . : Yes IPv4 Address. . . . . . . . . . . : 192.168.0.82(Preferred) Subnet Mask . . . . . . . . . . . : 255.255.255.0 Default Gateway . . . . . . . . . : 192.168.0.1 DNS Servers . . . . . . . . . . . : 192.168.0.1 208.67.222.222 NetBIOS over Tcpip. . . . . . . . : Enabled Yet straight after confirming my ip and subnet as above i can go ahead and ping the remote machine: C:\Users\HeavenCore>ping 192.168.1.2 Pinging 192.168.1.2 with 32 bytes of data: Reply from 192.168.1.2: bytes=32 time=48ms TTL=127 Reply from 192.168.1.2: bytes=32 time=23ms TTL=127 Reply from 192.168.1.2: bytes=32 time=103ms TTL=127 Reply from 192.168.1.2: bytes=32 time=25ms TTL=127 Ping statistics for 192.168.1.2: Packets: Sent = 4, Received = 4, Lost = 0 (0% loss), Approximate round trip times in milli-seconds: Minimum = 23ms, Maximum = 103ms, Average = 49ms Also, note on the ping how the times are 35ms ish, this clearly shows the pings are to the remote host and not something on my local lan (all stuff on my local lan pings in 0ms) - plus i verified the host was actually the host via RDP. My Question: Can an IPSec tunnel stay up some how after a reboot without use of the VPN client? (well, i can clearly see that it can) - where in windows is there visibility of this? how does my machine know where to route the packets? I appreciate any insights & thoughts!

    Read the article

  • How do I prevent TCP connection freezes over an OpenVPN network?

