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  • C# listview add items

    - by Eyla
    Greetings, I have a method to capture packets. after capturing I want to add each packet as a row in listview in runting time so as soon as I capture packet I want to add it to the list view. the problem that when I use items.Add() I will get overload argument error. please advice!! here is my code: private void packetCapturingThreadMethod() { Packet packet = null; int countOfPacketCaptures = 0; while ((packet = device.GetNextPacket()) != null) { packet = device.GetNextPacket(); if (packet is TCPPacket) { TCPPacket tcp = (TCPPacket)packet; myPacket tempPacket = new myPacket(); tempPacket.packetType = "TCP"; tempPacket.sourceAddress = Convert.ToString(tcp.SourceAddress); tempPacket.destinationAddress = Convert.ToString(tcp.DestinationAddress); tempPacket.sourcePort = Convert.ToString(tcp.SourcePort); tempPacket.destinationPort = Convert.ToString(tcp.DestinationPort); tempPacket.packetMessage = Convert.ToString(tcp.Data); packetsList.Add(tempPacket); string[] row = { packetsList[countOfPacketCaptures].packetType, packetsList[countOfPacketCaptures].sourceAddress, packetsList[countOfPacketCaptures].destinationAddress, packetsList[countOfPacketCaptures].sourcePort, packetsList[countOfPacketCaptures].destinationPort, packetsList[countOfPacketCaptures].packetMessage }; try { listView1.Items.Add(packetsList[countOfPacketCaptures].packetType, packetsList[countOfPacketCaptures].sourceAddress, packetsList[countOfPacketCaptures].destinationAddress, packetsList[countOfPacketCaptures].sourcePort, packetsList[countOfPacketCaptures].destinationPort, packetsList[countOfPacketCaptures].packetMessage) ; countOfPacketCaptures++; lblCapturesLabels.Text = Convert.ToString(countOfPacketCaptures);} catch (Exception e) { } } else if (packet is UDPPacket) { UDPPacket udp = (UDPPacket)packet; myPacket tempPacket = new myPacket(); tempPacket.packetType = "UDP"; tempPacket.sourceAddress = Convert.ToString(udp.SourceAddress); tempPacket.destinationAddress = Convert.ToString(udp.DestinationAddress); tempPacket.sourcePort = Convert.ToString(udp.SourcePort); tempPacket.destinationPort = Convert.ToString(udp.DestinationPort); tempPacket.packetMessage = Convert.ToString(udp.Data); packetsList.Add(tempPacket); string[] row = { packetsList[countOfPacketCaptures].packetType, packetsList[countOfPacketCaptures].sourceAddress, packetsList[countOfPacketCaptures].destinationAddress, packetsList[countOfPacketCaptures].sourcePort, packetsList[countOfPacketCaptures].destinationPort, packetsList[countOfPacketCaptures].packetMessage }; try { dgwPacketInfo.Rows.Add(row); countOfPacketCaptures++; lblCapturesLabels.Text = Convert.ToString(countOfPacketCaptures); } catch (Exception e) { } } } }

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  • LSP packet modify

    - by kellogs
    Hello, anybody care to share some insights on how to use LSP for packet modifying ? I am using the non IFS subtype and I can see how (pseudo?) packets first enter WSPRecv. But how do I modify them ? My inquiry is about one single HTTP response that causes WSPRecv to be called 3 times :((. I need to modify several parts of this response, but since it comes in 3 slices, it is pretty hard to modify it accordingly. And, maybe on other machines or under different conditions (such as high traffic) there would only be one sole WSPRecv call, or maybe 10 calls. What is the best way to work arround this (please no NDIS :D), and how to properly change the buffer (lpBuffers-buf) by increasing it ? int WSPAPI WSPRecv( SOCKET s, LPWSABUF lpBuffers, DWORD dwBufferCount, LPDWORD lpNumberOfBytesRecvd, LPDWORD lpFlags, LPWSAOVERLAPPED lpOverlapped, LPWSAOVERLAPPED_COMPLETION_ROUTINE lpCompletionRoutine, LPWSATHREADID lpThreadId, LPINT lpErrno ) { LPWSAOVERLAPPEDPLUS ProviderOverlapped = NULL; SOCK_INFO *SocketContext = NULL; int ret = SOCKET_ERROR; *lpErrno = NO_ERROR; // // Find our provider socket corresponding to this one // SocketContext = FindAndRefSocketContext(s, lpErrno); if ( NULL == SocketContext ) { dbgprint( "WSPRecv: FindAndRefSocketContext failed!" ); goto cleanup; } // // Check for overlapped I/O // if ( NULL != lpOverlapped ) { /*bla bla .. not interesting in my case*/ } else { ASSERT( SocketContext->Provider->NextProcTable.lpWSPRecv ); SetBlockingProvider(SocketContext->Provider); ret = SocketContext->Provider->NextProcTable.lpWSPRecv( SocketContext->ProviderSocket, lpBuffers, dwBufferCount, lpNumberOfBytesRecvd, lpFlags, lpOverlapped, lpCompletionRoutine, lpThreadId, lpErrno); SetBlockingProvider(NULL); //is this the place to modify packet length and contents ? if (strstr(lpBuffers->buf, "var mapObj = null;")) { int nLen = strlen(lpBuffers->buf) + 200; /*CHAR *szNewBuf = new CHAR[]; CHAR *pIndex; pIndex = strstr(lpBuffers->buf, "var mapObj = null;"); nLen = strlen(strncpy(szNewBuf, lpBuffers->buf, (pIndex - lpBuffers->buf) * sizeof (CHAR))); nLen = strlen(strncpy(szNewBuf + nLen * sizeof(CHAR), "var com = null;\r\n", 17 * sizeof(CHAR))); pIndex += 18 * sizeof(CHAR); nLen = strlen(strncpy(szNewBuf + nLen * sizeof(CHAR), pIndex, 1330 * sizeof (CHAR))); nLen = strlen(strncpy(szNewBuf + nLen * sizeof(CHAR), "if (com == null)\r\n" \ "com = new ActiveXObject(\"InterCommJS.Gateway\");\r\n" \ "com.lat = latitude;\r\n" \ "com.lon = longitude;\r\n}", 111 * sizeof (CHAR))); pIndex = strstr(szNewBuf, "Content-Length:"); pIndex += 16 * sizeof(CHAR); strncpy(pIndex, "1465", 4 * sizeof(CHAR)); lpBuffers->buf = szNewBuf; lpBuffers->len += 128;*/ } if ( SOCKET_ERROR != ret ) { SocketContext->BytesRecv += *lpNumberOfBytesRecvd; } } cleanup: if ( NULL != SocketContext ) DerefSocketContext( SocketContext, lpErrno ); return ret; } Thank you

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  • Tools to help with analysing log files

    - by peter
    I am developing a C# .NET application. In the app.config file I add trace logging as shown, <?xml version="1.0" encoding="UTF-8" ?> <configuration> <system.diagnostics> <trace autoflush="true" /> <sources> <source name="System.Net.Sockets" maxdatasize="1024"> <listeners> <add name="MyTraceFile"/> </listeners> </source> </sources> <sharedListeners> <add name="MyTraceFile" type="System.Diagnostics.TextWriterTraceListener" initializeData="System.Net.trace.log" /> </sharedListeners> <switches> <add name="System.Net" value="Verbose" /> </switches> </system.diagnostics> </configuration> Are there any good tools around to analyse the log file that is output? The output looks like this, System.Net.Sockets Verbose: 0 : [5900] Data from Socket#8764489::Send DateTime=2010-04-07T03:22:40.1067012Z System.Net.Sockets Verbose: 0 : [5900] 00000000 : 4D 49 4D 45 2D 56 65 72-73 69 6F 6E 3A 20 31 2E : MIME-Version: 1. DateTime=2010-04-07T03:22:40.1067012Z System.Net.Sockets Verbose: 0 : [5900] 00000060 : 65 3A 20 37 20 41 70 72-20 32 30 31 30 20 31 35 : e: 7 Apr 2010 15 DateTime=2010-04-07T03:22:40.1067012Z System.Net.Sockets Verbose: 0 : [5900] 00000070 : 3A 32 32 3A 34 30 20 2B-31 32 30 30 0D 0A 53 75 : :22:40 +1200..Su DateTime=2010-04-07T03:22:40.1067012Z System.Net.Sockets Verbose: 0 : [5900] 00000080 : 62 6A 65 63 74 3A 20 5B-45 72 72 6F 72 5D 20 45 : bject: [Error] E DateTime=2010-04-07T03:22:40.1067012Z System.Net.Sockets Verbose: 0 : [5900] 00000090 : 78 63 65 70 74 69 6F 6E-20 69 6E 20 53 79 6E 63 : xception in Sync DateTime=2010-04-07T03:22:40.1067012Z System.Net.Sockets Verbose: 0 : [5900] 000000A0 : 53 65 72 76 69 63 65 20-28 32 30 30 38 2E 30 2E : Service (2008.0. DateTime=2010-04-07T03:22:40.1067012Z System.Net.Sockets Verbose: 0 : [5900] 000000B0 : 33 30 34 2E 31 32 33 34-32 29 0D 0A 43 6F 6E 74 : 304.12342)..Cont DateTime=2010-04-07T03:22:40.1067012Z Is there anything that can take the output shown above (my output is a text file 100mb in size), group together packets, and help out with finding particular issues I would like to hear about it. Thanks.