    - by Jason R
    New details added at the end of this question; it's possible that I'm zeroing in on the cause. I have a UDP OpenVPN-based VPN set up in tap mode (I need tap because I need the VPN to pass multicast packets, which doesn't seem to be possible with tun networks) with a handful of clients across the Internet. I've been experiencing frequent TCP connection freezes over the VPN. That is, I will establish a TCP connection (e.g. an SSH connection, but other protocols have similar issues), and at some point during the session, it seems that traffic will cease being transmitted over that TCP session. This seems to be related to points at which large data transfers occur, such as if I execute an ls command in an SSH session, or if I cat a long log file. Some Google searches turn up a number of answers like this previous one on Server Fault, indicating that the likely culprit is an MTU issue: that during periods of high traffic, the VPN is trying to send packets that get dropped somewhere in the pipes between the VPN endpoints. The above-linked answer suggests using the following OpenVPN configuration settings to mitigate the problem: fragment 1400 mssfix This should limit the MTU used on the VPN to 1400 bytes and fix the TCP maximum segment size to prevent the generation of any packets larger than that. This seems to mitigate the problem a bit, but I still frequently see the freezes. I've tried a number of sizes as arguments to the fragment directive: 1200, 1000, 576, all with similar results. I can't think of any strange network topology between the two ends that could trigger such a problem: the VPN server is running on a pfSense machine connected directly to the Internet, and my client is also connected directly to the Internet at another location. One other strange piece of the puzzle: if I run the tracepath utility, then that seems to band-aid the problem. A sample run looks like: [~]$ tracepath -n 192.168.100.91 1: 192.168.100.90 0.039ms pmtu 1500 1: 192.168.100.91 40.823ms reached 1: 192.168.100.91 19.846ms reached Resume: pmtu 1500 hops 1 back 64 The above run is between two clients on the VPN: I initiated the trace from 192.168.100.90 to the destination of 192.168.100.91. Both clients were configured with fragment 1200; mssfix; in an attempt to limit the MTU used on the link. The above results would seem to suggest that tracepath was able to detect a path MTU of 1500 bytes between the two clients. I would assume that it would be somewhat smaller due to the fragmentation settings specified in the OpenVPN configuration. I found that result somewhat strange. Even stranger, however: if I have a TCP connection in the stalled state (e.g. an SSH session with a directory listing that froze in the middle), then executing the tracepath command shown above causes the connection to start up again! I can't figure out any reasonable explanation for why this would be the case, but I feel like this might be pointing toward a solution to ultimately eradicate the problem. Does anyone have any recommendations for other things to try? Edit: I've come back and looked at this a bit further, and have found only more confounding information: I set the OpenVPN connection to fragment at 1400 bytes, as shown above. Then, I connected to the VPN from across the Internet and used Wireshark to look at the UDP packets that were sent to the VPN server while the stall occurred. None were greater than the specified 1400 byte count, so the fragmentation seems to be functioning properly. To verify that even a 1400-byte MTU would be sufficient, I pinged the VPN server using the following (Linux) command: ping <host> -s 1450 -M do This (I believe) sends a 1450-byte packet with fragmentation disabled (I at least verified that it didn't work if I set it to an obviously-too-large value like 1600 bytes). These seem to work just fine; I get replies back from the host with no issue. So, maybe this isn't an MTU issue at all. I'm just confused as to what else it might be! Edit 2: The rabbit hole just keeps getting deeper: I've now isolated the problem a bit more. It seems to be related to the exact OS that the VPN client uses. I have successfully duplicated the problem on at least three Ubuntu machines (versions 12.04 through 13.04). I can reliably duplicate an SSH connection freeze within a minute or so by just cat-ing a large log file. However, if I do the same test using a CentOS 6 machine as a client, then I don't see the problem! I've tested using the exact same OpenVPN client version as I was using on the Ubuntu machines. I can cat log files for hours without seeing the connection freeze. This seems to provide some insight as to the ultimate cause, but I'm just not sure what that insight is. I have examined the traffic over the VPN using Wireshark. I'm not a TCP expert, so I'm not sure what to make of the gory details, but the gist is that at some point, a UDP packet gets dropped due to the limited bandwidth of the Internet link, causing TCP retransmissions inside the VPN tunnel. On the CentOS client, these retransmissions occur properly and things move on happily. At some point with the Ubuntu clients, though, the remote end starts retransmitting the same TCP segment over and over (with the transmit delay increasing between each retransmission). The client sends what looks like a valid TCP ACK to each retransmission, but the remote end still continues to transmit the same TCP segment periodically. This extends ad infinitum and the connection stalls. My question here would be: Does anyone have any recommendations for how to troubleshoot and/or determine the root cause of the TCP issue? It's as if the remote end isn't accepting the ACK messages sent by the VPN client. One common difference between the CentOS node and the various Ubuntu releases is that Ubuntu has a much more recent Linux kernel version (from 3.2 in Ubuntu 12.04 to 3.8 in 13.04). A pointer to some new kernel bug maybe? I'm assuming that if that were so, then I wouldn't be the only one experiencing the problem; I don't think this seems like a particularly exotic setup.

    Read the article

  • Capturing network traffic in ruby - pcap related issues

    - by Acidburn2k
    What I need is to write very simple application, which would listen to network traffic, filter out some packets based on various layer 4/5 information and then dump those information into database. I am quite confused on which pcap gem/plugin should I use. The basic pcap implemention seem to be a bit outdated (no changes since 2001) and doesn't work properly. I also tried pcaprub, but I am not quite sure how to get around with this library. It seem to capture raw packets without te ability to actualy get any data out of the pcap dump. Do you have any advices on how can I realize this simple task? Thanks in advance. :-)

    Read the article

  • MTU mismatch between GetIfEntry and netsh

    - by ChrisJ
    I'm working on pseudo-transport layer software that runs over UDP, providing reliable connection-oriented transmission, as an alternative to TCP. In order to maximize network efficiency, we query the MTU of the "best" network adapter upon initialization. MIB_IFROW row = {0}; row.dwIndex = dwBestIfIndex; dwRes = GetIfEntry(&row); Searching online I found that you can use the following netsh commands to query for this same value, from a command prompt (not a C++ API) netsh interface ipv4 show interfaces netsh interface ipv4 show subinterfaces The troubling issue is that while row.dwMtu may be set to 1500, snooping the network traffic on the sending laptop shows that our packets are fragmented into 1300 byte packets. netsh also reports that the MTU is 1300. Clearly the value reported by netsh command is the actual used values. Anyone know what API I can call to get the same values as netsh?