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  • architecture python question

    - by tom smith
    hi. creating a distributed crawling python app. it consists of a master server, and associated client apps that will run on client servers. the purpose of the client app is to run across a targeted site, to extract specific data. the clients need to go "deep" within the site, behind multiple levels of forms, so each client is specifically geared towards a given site. each client app looks something like main: parse initial url call function level1 (data1) function level1 (data) parse the url, for data1 use the required xpath to get the dom elements call the next function call level2 (data) function level2 (data2) parse the url, for data2 use the required xpath to get the dom elements call the next function call level3 function level3 (dat3) parse the url, for data3 use the required xpath to get the dom elements call the next function call level4 function level4 (data) parse the url, for data4 use the required xpath to get the dom elements at the final function.. --all the data output, and eventually returned to the server --at this point the data has elements from each function... my question: given that the number of calls that is made to the child function by the current function varies, i'm trying to figure out the best approach. each function essentialy fetches a page of content, and then parses the page using a number of different XPath expressions, combined with different regex expressions depending on the site/page. if i run a client on a single box, as a sequential process, it'll take awhile, but the load on the box is rather small. i've thought of attempting to implement the child functions as threads from the current function, but that could be a nightmare, as well as quickly bring the "box" to its knees! i've thought of breaking the app up in a manner that would allow the master to essentially pass packets to the client boxes, in a way to allow each client/function to be run directly from the master. this process requires a bit of rewrite, but it has a number of advantages. a bunch of redundancy, and speed. it would detect if a section of the process was crashing and restart from that point. but not sure if it would be any faster... i'm writing the parsing scripts in python.. so... any thoughts/comments would be appreciated... i can get into a great deal more detail, but didn't want to bore anyone!! thanks! tom

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  • Tools to Help out with

    - by peter
    I am developing a C# .NET application. In the app.config file I add trace logging as shown, <?xml version="1.0" encoding="UTF-8" ?> <configuration> <system.diagnostics> <trace autoflush="true" /> <sources> <source name="System.Net.Sockets" maxdatasize="1024"> <listeners> <add name="MyTraceFile"/> </listeners> </source> </sources> <sharedListeners> <add name="MyTraceFile" type="System.Diagnostics.TextWriterTraceListener" initializeData="System.Net.trace.log" /> </sharedListeners> <switches> <add name="System.Net" value="Verbose" /> </switches> </system.diagnostics> </configuration> Are there any good tools around to analyse the log file that is output? The output looks like this, System.Net.Sockets Verbose: 0 : [5900] Data from Socket#8764489::Send DateTime=2010-04-07T03:22:40.1067012Z System.Net.Sockets Verbose: 0 : [5900] 00000000 : 4D 49 4D 45 2D 56 65 72-73 69 6F 6E 3A 20 31 2E : MIME-Version: 1. DateTime=2010-04-07T03:22:40.1067012Z System.Net.Sockets Verbose: 0 : [5900] 00000060 : 65 3A 20 37 20 41 70 72-20 32 30 31 30 20 31 35 : e: 7 Apr 2010 15 DateTime=2010-04-07T03:22:40.1067012Z System.Net.Sockets Verbose: 0 : [5900] 00000070 : 3A 32 32 3A 34 30 20 2B-31 32 30 30 0D 0A 53 75 : :22:40 +1200..Su DateTime=2010-04-07T03:22:40.1067012Z System.Net.Sockets Verbose: 0 : [5900] 00000080 : 62 6A 65 63 74 3A 20 5B-45 72 72 6F 72 5D 20 45 : bject: [Error] E DateTime=2010-04-07T03:22:40.1067012Z System.Net.Sockets Verbose: 0 : [5900] 00000090 : 78 63 65 70 74 69 6F 6E-20 69 6E 20 53 79 6E 63 : xception in Sync DateTime=2010-04-07T03:22:40.1067012Z System.Net.Sockets Verbose: 0 : [5900] 000000A0 : 53 65 72 76 69 63 65 20-28 32 30 30 38 2E 30 2E : Service (2008.0. DateTime=2010-04-07T03:22:40.1067012Z System.Net.Sockets Verbose: 0 : [5900] 000000B0 : 33 30 34 2E 31 32 33 34-32 29 0D 0A 43 6F 6E 74 : 304.12342)..Cont DateTime=2010-04-07T03:22:40.1067012Z Is there anything that can take the output shown above (my output is a text file 100mb in size), group together packets, and help out with finding particular issues I would like to hear about it. Thanks.

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  • Anybody Know of any Tools to help Analysing .NET Trace Log Files?

    - by peter
    I am developing a C# .NET application. In the app.config file I add trace logging as shown, <?xml version="1.0" encoding="UTF-8" ?> <configuration> <system.diagnostics> <trace autoflush="true" /> <sources> <source name="System.Net.Sockets" maxdatasize="1024"> <listeners> <add name="MyTraceFile"/> </listeners> </source> </sources> <sharedListeners> <add name="MyTraceFile" type="System.Diagnostics.TextWriterTraceListener" initializeData="System.Net.trace.log" /> </sharedListeners> <switches> <add name="System.Net" value="Verbose" /> </switches> </system.diagnostics> </configuration> Are there any good tools around to analyse the log file that is output? The output looks like this, System.Net.Sockets Verbose: 0 : [5900] Data from Socket#8764489::Send DateTime=2010-04-07T03:22:40.1067012Z System.Net.Sockets Verbose: 0 : [5900] 00000000 : 4D 49 4D 45 2D 56 65 72-73 69 6F 6E 3A 20 31 2E : MIME-Version: 1. DateTime=2010-04-07T03:22:40.1067012Z System.Net.Sockets Verbose: 0 : [5900] 00000060 : 65 3A 20 37 20 41 70 72-20 32 30 31 30 20 31 35 : e: 7 Apr 2010 15 DateTime=2010-04-07T03:22:40.1067012Z System.Net.Sockets Verbose: 0 : [5900] 00000070 : 3A 32 32 3A 34 30 20 2B-31 32 30 30 0D 0A 53 75 : :22:40 +1200..Su DateTime=2010-04-07T03:22:40.1067012Z System.Net.Sockets Verbose: 0 : [5900] 00000080 : 62 6A 65 63 74 3A 20 5B-45 72 72 6F 72 5D 20 45 : bject: [Error] E DateTime=2010-04-07T03:22:40.1067012Z System.Net.Sockets Verbose: 0 : [5900] 00000090 : 78 63 65 70 74 69 6F 6E-20 69 6E 20 53 79 6E 63 : xception in Sync DateTime=2010-04-07T03:22:40.1067012Z System.Net.Sockets Verbose: 0 : [5900] 000000A0 : 53 65 72 76 69 63 65 20-28 32 30 30 38 2E 30 2E : Service (2008.0. DateTime=2010-04-07T03:22:40.1067012Z System.Net.Sockets Verbose: 0 : [5900] 000000B0 : 33 30 34 2E 31 32 33 34-32 29 0D 0A 43 6F 6E 74 : 304.12342)..Cont DateTime=2010-04-07T03:22:40.1067012Z Is there anything that can take the output shown above (my output is a text file 100mb in size), group together packets, and help out with finding particular issues I would like to hear about it. Thanks.

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  • Python and C++ Sockets converting packet data

    - by yeus
    First of all, to clarify my goal: There exist two programs written in C in our laboratory. I am working on a Proxy Server (bidirectional) for them (which will also mainpulate the data). And I want to write that proxy server in Python. It is important to know that I know close to nothing about these two programs, I only know the definition file of the packets. Now: assuming a packet definition in one of the C++ programs reads like this: unsigned char Packet[0x32]; // Packet[Length] int z=0; Packet[0]=0x00; // Spare Packet[1]=0x32; // Length Packet[2]=0x01; // Source Packet[3]=0x02; // Destination Packet[4]=0x01; // ID Packet[5]=0x00; // Spare for(z=0;z<=24;z+=8) { Packet[9-z/8]=((int)(720000+armcontrolpacket->dof0_rot*1000)/(int)pow((double)2,(double)z)); Packet[13-z/8]=((int)(720000+armcontrolpacket->dof0_speed*1000)/(int)pow((double)2,(double)z)); Packet[17-z/8]=((int)(720000+armcontrolpacket->dof1_rot*1000)/(int)pow((double)2,(double)z)); Packet[21-z/8]=((int)(720000+armcontrolpacket->dof1_speed*1000)/(int)pow((double)2,(double)z)); Packet[25-z/8]=((int)(720000+armcontrolpacket->dof2_rot*1000)/(int)pow((double)2,(double)z)); Packet[29-z/8]=((int)(720000+armcontrolpacket->dof2_speed*1000)/(int)pow((double)2,(double)z)); Packet[33-z/8]=((int)(720000+armcontrolpacket->dof3_rot*1000)/(int)pow((double)2,(double)z)); Packet[37-z/8]=((int)(720000+armcontrolpacket->dof3_speed*1000)/(int)pow((double)2,(double)z)); Packet[41-z/8]=((int)(720000+armcontrolpacket->dof4_rot*1000)/(int)pow((double)2,(double)z)); Packet[45-z/8]=((int)(720000+armcontrolpacket->dof4_speed*1000)/(int)pow((double)2,(double)z)); Packet[49-z/8]=((int)armcontrolpacket->timestamp/(int)pow(2.0,(double)z)); } if(SendPacket(sock,(char*)&Packet,sizeof(Packet))) return 1; return 0; What would be the easiest way to receive that data, convert it into a readable python format, manipulate them and send them forward to the receiver?

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  • How to stream semi-live audio over internet

    - by Thomas Tempelmann
    I want to write something like Skype, i.e. I have a constant audio stream on one computer and then recompress it in a format that's suitable for a latent internet connection, receive it on the other end and play it. Let's also assume that the internet connection is fairly modern and fast, i.e. DSL or alike, no slow connections over phone and such. The involved computers will also be rather modern (Dual Core Intel CPUs at 2GHz or more). I know how to handle the audio on the machines. What I don't know is how to transmit the audio in an efficient way. The challenges are: I'd like get good audio quality across the line. The stream should be received without drops. The stream may, however, be received with a little delay (a second delay is acceptable). I imagine that the transport software could first determine the average (and max) latency, then start the stream and tell the receiver to wait for that max latency before starting to play the audio. With that, if the latency doesn't get any higher, the entire stream will be playable on the other side without stutter or drops. If, due to unexpected IP latencies or blockages, the stream does get cut off, I want to be able to notice this so that I can take actions (e.g. abort the stream) and eventually start a new transmission. What are my options if I want do use ready-made software for the compression and tranmission? I have no intention to write my own audio compression engine, really. OTOH, I plan to sell the solution in a vertical market, meaning I can afford a few dollars of license fees per copy, but not $100s. I guess the simplest solution would be to just open a TCP stream, send a few packets back and forth to determine their running time (or even use UDP for that), then use the results as the guide for my max latency value, then simply fire the audio data in its raw form (uncompressed 16 bit stereo), along with a timing code over the TCP connection. The receiver reads the data and plays it with the pre-determined delay. That might just work with the type of fast connection I expect. I just wonder if there are better solutions to reach this goal, with better performance (lower latency) and less data (compressed). BTW, I first try to implement this on OS X, but might want to do it on Windows, too, if it proves successful.

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  • Monitoring UDP socket in glib(mm) eats up CPU time

    - by Gyorgy Szekely
    Hi, I have a GTKmm Windows application (built with MinGW) that receives UDP packets (no sending). The socket is native winsock and I use glibmm IOChannel to connect it to the application main loop. The socket is read with recvfrom. My problem is: this setup eats 25% percent CPU time on a 3GHz workstation. Can somebody tell me why? The application is idle in this case, and if I remove the UDP code, CPU usage drops down to almost zero. As the application has to perform some CPU intensive tasks, I could image better ways to spend that 25% Here are some code excerpts: (sorry for the printf's ;) ) /* bind */ void UDPInterface::bindToPort(unsigned short port) { struct sockaddr_in target; WSADATA wsaData; target.sin_family = AF_INET; target.sin_port = htons(port); target.sin_addr.s_addr = 0; if ( WSAStartup ( 0x0202, &wsaData ) ) { printf("WSAStartup failed!\n"); exit(0); // :) WSACleanup(); } sock = socket( AF_INET, SOCK_DGRAM, 0 ); if (sock == INVALID_SOCKET) { printf("invalid socket!\n"); exit(0); } if (bind(sock,(struct sockaddr*) &target, sizeof(struct sockaddr_in) ) == SOCKET_ERROR) { printf("failed to bind to port!\n"); exit(0); } printf("[UDPInterface::bindToPort] listening on port %i\n", port); } /* read */ bool UDPInterface::UDPEvent(Glib::IOCondition io_condition) { recvfrom(sock, (char*)buf, BUF_SIZE*4, 0, NULL, NULL); /* process packet... */ } /* glibmm connect */ Glib::RefPtr channel = Glib::IOChannel::create_from_win32_socket(udp.sock); Glib::signal_io().connect( sigc::mem_fun(udp, &UDPInterface::UDPEvent), channel, Glib::IO_IN ); I've read here in some other question, and also in glib docs (g_io_channel_win32_new_socket()) that the socket is put into nonblocking mode, and it's "a side-effect of the implementation and unavoidable". Does this explain the CPU effect, it's not clear to me? Whether or not I use glib to access the socket or call recvfrom() directly doesn't seem to make much difference, since CPU is used up before any packet arrives and the read handler gets invoked. Also glibmm docs state that it's ok to call recvfrom() even if the socket is polled (Glib::IOChannel::create_from_win32_socket()) I've tried compiling the program with -pg and created a per function cpu usage report with gprof. This wasn't usefull because the time is not spent in my program, but in some external glib/glibmm dll.

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  • IEnumerable<T> ToArray usage, is it a copy or a pointer?

    - by Daniel
    I am parsing an arbitrary length byte array that is going to be passed around to a few different layers of parsing. Each parser creates a Header and a Packet payload just like any ordinary encapsulation. And my problem lies in how the encapsulation holds its packet byte array payload. Say i have a 100 byte array, and it has 3 levels of encapsulation. 3 packet objects will be created and i want to set the payload of these packets to the corresponding position in the byte array of the packet. For example lets say the payload size is 20 for all levels, then imagine it has a public byte[] Payload on each object. However the problem is that this byte[] Payload is a copy of the original 100 bytes. So i'm going to end up with 160 bytes in memory instead of 100. If it were in c++ i could just easily use a pointer however i'm writing this in c#. So i created the following class: public class PayloadSegment<T> : IEnumerable<T> { public readonly T[] Array; public readonly int Offset; public readonly int Count; public PayloadSegment(T[] array, int offset, int count) { this.Array = array; this.Offset = offset; this.Count = count; } public T this[int index] { get { if (index < 0 || index >= this.Count) throw new IndexOutOfRangeException(); else return Array[Offset + index]; } set { if (index < 0 || index >= this.Count) throw new IndexOutOfRangeException(); else Array[Offset + index] = value; } } public IEnumerator<T> GetEnumerator() { for (int i = Offset; i < Offset + Count; i++) yield return Array[i]; } System.Collections.IEnumerator System.Collections.IEnumerable.GetEnumerator() { IEnumerator<T> enumerator = this.GetEnumerator(); while (enumerator.MoveNext()) { yield return enumerator.Current; } } } This way i can simply reference a position inside the original byte array but use positional indexing. However if i do something like: PayloadSegment<byte> something = new PayloadSegment<byte>(someArray, 5, 10); byte[] somethingArray = something.ToArray(); Will the somethingArray be a copy of the bytes, or a reference to the original PayloadSegment which in turn is a reference to the original byte array? Sorry it was hard to word this lol _<

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  • Correct usage of socket_select().

    - by Mark Tomlin
    What is the correct way to use socket_select within PHP to send and receive data? I have a connection to the server that allows for both TCP & UDP packet connections, I am utilizing both. Within these connections I'm both sending and receiving packets on the same port, but the TCP packet will be sent on one port (29999) and UDP will be sent on another port (30000). The transmission type will be that of AF_INET. The IP address will be loopback 127.0.0.1. I have many questions on how to create a socket connection within this scenario. For example, is it better to use socket_create_pair to make the connection, or use just socket_create followed by socket_connect, and then implement socket_select? There is a chance that no data will be sent from the server to the client, and it is up to the client to maintain the connection. This will be done by utilizing the time out function within the socket_select call. Should no data be sent within the time limit, the socket_select function will break and a keep alive packet can then be sent. The following script is of the client. // Create $TCP = socket_create(AF_INET, SOCK_STREAM, SOL_TCP); $UDP = socket_create(AF_INET, SOCK_DGRAM, SOL_UDP); // Misc $isAlive = TRUE; $UDPPort = 30000; define('ISP_ISI', 1); // Connect socket_connect($TCP, '127.0.0.1', 29999); socket_connect($UDP, '127.0.0.1', $UDPPort); // Construct Parameters $recv = array($TCP, $UDP); $null = NULL; // Make The Packet to Send. $packet = pack('CCCxSSxCSa16a16', 44, ISP_ISI, 1, $UDPPort, 0, '!', 0, 'AdminPass', 'SocketSelect'); // Send ISI (InSim Init) Packet socket_write($TCP, $packet); /* Main Program Loop */ while ($isAlive == TRUE) { // Socket Select $sock = socket_select($recv, $null, $null, 5); // Check Status if ($sock === FALSE) $isAlive = FALSE; # Error else if ($sock > 0) # How does one check to find what socket changed? else # Something else happed, don't know what as it's not in the documentation, Could this be our timeout getting tripped? }

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  • Displaying a notification when bluetooth is disconnected - Android

    - by Ryan T
    I am trying to create a program that will display a notification to the user if a Blue tooth device suddenly comes out of range from my Android device. I currently have the following code but no notification is displayed. I was wondering if it was possible I shouldn't use ACTION_ACL_DISCONNECTED because I believe the bluetooth stack would be expecting packets that state a disconnect is requested. My requirements state that the bluetooth device will disconnect without warning. Thank you for any assistance! BluetoothNotification.java: //This is where the notification is created. import android.app.Activity; import android.app.Notification; import android.app.NotificationManager; import android.app.PendingIntent; import android.content.Context; import android.content.Intent; import android.os.Bundle; import android.app.Activity; import android.app.Notification; import android.app.NotificationManager; import android.app.PendingIntent; import android.content.Context; import android.content.Intent; import android.os.Bundle; public class BluetoothNotification extends Activity { public static final int NOTIFICATION_ID = 1; /** Called when the activity is first created. */ @Override public void onCreate(Bundle savedInstanceState) { super.onCreate(savedInstanceState); setContentView(R.layout.main); /** Define configuration for our notification */ int icon = R.drawable.logo; CharSequence tickerText = "This is a sample notification"; long when = System.currentTimeMillis(); Context context = getApplicationContext(); CharSequence contentTitle = "Sample notification"; CharSequence contentText = "This notification has been generated as a result of BT Disconnecting"; Intent notificationIntent = new Intent(this, BluetoothNotification.class); PendingIntent contentIntent = PendingIntent.getActivity(this, 0, notificationIntent, 0); /** Initialize the Notification using the above configuration */ final Notification notification = new Notification(icon, tickerText, when); notification.setLatestEventInfo(context, contentTitle, contentText, contentIntent); /** Retrieve reference from NotificationManager */ String ns = Context.NOTIFICATION_SERVICE; final NotificationManager mNotificationManager = (NotificationManager) getSystemService(ns); mNotificationManager.notify(NOTIFICATION_ID, notification); finish(); } } Snippet from OnCreate: //Located in Controls.java IntentFilter filter1 = new IntentFilter(BluetoothDevice.ACTION_ACL_DISCONNECTED); this.registerReceiver(mReceiver, filter1); Snippet from Controls.java: private final BroadcastReceiver mReceiver = new BroadcastReceiver() { @Override public void onReceive(Context context, Intent intent) { String action = intent.getAction(); BluetoothDevice device = intent.getParcelableExtra(BluetoothDevice.EXTRA_DEVICE); if (BluetoothDevice.ACTION_ACL_DISCONNECTED.equals(action)) { //Device has disconnected NotificationManager nm = (NotificationManager) getSystemService(NOTIFICATION_SERVICE); } } };