    Read the article

  • How to debug packet loss ?

    - by Gene Vincent
    I wrote a C++ application (running on Linux) that serves an RTP stream of about 400 kbps. To most destinations this works fine, but some destinations expericence packet loss. The problematic destinations seem to have a slower connection in common, but it should be plenty fast enough for the stream I'm sending. Since these destinations are able to receive similar RTP streams for other applications without packet loss, my application might be at fault. I already verified a few things: - in a tcpdump, I see all RTP packets going out on the sending machine - there is a UDP send buffer in place (I tried sizes between 64KB and 300KB) - the RTP packets mostly stay below 1400 bytes to avoid fragmentation What can a sending application do to minimize the possibility of packet loss and what would be the best way to debug such a situation ?

    Read the article

  • Does any one know of a packet generator?

    - by Benoit
    We have a networked device, and we would like to perform some tests on how it handles malformed packets. Is there a product out there that can generate arbitrary packets and packet sequences? I would like to be able to specify a set of TCP/IP payloads and it would open a connection and send the data. Obviouly, the TCP/IP checksum should be calculated correctly, etc... Kind of like a wireshark in reverse. Note that I am not interested in network loading and blasting millions of packet.

    Read the article

  • 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?

    Read the article

  • Java library to encode / decode AMF

    - by Ceilingfish
    Hi chaps, I currently have a Java server that talks to a Flash client by passing JSON encoded data over a binary socket connection. Is there a way on either side to encode / decode packets as AMF instead of JSON? It seems to me that there should be some native support in Flash player for doing this? All the implementations I have found of AMF serialization seem to be embedded inside an application framework. Simiarly so, does anyone know if it's possible to decode AMF packets independently of a connection implementation in Flash?

    Read the article

  • libnet that properly calculates checksum on IPV6

    - by VeaEm
    I have recently started playing around with libnet and using it to generate IPV6 packets. I am very new at programming, however, I am quite happy with the library. I have one problem with it though. It seems that libnet currently does not have the ability to properly calculate checksums on IPV6 packets. Being so new to programming, I am not yet capable of fixing this problem (although I am learning, so that one day I can). I am curious, has anyone run across a version of the library that can do this properly? Thanks!

    Read the article

  • Does Android support near real time push notification

    - by j pimmel
    I recently learned about the ability of iPhone apps to receive nearly instantaneous notifications to apps. This is provided in the form of push notifications, a bespoke protocol which keeps an always on data connection to the iPhone and messages binary packets to the app, which pops up alerts incredibly quickly, between 0.5 - 5 seconds from server app send to phone app response time. This is sent as data - rather than SMS - in very very small packets charged as part of the data plan not as incoming messages. I would like to know if using Android there is either a similar facility, or whether it's possible to implement something close to this using Android APIs. To clarify I define similar as: Not an SMS message, but some data driven solution As real time as is possible Is scalable - ie: as the server part of a mobile app, I could notify thousands of app instances in seconds I appreciate the app could be pull based, HTTP request/response style, but ideally I don't want to to be polling that heavily just to check for notification .. besides which it's like drip draining the data plan.

    Read the article

  • How does one capture H.323 voice traffic on a VOIP network?

    - by Chris Holmes
    What I am trying to do is capture the WAV data of a phone conversation on a VOIP network using SharpPCap/PCap.Net. We are using the H.323 recommendation and my understanding is that voice data is located in the RTP packets. However, there is no way to heuristically determine if a UDP packet is a RTP packet, so we have to do more work before we can capture the data. The H.323 recommendation apparently uses a lot of traffic on specific TCP ports to negotiate the call before the WAV data is sent via RTP. However, I am having very little luck determining what data is actually sent on those TCP ports, when it is sent, what the packets look like, how to handle it, etc. If anyone has any information on how to go about this I'd really appreciate it. My Google-Fu seems to be failing me on this one.

    Read the article

  • C# Proxy using Sockets, how should I do this?

    - by Kin
    I'm writing a proxy using .NET and C#. I haven't done much Socket programming, and I am not sure the best way to go about it. What would be the best way to implement this? Should I use Synchronous Sockets, Asynchronous sockets? Please help! It must... Accept Connections from the client on two different ports, and be able to receive data on both ports at the same time. Connect to the server on two different ports, and be able to send data on both ports as the same time. Immediately connect to the server and start forwarding packets as soon as a client connection is made. Forward packets in the same order they were received. Be as low latency as possible. I don't need the ability for multiple clients to connect to the proxy, but it would be a nice feature if its easy to implement. Client --------- Proxy ------- Server ---|-----------------|----------------| Port <-------- Port <------- Port Port <-------- Port <------- Port

    Read the article

  • How to implement blocking request-reply using Java concurrency primitives?

    - by Uri
    My system consists of a "proxy" class that receives "request" packets, marshals them and sends them over the network to a server, which unmarshals them, processes, and returns some "response packet". My "submit" method on the proxy side should block until a reply is received to the request (packets have ids for identification and referencing purposes) or until a timeout is reached. If I was building this in early versions of Java, I would likely implement in my proxy a collection of "pending messages ids", where I would submit a message, and wait() on the corresponding id (with a timeout). When a reply was received, the handling thread would notify() on the corresponding id. Is there a better way to achieve this using an existing library class, perhaps in java.util.concurrency? If I went with the solution described above, what is the correct way to deal with the potential race condition where a reply arrives before wait() is invoked?

    Read the article

  • Create multiple TCP Connections in C# then wait for data

    - by Ryan French
    Hi Everyone, I am currently creating a Windows Service that will create TCP connections to multiple machines (same socket on all machines) and then listen for 'events' from those machines. I am attempting to write the code to create a connection and then spawn a thread that listens to the connection waiting for packets from the machine, then decode the packets that come through, and call a function depending on the payload of the packet. The problem is I'm not entirely sure how to do that in C#. Does anyone have any helpful suggestions or links that might help me do this? Thanks in advance for any help!

    Read the article

  • How to implement RFC 3393 (Ipdv packet delay varation) in C?

    - by sagar
    Hello , I am building an Ethernet Application in which i will be sending packets from one side and receiving it on the other side. I want to calculate delay in packets at the receiver side as in RFC 3393. So I have to put a timestamps in the packet at the sender side and then take the timestamps at the receiver side as soon as i receive the packet . Subtracting the values i will get the difference in timestamps and then subtracting this value with subsequent difference i will get One way ipdv delay . Both the clocks are not synchronized . So any help is greatly appreciated. Thank you.

    Read the article

  • C# Socket ReceiveAll

    - by rielz
    Hey there! I am trying to capture ip packets in c#. Everything is working fine, except that i only get outgoing packets. My Code: using (Socket sock = new Socket(AddressFamily.InterNetwork, SocketType.Raw, ProtocolType.IP)) { sock.Bind(new IPEndPoint(LOCALHOST, 0)); sock.SetSocketOption(SocketOptionLevel.IP, SocketOptionName.HeaderIncluded, true); sock.IOControl(IOControlCode.ReceiveAll, BitConverter.GetBytes(1), null); while (true) { byte[] buffer = new byte[sock.ReceiveBufferSize]; int count = sock.Receive(buffer); // ... } } Does anyone have an idea? :( Doesnt find any solutions at Google, ... Thank you in advance.