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  • Abnormally disconnected TCP sockets and write timeout

    - by James
    Hello I will try to explain the problem in shortest possible words. I am using c++ builder 2010. I am using TIdTCPServer and sending voice packets to a list of connected clients. Everything works ok untill any client is disconnected abnormally, For example power failure etc. I can reproduce similar disconnect by cutting the ethernet connection of a connected client. So now we have a disconnected socket but as you know it is not yet detected at server side so server will continue to try to send data to that client too. But when server try to write data to that disconnected client ...... Write() or WriteLn() HANGS there in trying to write, It is like it is wating for somekind of Write timeout. This hangs the hole packet distribution process as a result creating a lag in data transmission to all other clients. After few seconds "Socket Connection Closed" Exception is raised and data flow continues. Here is the code try { EnterCriticalSection(&SlotListenersCriticalSection); for(int i=0;i<SlotListeners->Count;i++) { try { //Here the process will HANG for several seconds on a disconnected socket ((TIdContext*) SlotListeners->Objects[i])->Connection->IOHandler->WriteLn("Some DATA"); }catch(Exception &e) { SlotListeners->Delete(i); } } }__finally { LeaveCriticalSection(&SlotListenersCriticalSection); } Ok i already have a keep alive mechanism which disconnect the socket after n seconds of inactivity. But as you can imagine, still this mechnism cant sync exactly with this braodcasting loop because this braodcasting loop is running almost all the time. So is there any Write timeouts i can specify may be through iohandler or something ? I have seen many many threads about "Detecting disconnected tcp socket" but my problem is little different, i need to avoid that hangup for few seconds during the write attempt. So is there any solution ? Or should i consider using some different mechanism for such data broadcasting for example the broadcasting loop put the data packet in some kind of FIFO buffer and client threads continuously check for available data and pick and deliver it to themselves ? This way if one thread hangs it will not stop/delay the over all distribution thread. Any ideas please ? Thanks for your time and help. Regards Jams

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  • waiting for 2 different events in a single thread

    - by João Portela
    component A (in C++) - is blocked waiting for alarm signals (not relevant) and IO signals (1 udp socket). has one handler for each of these. component B (java) - has to receive the same information the component A udp socket receives. periodicaly gives instructions that should be sent through component A udp socket. How to join both components? it is strongly desirable that: the changes to attach component B to component A are minimal (its not my code and it is not very pleasent to mess with). the time taken by the new operations (usually communicating with component B) interfere very little with the usual processing time of component A - this means that if the operations are going to take a "some" time I would rather use a thread or something to do them. note: since component A receives udp packets more frequently that it has component B instructions to forward, if necessary, it can only forward the instructions (when available) from the IO handler. my initial ideia was to develop a component C (in C++) that would sit inside the component A code (is this called an adapter?) that when instanciated starts the java process and makes the necessary connections (that not so little overhead in the initialization is not a problem). It would have 2 stacks, one for the data to give component B (lets call it Bstack) and for the data to give component A (lets call it Astack). It would sit on its thread (lets call it new-thread) waiting for data to be available in Bstack to send it over udp, and listen on the udp socket to put data on the Astack. This means that the changes to component A are only: when it receives a new UDP packet put it on the Bstack, and if there is something on the Astack sent it over its UDP socket (I decided for this because this socket would only be used in the main thread). One of the problems is that I don't know how to wait for both of these events at the same time using only one thread. so my questions are: Do I really need to use the main thread to send the data over component A socket or can I do it from the new-thread? (I think the answer is no, but I'm not sure about race conditions on sockets) how to I wait for both events? boost::condition_variable or something similar seems the solution in the case of the stack and boost::asio::io_service io_service.run() seems like the thing to use for the socket. Is there any other alternative solution for this problem that I'm not aware of? Thanks for reading this long text but I really wanted you to understand the problem.

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  • How do you unit test the real world?

    - by Kim Sun-wu
    I'm primarily a C++ coder, and thus far, have managed without really writing tests for all of my code. I've decided this is a Bad Idea(tm), after adding new features that subtly broke old features, or, depending on how you wish to look at it, introduced some new "features" of their own. But, unit testing seems to be an extremely brittle mechanism. You can test for something in "perfect" conditions, but you don't get to see how your code performs when stuff breaks. A for instance is a crawler, let's say it crawls a few specific sites, for data X. Do you simply save sample pages, test against those, and hope that the sites never change? This would work fine as regression tests, but, what sort of tests would you write to constantly check those sites live and let you know when the application isn't doing it's job because the site changed something, that now causes your application to crash? Wouldn't you want your test suite to monitor the intent of the code? The above example is a bit contrived, and something I haven't run into (in case you haven't guessed). Let me pick something I have, though. How do you test an application will do its job in the face of a degraded network stack? That is, say you have a moderate amount of packet loss, for one reason or the other, and you have a function DoSomethingOverTheNetwork() which is supposed to degrade gracefully when the stack isn't performing as it's supposed to; but does it? The developer tests it personally by purposely setting up a gateway that drops packets to simulate a bad network when he first writes it. A few months later, someone checks in some code that modifies something subtly, so the degradation isn't detected in time, or, the application doesn't even recognize the degradation, this is never caught, because you can't run real world tests like this using unit tests, can you? Further, how about file corruption? Let's say you're storing a list of servers in a file, and the checksum looks okay, but the data isn't really. You want the code to handle that, you write some code that you think does that. How do you test that it does exactly that for the life of the application? Can you? Hence, brittleness. Unit tests seem to test the code only in perfect conditions(and this is promoted, with mock objects and such), not what they'll face in the wild. Don't get me wrong, I think unit tests are great, but a test suite composed only of them seems to be a smart way to introduce subtle bugs in your code while feeling overconfident about it's reliability. How do I address the above situations? If unit tests aren't the answer, what is? Thanks!

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  • How to convert m4a file to aac adts file in Xcode?

    - by Bird Hsuie
    I have a mp4 file copied from iPod lib and saved to my Document for my next step, I need it to convert to .mp3 or .aac(ADTS type) I use this code and failed... -(IBAction)compressFile:(id)sender{ NSLog (@"handleConvertToPCMTapped"); // open an ExtAudioFile NSLog (@"opening %@", exportURL); ExtAudioFileRef inputFile; CheckResult (ExtAudioFileOpenURL((__bridge CFURLRef)exportURL, &inputFile), "ExtAudioFileOpenURL failed"); // prepare to convert to a plain ol' PCM format AudioStreamBasicDescription myPCMFormat; myPCMFormat.mSampleRate = 44100; // todo: or use source rate? myPCMFormat.mFormatID = kAudioFormatMPEGLayer3 ; myPCMFormat.mFormatFlags = kAudioFormatFlagsCanonical; myPCMFormat.mChannelsPerFrame = 2; myPCMFormat.mFramesPerPacket = 1; myPCMFormat.mBitsPerChannel = 16; myPCMFormat.mBytesPerPacket = 4; myPCMFormat.mBytesPerFrame = 4; CheckResult (ExtAudioFileSetProperty(inputFile, kExtAudioFileProperty_ClientDataFormat, sizeof (myPCMFormat), &myPCMFormat), "ExtAudioFileSetProperty failed"); // allocate a big buffer. size can be arbitrary for ExtAudioFile. // you have 64 KB to spare, right? UInt32 outputBufferSize = 0x10000; void* ioBuf = malloc (outputBufferSize); UInt32 sizePerPacket = myPCMFormat.mBytesPerPacket; UInt32 packetsPerBuffer = outputBufferSize / sizePerPacket; // set up output file NSString *outputPath = [myDocumentsDirectory() stringByAppendingPathComponent:@"m_export.mp3"]; NSURL *outputURL = [NSURL fileURLWithPath:outputPath]; NSLog (@"creating output file %@", outputURL); AudioFileID outputFile; CheckResult(AudioFileCreateWithURL((__bridge CFURLRef)outputURL, kAudioFileCAFType, &myPCMFormat, kAudioFileFlags_EraseFile, &outputFile), "AudioFileCreateWithURL failed"); // start convertin' UInt32 outputFilePacketPosition = 0; //in bytes while (true) { // wrap the destination buffer in an AudioBufferList AudioBufferList convertedData; convertedData.mNumberBuffers = 1; convertedData.mBuffers[0].mNumberChannels = myPCMFormat.mChannelsPerFrame; convertedData.mBuffers[0].mDataByteSize = outputBufferSize; convertedData.mBuffers[0].mData = ioBuf; UInt32 frameCount = packetsPerBuffer; // read from the extaudiofile CheckResult (ExtAudioFileRead(inputFile, &frameCount, &convertedData), "Couldn't read from input file"); if (frameCount == 0) { printf ("done reading from file"); break; } // write the converted data to the output file CheckResult (AudioFileWritePackets(outputFile, false, frameCount, NULL, outputFilePacketPosition / myPCMFormat.mBytesPerPacket, &frameCount, convertedData.mBuffers[0].mData), "Couldn't write packets to file"); NSLog (@"Converted %ld bytes", outputFilePacketPosition); // advance the output file write location outputFilePacketPosition += (frameCount * myPCMFormat.mBytesPerPacket); } // clean up ExtAudioFileDispose(inputFile); AudioFileClose(outputFile); // show size in label NSLog (@"checking file at %@", outputPath); [self transMitFile:outputPath]; if ([[NSFileManager defaultManager] fileExistsAtPath:outputPath]) { NSError *fileManagerError = nil; unsigned long long fileSize = [[[NSFileManager defaultManager] attributesOfItemAtPath:outputPath error:&fileManagerError] fileSize]; } any suggestion?.......thanks for your great help!