    Read the article

  • Socket ReceiveAll

    - by rielz
    I am trying to capture ip packets in c#. Everything is working fine, except that i only get outgoing packets. My Code: using (Socket sock = new Socket(AddressFamily.InterNetwork, SocketType.Raw, ProtocolType.IP)) { sock.Bind(new IPEndPoint(MYADDRESS, 0)); sock.SetSocketOption(SocketOptionLevel.IP, SocketOptionName.HeaderIncluded, true); sock.IOControl(IOControlCode.ReceiveAll, BitConverter.GetBytes(1), null); while (true) { byte[] buffer = new byte[sock.ReceiveBufferSize]; int count = sock.Receive(buffer); // ... } } The problem is definitely my pc! But maybe there is a workaround ...

    Read the article

  • Building an http packet in libnet(tcp packet), Please help us as soon as posible. we are stuck!

    - by Hila
    we are building a NAT program,we change each packet that comes from our internal subnet, change it's source IP address by libnet functions.( catch the packet with libpcap, put it sniff structures and build the new packet with libnet) over TCP, the syn/ack packets are good after the change, and when a HTTP-GET request is coming, we can see by wireshark that there is an error on the checksum field.. all the other fields are exactly the same as the original packet. Is anyone knows what can cause this problem? the new checksum in other packets is calculated as it should be.. but in the HTTP packet it doesn't..

    Read the article

  • selecting ppp of multiple interfaces

    - by Neeraj
    Hi everyone, I am currently having a hard time on getting the mobile broadband connection running on ubuntu 10.04(lucid lynx). I am using a USB modem and used wvdial to connect to the web It went like: Sending ATZ ... OK sending some more flags OK modem initialized connecting Local IP x.x.x.x Remote IP y.y.y.y Primary DNS z.z.z.z Secondary DNS a.a.a.a (Some more output) I tested this with invalid username to make sure it is really connected and i think it was connected as connection failed with invalid usernames. Now when I do a remote ping say 8.8.8.8 (Google DNS servers), the ping says unreachable I think this might be because the system may be using the ethernet to send packets which was indeed disconnected. So can anyone help me out with this. How can I select ppp as the interface to send packets or is there some other problem. A command line solution will be appreciated as my network manager applet doesnt works correctly. Any help is much much appreciated. -- thx

    Read the article

  • how does teamviewer find my computer even if my comp. behind of the firewall and firewall isn't conf

    - by uzay95
    Did you use teamviewer? (comic question i know... Who doesn't use it?) Do you have any idea how does teamviewer make connection even if i am behind the router, firewall, switch and my local firewall..? I'm trying to imagine a connection that is between remote machinge and my computer. Remote machine is sending the packets (and its header (for instance, destination IP, message body)) to me but it only knows my id number(which is given by my local teamviewer application). And this packets are reaching to my computer even if there is a juniper firewall (and also my windows firewall). What kind a message body is recieving by computer? (of course it is not like xml, text, html, excel :) Do you have any idea? PS. Please share your knowledge like you are explaining to beginner level user.

    Read the article

  • A non-blocking server with java.io

    - by Jon
    Everybody knows that java IO is blocking, and java NIO is non-blocking. In IO you will have to use the thread per client pattern, in NIO you can use one thread for all clients. Now my question follows: is it possible to make a non-blocking design using only the Java IO api. (not NIO) I was thinking about a pattern like this (obviously very simplified); List<Socket> li; for (Socket s : li) { InputStream in = s.getInputStream(); byte[] data = in.available(); in.read(data); // processData(data); (decoding packets, encoding outgoing packets } Also note that the client will always be ready for reading data. What are your opinions on this? Will this be suitable for a server that should at least hold a few hundred of clients without major performance issues?