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  • probelm with recv() on a tcp connection

    - by michael
    Hi, I am simulating TCP communication on windows in C I have sender and a receiver communicating. sender sends packets of specific size to receiver. receiver gets them and send an ACK for each packet it received back to the sender. If the sender didn't get a specific packet (they are numbered in a header inside the packet) it sends the packet again to the receiver. Here is the getPacket function on the receiver side: //get the next packet from the socket. set the packetSize to -1 //if it's the first packet. //return: total bytes read // return: 0 if socket has shutdown on sender side, -1 error, else number of bytes received int getPakcet(char *chunkBuff,int packetSize,SOCKET AcceptSocket){ int totalChunkLen = 0; int bytesRecv=-1; bool firstTime=false; if (packetSize==-1) { packetSize=MAX_PACKET_LENGTH; firstTime=true; } int needToGet=packetSize; do { char* recvBuff; recvBuff = (char*)calloc(needToGet,sizeof(char)); if(recvBuff == NULL){ fprintf(stderr,"Memory allocation problem\n"); return -1; } bytesRecv = recv(AcceptSocket, recvBuff, needToGet, 0); if (bytesRecv == SOCKET_ERROR){ fprintf(stderr,"recv() error %ld.\n", WSAGetLastError()); totalChunkLen=-1; return -1; } if (bytesRecv == 0){ fprintf(stderr,"recv(): socket has shutdown on sender side"); return 0; } else if(bytesRecv > 0) { memcpy(chunkBuff + totalChunkLen,recvBuff,bytesRecv); totalChunkLen+=bytesRecv; } needToGet-=bytesRecv; } while ((totalChunkLen < packetSize) && (!firstTime)); return totalChunkLen; } i use firstTime because for the first time the receiver doesn't know the normal package size that the sender is going to send to it, so i use a MAX_PACKET_LENGTH to get a package and then set the normal package size to the num of bytes i have received my problem is the last package. it's size is less than the package size so lets say last package size is 2 and the normal package size is 4. so recv() gets two bytes, continues to the while condition, then totalChunkLen < packetSize because 2<4 so it iterates the loop again and the gets stuck in recv() because it's blocking because the sender has nothing to send. on the sender side i can't close the connection because i didn't ACK back, so it's kind of a deadlock. receiver is stuck because it's waiting for more packages but sender has nothing to send. i don't want to use a timeout for recv() or to insert a special character to the package header to mark that it is the last one what can i do ? thanks

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  • Socket connection to a telnet-based server hangs on read

    - by mixwhit
    I'm trying to write a simple socket-based client in Python that will connect to a telnet server. I can test the server by telnetting to its port (5007), and entering text. It responds with a NAK (error) or an AK (success), sometimes accompanied by other text. Seems very simple. I wrote a client to connect and communicate with the server, but it hangs on the first attempt to read the response. The connection is successful. Queries like getsockname and getpeername are successful. The send command returns a value that equals the number of characters I'm sending, so it seems to be sending correctly. But in the end, it always hangs when I try to read the response. I've tried using both file-based objects like readline and write (via socket.makefile), as well as using send and recv. With the file object I tried making it with "rw" and reading and writing via that object, and later tried one object for "r" and another for "w" to separate them. None of these worked. I used a packet sniffer to watch what's going on. I'm not versed in all that I'm seeing, but during a telnet session I can see my typed text and the server's text coming back. During my Python socket connection, I can see my text going to the server, but packets back don't seem to have any text in them. Any ideas on what I'm doing wrong, or any strategies to try? Here's the code I'm using (in this case, it's with send and recv): #!/usr/bin/python host = "localhost" port = 5007 msg = "HELLO EMC 1 1" msg2 = "HELLO" import socket import sys try: skt = socket.socket(socket.AF_INET, socket.SOCK_STREAM) except socket.error, e: print("Error creating socket: %s" % e) sys.exit(1) try: skt.connect((host,port)) except socket.gaierror, e: print("Address-related error connecting to server: %s" % e) sys.exit(1) except socket.error, e: print("Error connecting to socket: %s" % e) sys.exit(1) try: print(skt.send(msg)) print("SEND: %s" % msg) except socket.error, e: print("Error sending data: %s" % e) sys.exit(1) while 1: try: buf = skt.recv(1024) print("RECV: %s" % buf) except socket.error, e: print("Error receiving data: %s" % e) sys.exit(1) if not len(buf): break sys.stdout.write(buf)

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  • Load HTML NSString into a UIWebView

    - by ehenrik
    Im doing a project where I connect to a webpage using the NSURLConnection to be able to monitor the status codes that are returned (200 OK / 404 ERROR). I would like to send the user to the top url www.domain.com if I recieve 404 as status code and if i recieve as 200 status code I would like to load the page in to a webview. I have seen several implementations of this problem by creating a new request but I feel that it is unnecessary since you already received the html in the first request so i would just like to load that HTML in to the webView. So i try to use the [webView loadHTMLFromString: baseURL:] but it doesn't always work, I have noticed that when i print the NSString with html in the connectionDidFinnishLoading it sometimes is null and when I monitor these cases by printing the html in didReceiveData a random number of the last packets is NULL (differs between 2-10). It is always the same webpages that doesn't get loaded. If I load them to my webView using [webView loadRequest:myRequest] it always works. My implementation looks like this perhaps someone of you can see what Im doing wrong. I create my first request with a button click. -(IBAction)buttonClick:(id)sender { NSURL *url = [NSURL URLWithString:@"http://www.domain.com/page2/apa.html"]; NSURLRequest *theRequest = [NSURLRequest requestWithURL:url] NSURLConnection *theConnection = [[NSURLConnection alloc] initWithRequest:theRequest delegate:self]; if( theConnection ) { webData = [[NSMutableData data] retain]; } else { } } Then I monitor the response code in the didReceiveResponse method by casting the request to a NSHTTPURLResponse to be able to access the status codes and then setting a Bool depending on the status code. -(void)connection:(NSURLConnection *)connection didReceiveResponse:(NSURLResponse *)response { NSHTTPURLResponse *ne = (NSHTTPURLResponse *)response; if ([ne statusCode] == 200){ ok = TRUE; } [webData setLength: 0]; } I then check the bools value in connectionDidFinnishLoading. If I log the html NSString I get the source of the webpage so i know that it isn't an empty string. -(void)connectionDidFinishLoading:(NSURLConnection *)connection { NSString *html = [[NSString alloc] initWithBytes: [webData mutableBytes] length:[webData length] encoding:NSUTF8StringEncoding]; NSURL *url = [NSURL URLWithString:@"http://www.domain.com/"]; if (ok){ [webView loadHTMLString:html baseURL:url]; ok = FALSE; } else{ //Create a new request to www.domain.com } } webData is an instance variable and I load it in didReceiveData like this. -(void)connection:(NSURLConnection *)connection didReceiveData:(NSData *)data { [webData appendData:data]; }

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  • Cisco VPN Client dropping connection

    - by IT Team
    Using Windows XP and Cisco VPN client version 5.0.4.xxx to connect to a remote customer site. We are able to establish the connection and start an RDP session, but within 1-2 minutes the connection drops and the VPN connection disconnects. The PC making the connection is on a DMZ which is NATed to a public IP address. If we move the PC directly onto the internet without being on the DMZ the connection works and we don't encounter any disconnects. We use a PIX 515E running 7.2.4 and don't have any problems with similar setups connecting to other customer sites from the DMZ. The VPN setup on the client side is pretty basic, using IPSec over TCP port 10000. Not sure what device they are using on the peer, but my guess would be an ASA. Any idea as to what the problem would be? Below is the logs from the VPN client when the problem occurs. The real IP address has been changed to: RemotePeerIP. 4 14:39:30.593 09/23/09 Sev=Info/4 CM/0x63100024 Attempt connection with server "RemotePeerIP" 5 14:39:30.593 09/23/09 Sev=Info/6 CM/0x6310002F Allocated local TCP port 1942 for TCP connection. 6 14:39:30.796 09/23/09 Sev=Info/4 IPSEC/0x63700008 IPSec driver successfully started 7 14:39:30.796 09/23/09 Sev=Info/4 IPSEC/0x63700014 Deleted all keys 8 14:39:30.796 09/23/09 Sev=Info/6 IPSEC/0x6370002C Sent 256 packets, 0 were fragmented. 9 14:39:30.796 09/23/09 Sev=Info/6 IPSEC/0x63700020 TCP SYN sent to RemotePeerIP, src port 1942, dst port 10000 10 14:39:30.796 09/23/09 Sev=Info/6 IPSEC/0x6370001C TCP SYN-ACK received from RemotePeerIP, src port 10000, dst port 1942 11 14:39:30.796 09/23/09 Sev=Info/6 IPSEC/0x63700021 TCP ACK sent to RemotePeerIP, src port 1942, dst port 10000 12 14:39:30.796 09/23/09 Sev=Warning/3 IPSEC/0xA370001C Bad cTCP trailer, Rsvd 26984, Magic# 63697672h, trailer len 101, MajorVer 13, MinorVer 10 13 14:39:30.796 09/23/09 Sev=Info/4 CM/0x63100029 TCP connection established on port 10000 with server "RemotePeerIP" 14 14:39:31.296 09/23/09 Sev=Info/4 CM/0x63100024 Attempt connection with server "RemotePeerIP" 15 14:39:31.296 09/23/09 Sev=Info/6 IKE/0x6300003B Attempting to establish a connection with RemotePeerIP. 16 14:39:31.296 09/23/09 Sev=Info/4 IKE/0x63000013 SENDING ISAKMP OAK AG (SA, KE, NON, ID, VID(Xauth), VID(dpd), VID(Frag), VID(Unity)) to RemotePeerIP 17 14:39:36.296 09/23/09 Sev=Info/4 IKE/0x63000021 Retransmitting last packet! 18 14:39:36.296 09/23/09 Sev=Info/4 IKE/0x63000013 SENDING ISAKMP OAK AG (Retransmission) to RemotePeerIP 19 14:39:41.296 09/23/09 Sev=Info/4 IKE/0x63000021 Retransmitting last packet! 20 14:39:41.296 09/23/09 Sev=Info/4 IKE/0x63000013 SENDING ISAKMP OAK AG (Retransmission) to RemotePeerIP 21 14:39:46.296 09/23/09 Sev=Info/4 IKE/0x63000021 Retransmitting last packet! 22 14:39:46.296 09/23/09 Sev=Info/4 IKE/0x63000013 SENDING ISAKMP OAK AG (Retransmission) to RemotePeerIP 23 14:39:51.328 09/23/09 Sev=Info/4 IKE/0x63000017 Marking IKE SA for deletion (I_Cookie=AEFC3FFF0405BBD6 R_Cookie=0000000000000000) reason = DEL_REASON_PEER_NOT_RESPONDING 24 14:39:51.828 09/23/09 Sev=Info/4 IKE/0x6300004B Discarding IKE SA negotiation (I_Cookie=AEFC3FFF0405BBD6 R_Cookie=0000000000000000) reason = DEL_REASON_PEER_NOT_RESPONDING 25 14:39:51.828 09/23/09 Sev=Info/4 CM/0x63100014 Unable to establish Phase 1 SA with server "RemotePeerIP" because of "DEL_REASON_PEER_NOT_RESPONDING" 26 14:39:51.828 09/23/09 Sev=Info/5 CM/0x63100025 Initializing CVPNDrv 27 14:39:51.828 09/23/09 Sev=Info/4 CM/0x6310002D Resetting TCP connection on port 10000 28 14:39:51.828 09/23/09 Sev=Info/6 CM/0x63100030 Removed local TCP port 1942 for TCP connection. 29 14:39:51.828 09/23/09 Sev=Info/6 CM/0x63100046 Set tunnel established flag in registry to 0. 30 14:39:51.828 09/23/09 Sev=Info/4 IKE/0x63000001 IKE received signal to terminate VPN connection 31 14:39:52.328 09/23/09 Sev=Info/6 IPSEC/0x63700023 TCP RST sent to RemotePeerIP, src port 1942, dst port 10000 32 14:39:52.328 09/23/09 Sev=Info/4 IPSEC/0x63700014 Deleted all keys 33 14:39:52.328 09/23/09 Sev=Info/4 IPSEC/0x63700014 Deleted all keys 34 14:39:52.328 09/23/09 Sev=Info/4 IPSEC/0x63700014 Deleted all keys 35 14:39:52.328 09/23/09 Sev=Info/4 IPSEC/0x6370000A IPSec driver successfully stopped Thank you for any help you can provide.

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  • PPPTP VPN from Ubuntu cannot connect

    - by Andrea Polci
    I'm trying to configure under Linux (Kubuntu 9.10) a VPN I already use from Windows. I installed the network-manager-pptp package and added the vpn under Network Manager. These are the parameter under "advanced" button: Authentication Methods: PAP, CHAP, MSCHAP, SMCHAP2, EAP (I tried also with MSCHAP and MSCHAP2 only) Use MPPE Encryption: yes Crypto: Any Use stateful encryption: no Compression: Allow BSD compression: yes Allow Deflate compression: yes Allow TCP header compression: yes Send PPP echo packets: no When I try to connnect it doesn't work and this is what I get in the system log: 2010-04-08 13:53:47 pcelena NetworkManager <info> Starting VPN service 'org.freedesktop.NetworkManager.pptp'... 2010-04-08 13:53:47 pcelena NetworkManager <info> VPN service 'org.freedesktop.NetworkManager.pptp' started (org.freedesktop.NetworkManager.pptp), PID 4931 2010-04-08 13:53:47 pcelena NetworkManager <info> VPN service 'org.freedesktop.NetworkManager.pptp' just appeared, activating connections 2010-04-08 13:53:47 pcelena pppd[4932] Plugin /usr/lib/pppd/2.4.5//nm-pptp-pppd-plugin.so loaded. 2010-04-08 13:53:47 pcelena NetworkManager <info> VPN plugin state changed: 3 2010-04-08 13:53:47 pcelena pppd[4932] pppd 2.4.5 started by root, uid 0 2010-04-08 13:53:47 pcelena NetworkManager <info> VPN connection 'MYVPN' (Connect) reply received. 2010-04-08 13:53:47 pcelena NetworkManager SCPlugin-Ifupdown: devices added (path: /sys/devices/virtual/net/ppp0, iface: ppp0) 2010-04-08 13:53:47 pcelena NetworkManager SCPlugin-Ifupdown: device added (path: /sys/devices/virtual/net/ppp0, iface: ppp0): no ifupdown configuration found. 2010-04-08 13:53:47 pcelena pppd[4932] Using interface ppp0 2010-04-08 13:53:47 pcelena pppd[4932] Connect: ppp0 <--> /dev/pts/2 2010-04-08 13:53:47 pcelena pptp[4934] nm-pptp-service-4931 log[main:pptp.c:314]: The synchronous pptp option is NOT activated 2010-04-08 13:53:47 pcelena pptp[4927] nm-pptp-service-4918 log[ctrlp_rep:pptp_ctrl.c:251]: Sent control packet type is 7 'Outgoing-Call-Request' 2010-04-08 13:53:47 pcelena pptp[4927] nm-pptp-service-4918 log[ctrlp_disp:pptp_ctrl.c:858]: Received Outgoing Call Reply. 2010-04-08 13:53:47 pcelena pptp[4927] nm-pptp-service-4918 log[ctrlp_disp:pptp_ctrl.c:897]: Outgoing call established (call ID 1, peer's call ID 14800). 2010-04-08 13:53:48 pcelena pppd[4932] CHAP authentication succeeded 2010-04-08 13:53:48 pcelena pppd[4932] CHAP authentication succeeded 2010-04-08 13:53:48 pcelena pppd[4932] LCP terminated by peer 2010-04-08 13:53:48 pcelena pptp[4927] nm-pptp-service-4918 log[ctrlp_disp:pptp_ctrl.c:929]: Call disconnect notification received (call id 14800) 2010-04-08 13:53:48 pcelena pptp[4927] nm-pptp-service-4918 log[ctrlp_disp:pptp_ctrl.c:788]: Received Stop Control Connection Request. 2010-04-08 13:53:48 pcelena pptp[4927] nm-pptp-service-4918 log[ctrlp_rep:pptp_ctrl.c:251]: Sent control packet type is 4 'Stop-Control-Connection-Reply' 2010-04-08 13:53:48 pcelena pptp[4927] nm-pptp-service-4918 log[callmgr_main:pptp_callmgr.c:258]: Closing connection (shutdown) 2010-04-08 13:53:48 pcelena pptp[4927] nm-pptp-service-4918 log[ctrlp_rep:pptp_ctrl.c:251]: Sent control packet type is 12 'Call-Clear-Request' 2010-04-08 13:53:48 pcelena pptp[4927] nm-pptp-service-4918 log[callmgr_main:pptp_callmgr.c:258]: Closing connection (shutdown) 2010-04-08 13:53:48 pcelena pptp[4927] nm-pptp-service-4918 log[ctrlp_rep:pptp_ctrl.c:251]: Sent control packet type is 12 'Call-Clear-Request' 2010-04-08 13:53:48 pcelena pptp[4927] nm-pptp-service-4918 log[call_callback:pptp_callmgr.c:79]: Closing connection (call state) 2010-04-08 13:53:48 pcelena pppd[4932] Modem hangup 2010-04-08 13:53:48 pcelena pppd[4932] Connection terminated. 2010-04-08 13:53:48 pcelena NetworkManager <info> VPN plugin failed: 1 2010-04-08 13:53:48 pcelena NetworkManager SCPlugin-Ifupdown: devices removed (path: /sys/devices/virtual/net/ppp0, iface: ppp0) 2010-04-08 13:53:48 pcelena pppd[4932] Exit. 2010-04-08 13:53:48 pcelena NetworkManager <info> VPN plugin failed: 1 2010-04-08 13:53:48 pcelena NetworkManager <info> VPN plugin state changed: 6 2010-04-08 13:53:48 pcelena NetworkManager <info> VPN plugin state change reason: 0 2010-04-08 13:53:48 pcelena NetworkManager <WARN> connection_state_changed(): Could not process the request because no VPN connection was active. 2010-04-08 13:53:48 pcelena NetworkManager <info> Policy set 'Auto eth0' (eth0) as default for routing and DNS. 2010-04-08 13:54:01 pcelena NetworkManager <debug> [1270727641.001390] ensure_killed(): waiting for vpn service pid 4931 to exit 2010-04-08 13:54:01 pcelena NetworkManager <debug> [1270727641.001479] ensure_killed(): vpn service pid 4931 cleaned up Does anyone has suggestion on what can be the problem and how to make it work?