    Read the article

  • bandwidth throttling C linux

    - by bob moch
    hi im currently creating a function to create a sleep time i can pause between packets for my port scanner im creating for personal/educational use for my home network. what im currently doing is opening /proc/net/dev and reading the 9th set of digits for the eth0 interface to find out the current packets being set and then reading it again and doing some math to figure out a delay to sleep between sending a packet to a port to identify it and fingerprint it. my problem is that no matter what throttle % i use it always seems to send the same rate of packets. i think its mainly my way of mathematically creating my sleep delay. edit:: dont mind the function declaration and the struct stuff all im doing is spawning this function in a thread and passing a pointer to a struct to the function, recreating the struct locally and then freeing the passed structs memory. void *bandwidthmonitor_cmd(void *param) { char cmdline[1024], *bytedata[19]; int i = 0, ii = 0; long long prevbytes = 0, currentbytes = 0, elapsedbytes = 0, byteusage = 0, maxthrottle = 0; command_struct bandwidth = *((command_struct *)param); free(param); //printf("speed: %d\n throttle: %d\n\n", UPLOAD_SPEED, bandwidth.throttle); maxthrottle = UPLOAD_SPEED * bandwidth.throttle / 100; //printf("max throttle:%lld\n", maxthrottle); FILE *f = fopen("/proc/net/dev", "r"); if(f != NULL) { while(1) { while(fgets(cmdline, sizeof(cmdline), f) != NULL) { cmdline[strlen(cmdline)] = '\0'; if(strncmp(cmdline, " eth0", 6) == 0) { bytedata[0] = strtok(cmdline, " "); while(bytedata[i] != NULL) { i++; bytedata[i] = strtok(NULL, " "); } bytedata[i + 1] = '\0'; currentbytes = atoi(bytedata[9]); } } i = 0; rewind(f); elapsedbytes = currentbytes - prevbytes; prevbytes = currentbytes; byteusage = 8 * (elapsedbytes / 1024); //printf("usage:%lld\n",byteusage); if(ii & 0x40) { SLEEP += (maxthrottle - byteusage) * -1.1;//-2.5; if(SLEEP < 0){ SLEEP = 0; } //printf("sleep:%d\n", SLEEP); } usleep(25000); ii++; } } return NULL; } SLEEP and UPLOAD_SPEED are global variables and UPLOAD_SPEED is in kb/s and generated via a speedtest function that gets the upload speed of my computer. this function is running inside a POSIX thread updating SLEEP which my threads doing the socket work grab to sleep by after every packet. as testing instead of only doing the ports i want to check i make it do all the ports over and over again so i can run dstat on a machine to check bandwidth and no matter what bandwidth.throttle is set to it always seems to generate the same amount of bandwidth to the dstat machine. the way i calculate how much i "should" throttle by is by finding the maximum throttle speed which is defined as maxthrottle = upload_speed * throttle / 100; for example if my upload speed was 1000kb/s and my throttle was 90 (90%) my max throttle would be 900kb/s from there it would find the current bytes sent from /proc/net/dev and then find my sleep time via incrementing or decrementing it via sleep += (maxthrottle - bytesysed) * -1.1; this should in theory increase or decrease the sleep time based on how many bytes used there are. the if(ii & 0x40) statement is just for some moderation control. it makes it so it only sets sleep to a new time every 30-40 iterations. final notes: the main problem is that the sleep timer does not seem to modify the speed of packets being set. or maybe its just my implementation because on a freshly restarted machine where /proc/net/dev has low numbers of bytes sent it seems to raise the sleep timer accordingly on my 60kb/s upload machine (ex if i set the throttle to 2 it will incline the sleep timer until network bandwidth out reaches the max bandwidth threshold, but when i try running it on a server which as been online forever it doesnt seem to work as nicely if at all. if anyone can suggest a new method of monitoring the network to adjust a sleep delay then let me know or if anyone sees a flaw in my code. thank you.