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  • iptables not allowing mysql connections to aliased ips?

    - by Curtis
    I have a fairly simple iptables firewall on a server that provides MySQL services, but iptables seems to be giving me very inconsistent results. The default policy on the script is as follows: iptables -P INPUT DROP I can then make MySQL public with the following rule: iptables -A INPUT -p tcp --dport 3306 -j ACCEPT With this rule in place, I can connect to MySQL from any source IP to any destination IP on the server without a problem. However, when I try to restrict access to just three IPs by replacing the above line with the following, I run into trouble (xxx=masked octect): iptables -A INPUT -p tcp --dport 3306 -m state --state NEW -s 208.XXX.XXX.184 -j ACCEPT iptables -A INPUT -p tcp --dport 3306 -m state --state NEW -s 208.XXX.XXX.196 -j ACCEPT iptables -A INPUT -p tcp --dport 3306 -m state --state NEW -s 208.XXX.XXX.251 -j ACCEPT Once the above rules are in place, the following happens: I can connect to the MySQL server from the .184, .196 and .251 hosts just fine as long as am connecting to the MySQL server using it's default IP address or an IP alias in the same subnet as the default IP address. I am unable to connect to MySQL using IP aliases that are assigned to the server from a different subnet than the server's default IP when I'm coming from the .184 or .196 hosts, but .251 works just fine. From the .184 or .196 hosts, a telnet attempt just hangs... # telnet 209.xxx.xxx.22 3306 Trying 209.xxx.xxx.22... If I remove the .251 line (making .196 the last rule added), the .196 host still can not connect to MySQL using IP aliases (so it's not the order of the rules that is causing the inconsistent behavior). I know, this particular test was silly as it shouldn't matter what order these three rules are added in, but I figured someone might ask. If I switch back to the "public" rule, all hosts can connect to the MySQL server using either the default or aliased IPs (in either subnet): iptables -A INPUT -p tcp --dport 3306 -j ACCEPT The server is running in a CentOS 5.4 OpenVZ/Proxmox container (2.6.32-4-pve). And, just in case you prefer to see the problem rules in the context of the iptables script, here it is (xxx=masked octect): # Flush old rules, old custom tables /sbin/iptables --flush /sbin/iptables --delete-chain # Set default policies for all three default chains /sbin/iptables -P INPUT DROP /sbin/iptables -P FORWARD DROP /sbin/iptables -P OUTPUT ACCEPT # Enable free use of loopback interfaces /sbin/iptables -A INPUT -i lo -j ACCEPT /sbin/iptables -A OUTPUT -o lo -j ACCEPT # All TCP sessions should begin with SYN /sbin/iptables -A INPUT -p tcp ! --syn -m state --state NEW -j DROP # Accept inbound TCP packets (Do this *before* adding the 'blocked' chain) /sbin/iptables -A INPUT -m state --state ESTABLISHED,RELATED -j ACCEPT # Allow the server's own IP to connect to itself /sbin/iptables -A INPUT -i eth0 -s 208.xxx.xxx.178 -j ACCEPT # Add the 'blocked' chain *after* we've accepted established/related connections # so we remain efficient and only evaluate new/inbound connections /sbin/iptables -N BLOCKED /sbin/iptables -A INPUT -j BLOCKED # Accept inbound ICMP messages /sbin/iptables -A INPUT -p ICMP --icmp-type 8 -j ACCEPT /sbin/iptables -A INPUT -p ICMP --icmp-type 11 -j ACCEPT # ssh (private) /sbin/iptables -A INPUT -p tcp --dport 22 -m state --state NEW -s xxx.xxx.xxx.xxx -j ACCEPT # ftp (private) /sbin/iptables -A INPUT -p tcp --dport 21 -m state --state NEW -s xxx.xxx.xxx.xxx -j ACCEPT # www (public) /sbin/iptables -A INPUT -p tcp --dport 80 -j ACCEPT /sbin/iptables -A INPUT -p tcp --dport 443 -j ACCEPT # smtp (public) /sbin/iptables -A INPUT -p tcp --dport 25 -j ACCEPT /sbin/iptables -A INPUT -p tcp --dport 2525 -j ACCEPT # pop (public) /sbin/iptables -A INPUT -p tcp --dport 110 -j ACCEPT # mysql (private) /sbin/iptables -A INPUT -p tcp --dport 3306 -m state --state NEW -s 208.xxx.xxx.184 -j ACCEPT /sbin/iptables -A INPUT -p tcp --dport 3306 -m state --state NEW -s 208.xxx.xxx.196 -j ACCEPT /sbin/iptables -A INPUT -p tcp --dport 3306 -m state --state NEW -s 208.xxx.xxx.251 -j ACCEPT Any ideas? Thanks in advance. :-)

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  • Windows 2008 R2 SMB / CIFS Logging to diagnose Brother MFC Network Scanning

    - by Steven Potter
    I am attempting to setup network scanning on a brother MFC-9970CDW printer. According to the Brother documentation, the printer is setup to connect to any CIFS network share. I applied all of the appropriate setting in the printer however I get a "sending error" when I try to scan a document. When I look at the logs of the 2008 R2 server that I am attempting to connect to; I can see in the security log where the printer successfully authenticates, however nothing else is logged. I would assume that immediately after the authentication, the printer is making a CIFS request and some sort of error is occurring, however I can't seem to find any way to log this information to find out what is going on. Is it possible to get Windows 2008 to log SMB/CIFS traffic? Followup: I installed Microsoft netmon and captured the packets associated with the transaction: 510 3:04:28 PM 7/9/2012 34.4277743 System 192.168.1.134 192.168.1.10 SMB SMB:C; Negotiate, Dialect = NT LM 0.12 {SMBOverTCP:30, TCP:29, IPv4:22} 511 3:04:28 PM 7/9/2012 34.4281246 System 192.168.1.10 192.168.1.134 SMB SMB:R; Negotiate, Dialect is NT LM 0.12 (#0), SpnegoToken (1.3.6.1.5.5.2) {SMBOverTCP:30, TCP:29, IPv4:22} 519 3:04:29 PM 7/9/2012 34.8986214 System 192.168.1.134 192.168.1.10 SMB SMB:C; Session Setup Andx, NTLM NEGOTIATE MESSAGE {SMBOverTCP:30, TCP:29, IPv4:22} 520 3:04:29 PM 7/9/2012 34.8989310 System 192.168.1.10 192.168.1.134 SMB SMB:R; Session Setup Andx, NTLM CHALLENGE MESSAGE - NT Status: System - Error, Code = (22) STATUS_MORE_PROCESSING_REQUIRED {SMBOverTCP:30, TCP:29, IPv4:22} 522 3:04:29 PM 7/9/2012 34.9022870 System 192.168.1.134 192.168.1.10 SMB SMB:C; Session Setup Andx, NTLM AUTHENTICATE MESSAGEVersion:v2, Domain: CORP, User: PRINTSUPOFF, Workstation: BRN001BA9AD1FE6 {SMBOverTCP:30, TCP:29, IPv4:22} 523 3:04:29 PM 7/9/2012 34.9032421 System 192.168.1.10 192.168.1.134 SMB SMB:R; Session Setup Andx {SMBOverTCP:30, TCP:29, IPv4:22} 525 3:04:29 PM 7/9/2012 34.9051855 System 192.168.1.134 192.168.1.10 SMB SMB:C; Tree Connect Andx, Path = \\192.168.1.10\IPC$, Service = ????? {SMBOverTCP:30, TCP:29, IPv4:22} 526 3:04:29 PM 7/9/2012 34.9053083 System 192.168.1.10 192.168.1.134 SMB SMB:R; Tree Connect Andx, Service = IPC {SMBOverTCP:30, TCP:29, IPv4:22} 528 3:04:29 PM 7/9/2012 34.9073573 System 192.168.1.134 192.168.1.10 DFSC DFSC:Get DFS Referral Request, FileName: \\192.168.1.10\NSCFILES, MaxReferralLevel: 3 {SMB:33, SMBOverTCP:30, TCP:29, IPv4:22} 529 3:04:29 PM 7/9/2012 34.9152042 System 192.168.1.10 192.168.1.134 SMB SMB:R; Transact2, Get Dfs Referral - NT Status: System - Error, Code = (549) STATUS_NOT_FOUND {SMB:33, SMBOverTCP:30, TCP:29, IPv4:22} 531 3:04:29 PM 7/9/2012 34.9169738 System 192.168.1.134 192.168.1.10 SMB SMB:C; Tree Disconnect {SMBOverTCP:30, TCP:29, IPv4:22} 532 3:04:29 PM 7/9/2012 34.9170688 System 192.168.1.10 192.168.1.134 SMB SMB:R; Tree Disconnect {SMBOverTCP:30, TCP:29, IPv4:22} As you can see, the DFS referral fails and the transaction is shut down. I can't see any reason for the DFS referral to fail. The only reference I can find online is: https://bugzilla.samba.org/show_bug.cgi?id=8003 Anyone have any ideas for a solution?

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  • OpenBSD configuration: Client unable to mount via NFS using Berkeley Automounter (amd)

    - by Rilindo
    What I am trying to do is to have my openBSD client (OpenBSD 4.9) auto mount a Linux NFS file system (Scientific Linux 6.1). So far, I am not sure if it is configured correctly. To get things out of the way, I am able to mount nfs manually: # mount_nfs -T -3 192.168.15.100:/exports /mnt # ls -la /mnt total 52 drwxr-xr-x 7 root wheel 4096 Oct 4 22:42 . drwxr-xr-x 16 root wheel 512 Nov 26 16:33 .. drwxrwxr-x 5 _sndio _sndio 4096 Oct 31 21:58 centos drwxr-xr-x 15 root wheel 4096 Nov 6 09:17 home drwxr-xr-x 5 root wheel 4096 Oct 31 21:27 sl drwxr-xr-x 3 root wheel 4096 Nov 19 16:02 sles drwxr-xr-x 17 503 503 4096 Nov 10 17:37 users # So connectivity is not an issue, as far as I can tell. As per man page, the following is configured in /etc/amd/auto.home: /defaults type:=nfs;sublink:=${key};opts:=rw,soft,intr,vers=3,proto=tcp * rhost:=192.168.15.100;rfs:=/exports In turn, /etc/amd/master is configured as such: # cat /etc/amd/master /exports amd.home Upon reboot, I can it see mount, but curiously enough, instead of the hostname: amd:24490 0 0 0 100% /exports From what I understand, amd acts a little different from FreeBSD. Still, I tried to see if I it can automount. Nope: ksh: cd: /exports/users - Resource temporarily unavailable # cd /exports/192.168.15.100/host/users ksh: cd: /exports/192.168.15.100/host/users - Resource temporarily unavailable A search in google doesn't help too much - it seems that automounting NFS with OpenBSD is not something that is usually done. Other than this, information is fairly sparse. I can, of course, always mount is permanently, but I tend to be a bit anal on convention, so no for now. :) Some direction would be appreciation. (And oh, in case you are a wondering, I tried FreeBSD way of using amd and that hasn't worked out - although I wouldn't mind an explanation of the difference between how FreeBSD implements and how OpenBSD implements it) UPDATE: After re-writing the map file several times, I got as far as actually communicating with the NFS server with this configuration: /defaults type:=nfs;rhost:=kerberos.monzell.com;rfs:=/exports;\ sublink:=${key};opts:=rw,nodev,nosuid,soft,intr,tcp,resvport * ${host}==${rhost};type:=nfs;fs:=${rfs};opts:=rw,nodev,nosuid,soft,intr,tcp,resvport However, for some reason, it seems that amd will only default to NFS version 2 over udp: # tcpdump dst kerberos tcpdump: listening on pcn0, link-type EN10MB tcpdump: WARNING: compensating for unaligned libpcap packets 20:38:28.558385 openbsd.monzell.com.856 > kerberos.monzell.com.sunrpc: udp 100 20:38:28.559154 openbsd.monzell.com.856 > kerberos.monzell.com.892: udp 96 20:38:30.592761 openbsd.monzell.com.856 > kerberos.monzell.com.nfsd: xid 0x22000000 (NFSv2) 40 null 20:38:33.558107 arp reply openbsd.monzell.com is-at 52:54:00:52:8f:66 I tried various options of forcing it to try to mount as nfsv3 such as: /defaults type:=nfs;rhost:=kerberos.monzell.com;rfs:=/exports;\ sublink:=${key};opts:=rw,nodev,nosuid,soft,intr,vers=3,proto=tcp,resvport * ${host}==${rhost};type:=nfs;fs:=${rfs};opts:=rw,nodev,nosuid,soft,intr,vers=3,proto=tcp,resvport or: /defaults type:=nfs;rhost:=kerberos.monzell.com;rfs:=/exports;\ sublink:=${key};opts:=rw,nodev,nosuid,soft,intr,vers=-3,proto=tcp,resvport * ${host}==${rhost};type:=nfs;fs:=${rfs};opts:=rw,nodev,nosuid,soft,intr,vers=3,proto=tcp,resvport Nothing yet still. Curious enough, OpenBSD mounts defaults to version 3, so I am not sure why it would start with version in amd. What would be the correct options to pass?

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  • Networking in VirtualBox

    - by Fat Bloke
    Networking in VirtualBox is extremely powerful, but can also be a bit daunting, so here's a quick overview of the different ways you can setup networking in VirtualBox, with a few pointers as to which configurations should be used and when. VirtualBox allows you to configure up to 8 virtual NICs (Network Interface Controllers) for each guest vm (although only 4 are exposed in the GUI) and for each of these NICs you can configure: Which virtualized NIC-type is exposed to the Guest. Examples include: Intel PRO/1000 MT Server (82545EM),  AMD PCNet FAST III (Am79C973, the default) or  a Paravirtualized network adapter (virtio-net). How the NIC operates with respect to your Host's physical networking. The main modes are: Network Address Translation (NAT) Bridged networking Internal networking Host-only networking NAT with Port-forwarding The choice of NIC-type comes down to whether the guest has drivers for that NIC.  VirtualBox, suggests a NIC based on the guest OS-type that you specify during creation of the vm, and you rarely need to modify this. But the choice of networking mode depends on how you want to use your vm (client or server) and whether you want other machines on your network to see it. So let's look at each mode in a bit more detail... Network Address Translation (NAT) This is the default mode for new vm's and works great in most situations when the Guest is a "client" type of vm. (i.e. most network connections are outbound). Here's how it works: When the guest OS boots,  it typically uses DHCP to get an IP address. VirtualBox will field this DHCP request and tell the guest OS its assigned IP address and the gateway address for routing outbound connections. In this mode, every vm is assigned the same IP address (10.0.2.15) because each vm thinks they are on their own isolated network. And when they send their traffic via the gateway (10.0.2.2) VirtualBox rewrites the packets to make them appear as though they originated from the Host, rather than the Guest (running inside the Host). This means that the Guest will work even as the Host moves from network to network (e.g. laptop moving between locations), and from wireless to wired connections too. However, how does another computer initiate a connection into a Guest?  e.g. connecting to a web server running in the Guest. This is not (normally) possible using NAT mode as there is no route into the Guest OS. So for vm's running servers we need a different networking mode.... Bridged Networking Bridged Networking is used when you want your vm to be a full network citizen, i.e. to be an equal to your host machine on the network. In this mode, a virtual NIC is "bridged" to a physical NIC on your host, like this: The effect of this is that each VM has access to the physical network in the same way as your host. It can access any service on the network such as external DHCP services, name lookup services, and routing information just as the host does. Logically, the network looks like this: The downside of this mode is that if you run many vm's you can quickly run out of IP addresses or your network administrator gets fed up with you asking for statically assigned IP addresses. Secondly, if your host has multiple physical NICs (e.g. Wireless and Wired) you must reconfigure the bridge when your host jumps networks.  Hmm, so what if you want to run servers in vm's but don't want to involve your network administrator? Maybe one of the next 2 modes is for you... Internal Networking When you configure one or more vm's to sit on an Internal network, VirtualBox ensures that all traffic on that network stays within the host and is only visible to vm's on that virtual network. Configuration looks like this: The internal network ( in this example "intnet" ) is a totally isolated network and so is very "quiet". This is good for testing when you need a separate, clean network, and you can create sophisticated internal networks with vm's that provide their own services to the internal network. (e.g. Active Directory, DHCP, etc). Note that not even the Host is a member of the internal network, but this mode allows vm's to function even when the Host is not connected to a network (e.g. on a plane). Note that in this mode, VirtualBox provides no "convenience" services such as DHCP, so your machines must be statically configured or one of the vm's needs to provide a DHCP/Name service. Multiple internal networks are possible and you can configure vm's to have multiple NICs to sit across internal and other network modes and thereby provide routes if needed. But all this sounds tricky. What if you want an Internal Network that the host participates on with VirtualBox providing IP addresses to the Guests? Ah, then for this, you might want to consider Host-only Networking... Host-only Networking Host-only Networking is like Internal Networking in that you indicate which network the Guest sits on, in this case, "vboxnet0": All vm's sitting on this "vboxnet0" network will see each other, and additionally, the host can see these vm's too. However, other external machines cannot see Guests on this network, hence the name "Host-only". Logically, the network looks like this: This looks very similar to Internal Networking but the host is now on "vboxnet0" and can provide DHCP services. To configure how a Host-only network behaves, look in the VirtualBox Manager...Preferences...Network dialog: Port-Forwarding with NAT Networking Now you may think that we've provided enough modes here to handle every eventuality but here's just one more... What if you cart around a mobile-demo or dev environment on, say, a laptop and you have one or more vm's that you need other machines to connect into? And you are continually hopping onto different (customer?) networks. In this scenario: NAT - won't work because external machines need to connect in. Bridged - possibly an option, but does your customer want you eating IP addresses and can your software cope with changing networks? Internal - we need the vm(s) to be visible on the network, so this is no good. Host-only - same problem as above, we want external machines to connect in to the vm's. Enter Port-forwarding to save the day! Configure your vm's to use NAT networking; Add Port Forwarding rules; External machines connect to "host":"port number" and connections are forwarded by VirtualBox to the guest:port number specified. For example, if your vm runs a web server on port 80, you could set up rules like this:  ...which reads: "any connections on port 8080 on the Host will be forwarded onto this vm's port 80".  This provides a mobile demo system which won't need re-configuring every time you open your laptop lid. Summary VirtualBox has a very powerful set of options allowing you to set up almost any configuration your heart desires. For more information, check out the VirtualBox User Manual on Virtual Networking. -FB 

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