    Read the article

  • Forking with a listening socket

    - by viraptor
    I'd like to make sure about the correctness of the way I try to use accept() on a socket. I know that in Linux it's safe to listen() on a socket, fork() N children and then recv() the packets in all of them without any synchronisation from the user side (the packets get more or less load-balanced between the children). But that's UDP. Does the same property hold for TCP and listen(), fork(), accept()? Can I just assume that it's ok to accept on a shared socket created by the parent, even when other children do the same? Is POSIX, BSD sockets or any other standard defining it somewhere?

    Read the article

  • Linux pptp client stops working after several hours

    - by Aron Rotteveel
    Here's the situation: Setup: 1 Windows Server 2008 machine acting as a Domain Controller and RRAS server 1 CentOS machine in a datacentre located elsewhere PPTP client running on CentOS machine, connected to the DC via When I connect to the DC, everything is working fine. I have set up a static IP for the dialup connection in my RRAS server so that the CentOS machine is automatically assigned the IP 192.168.1.240. Inside the VPN, it is not possible to access this machine on the local IP-address. Perfect. However, after several hours, it simply seems to stop working (IE: I cannot ping to or from this machine on the local network). The strange thing is, however: The DC shows the VPN client as still being connected The CentOS machine shows the network interface as being up There are no entries in my /var/log/messages that indicate a problem Output from ifconfig: ppp0 Link encap:Point-to-Point Protocol inet addr:192.168.1.240 P-t-P:192.168.1.160 Mask:255.255.255.255 UP POINTOPOINT RUNNING NOARP MULTICAST MTU:1396 Metric:1 RX packets:43 errors:0 dropped:0 overruns:0 frame:0 TX packets:58 errors:0 dropped:0 overruns:0 carrier:0 collisions:0 txqueuelen:3 RX bytes:4511 (4.4 KiB) TX bytes:15071 (14.7 KiB) Output from route -n: 192.168.1.160 0.0.0.0 255.255.255.255 UH 0 0 0 ppp0 192.168.1.0 0.0.0.0 255.255.255.0 U 0 0 0 ppp0 I have the following in my ip-up.local: route add -net 192.168.1.0 netmask 255.255.255.0 dev ppp0 The situation can be easily fixed by issueing a killall pppd and re-connecting. However, I obviously do not want to do this every X-hours or so. I have tried running pppd with both the debug as the kdebug flag but cannot find the cause of this problem. Currently, my ppp0 network interface seems to be running and the last log lines mentioning it are: Feb 19 14:10:40 graviton pppd[10934]: local IP address 192.168.1.240 Feb 19 14:10:40 graviton pppd[10934]: remote IP address 192.168.1.160 Feb 19 14:10:40 graviton pppd[10934]: Script /etc/ppp/ip-up started (pid 10952) Feb 19 14:10:40 graviton pppd[10934]: Script /etc/ppp/ip-up finished (pid 10952), status = 0x0 Feb 19 14:11:27 graviton pptp[10935]: anon log[decaps_gre:pptp_gre.c:414]: buffering packet 190 (expecting 189, lost or reordered) Feb 19 14:11:37 graviton pptp[10942]: anon log[logecho:pptp_ctrl.c:677]: Echo Request received. Feb 19 14:11:37 graviton pptp[10942]: anon log[ctrlp_rep:pptp_ctrl.c:251]: Sent control packet type is 6 'Echo-Reply' Feb 19 14:12:37 graviton pptp[10942]: anon log[logecho:pptp_ctrl.c:677]: Echo Request received. Feb 19 14:12:37 graviton pptp[10942]: anon log[ctrlp_rep:pptp_ctrl.c:251]: Sent control packet type is 6 'Echo-Reply' Feb 19 14:12:37 graviton pptp[10942]: anon log[logecho:pptp_ctrl.c:677]: Echo Reply received. Feb 19 14:13:37 graviton pptp[10942]: anon log[logecho:pptp_ctrl.c:677]: Echo Reply received. Feb 19 14:14:37 graviton pptp[10942]: anon log[logecho:pptp_ctrl.c:677]: Echo Reply received. Feb 19 14:15:37 graviton pptp[10942]: anon log[logecho:pptp_ctrl.c:677]: Echo Reply received. Feb 19 14:16:37 graviton pptp[10942]: anon log[logecho:pptp_ctrl.c:677]: Echo Reply received. Feb 19 14:19:37 graviton pptp[10942]: anon log[logecho:pptp_ctrl.c:677]: Echo Reply received. Feb 19 14:19:37 graviton pptp[10942]: anon log[logecho:pptp_ctrl.c:679]: no more Echo Reply/Request packets will be reported. I have enabled the persist option. The network interface is still running, but it is still impossible to send data through the VPN. Any help is appreciated.

    Read the article

  • Trouble with site-to-site OpenVPN & pfSense not passing traffic

    - by JohnCC
    I'm trying to get an OpenVPN tunnel going on pfSense 1.2.3-RELEASE running on embedded routers. I have a local LAN 10.34.43.0/254. The remote LAN is 10.200.1.0/24. The local pfSense is configured as the client, and the remote is configured as the server. My OpenVPN tunnel is using the IP range 10.99.89.0/24 internally. There are also some additional LANs on the remote side routed through the tunnel, but the issue is not with those since my connectivity fails before that point in the chain. The tunnel comes up fine and the logs look healthy. What I find is this:- I can ping and telnet to the remote LAN and the additional remote LANs from the local pfSense box's shell. I cannot ping or telnet to any remote LANs from the local network. I cannot ping or telnet to the local network from the remote LAN or the remote pfSense box's shell. If I tcpdump the tun interfaces on both sides and ping from the local LAN, I see the packets hit the tunnel locally, but they do not appear on the remote side (nor do they appear on the remote LAN interface if I tcpdump that). If I tcpdump the tun interfaces on both sides and ping from the local pfSense shell, I see the packets hit the tunnel locally, and exit the remote side. I can also tcpdump the remote LAN interface and see them pass there too. If I tcpdump the tun interfaces on both sides and ping from the remote pfSense shell, I see the packets hit the remote tun but they do not emerge from the local one. Here is the config file the remote side is using:- #user nobody #group nobody daemon keepalive 10 60 ping-timer-rem persist-tun persist-key dev tun proto udp cipher BF-CBC up /etc/rc.filter_configure down /etc/rc.filter_configure server 10.99.89.0 255.255.255.0 client-config-dir /var/etc/openvpn_csc push "route 10.200.1.0 255.255.255.0" lport <port> route 10.34.43.0 255.255.255.0 ca /var/etc/openvpn_server0.ca cert /var/etc/openvpn_server0.cert key /var/etc/openvpn_server0.key dh /var/etc/openvpn_server0.dh comp-lzo push "route 205.217.5.128 255.255.255.224" push "route 205.217.5.64 255.255.255.224" push "route 165.193.147.128 255.255.255.224" push "route 165.193.147.32 255.255.255.240" push "route 192.168.1.16 255.255.255.240" push "route 192.168.2.16 255.255.255.240" Here is the local config:- writepid /var/run/openvpn_client0.pid #user nobody #group nobody daemon keepalive 10 60 ping-timer-rem persist-tun persist-key dev tun proto udp cipher BF-CBC up /etc/rc.filter_configure down /etc/rc.filter_configure remote <host> <port> client lport 1194 ifconfig 10.99.89.2 10.99.89.1 ca /var/etc/openvpn_client0.ca cert /var/etc/openvpn_client0.cert key /var/etc/openvpn_client0.key comp-lzo You can see the relevant parts of the routing tables extracted from pfSense here http://pastie.org/5365800 The local firewall permits all ICMP from the LAN, and my PC is allowed everything to anywhere. The remote firewall treats its LAN as trusted and permits all traffic on that interface. Can anyone suggest why this is not working, and what I could try next?

    Read the article

< Previous Page | 24 25 26 27 28 29 30 31 32 33 34 35  | Next Page >