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  • Receiving broadcast messages

    - by Prasad
    Hi, I'm trying to receive broadcast messages using C# code in an ISDN network with BRI interface at my end. I see the packets sent to the broadcast ip address (239.255.255.255) on some ports using Comm View tool. But when I try to listen to this IP address, it says the address is not in a valid context. But when I send broadcast messages to 255.255.255.255 on a port, I can receive those messages with the below code.. What could be the problem with this ip address - 239.255.255.255 ? The code I use to listen to broadcast messages is.. UdpClient udp = new UdpClient(); IPEndPoint receiveEndPoint = new IPEndPoint(IPAddress.Any, 8013); // If I use IPAddress.Parse("239.255.255.255") to listen to, // it says "the address is not in a valid // context." udp.Client.Bind(receiveEndPoint); udp.BeginReceive(_Callback, udp); static private void _Callback(IAsyncResult iar) { try { UdpClient client = (UdpClient)iar.AsyncState; client.BeginReceive(_Callback, client); IPEndPoint ipRemote = new IPEndPoint(IPAddress.Any, 8013); byte[] rgb = client.EndReceive(iar, ref ipRemote); Console.WriteLine("Received {0} bytes: \"{1}\"", rgb.Length.ToString(), Encoding.UTF8.GetString(rgb)); } catch (ObjectDisposedException) { Console.WriteLine("closing listening socket"); } catch (Exception exc) { Console.WriteLine("Listening socket error: \"" + exc.Message + "\""); } } There are packets sent to the broadcast ipaddress (239.255.255.255) which I can see in Commview tool, but can't receive them from the code... Can anybody help me out please? Thanking you in advance, Prasad Kancharla.

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  • How to limit traffic using multicast over localhost

    - by Shane Holloway
    I'm using multicast UDP over localhost to implement a loose collection of cooperative programs running on a single machine. The following code works well on Mac OSX, Windows and linux. The flaw is that the code will receive UDP packets outside of the localhost network as well. For example, sendSock.sendto(pkt, ('192.168.0.25', 1600)) is received by my test machine when sent from another box on my network. import platform, time, socket, select addr = ("239.255.2.9", 1600) sendSock = socket.socket(socket.AF_INET, socket.SOCK_DGRAM, socket.IPPROTO_UDP) sendSock.setsockopt(socket.IPPROTO_IP, socket.IP_MULTICAST_TTL, 24) sendSock.setsockopt(socket.IPPROTO_IP, socket.IP_MULTICAST_IF, socket.inet_aton("127.0.0.1")) recvSock = socket.socket(socket.AF_INET, socket.SOCK_DGRAM, socket.IPPROTO_UDP) recvSock.setsockopt(socket.SOL_SOCKET, socket.SO_REUSEADDR, True) if hasattr(socket, 'SO_REUSEPORT'): recvSock.setsockopt(socket.SOL_SOCKET, socket.SO_REUSEPORT, True) recvSock.bind(("0.0.0.0", addr[1])) status = recvSock.setsockopt(socket.IPPROTO_IP, socket.IP_ADD_MEMBERSHIP, socket.inet_aton(addr[0]) + socket.inet_aton("127.0.0.1")); while 1: pkt = "Hello host: {1} time: {0}".format(time.ctime(), platform.node()) print "SEND to: {0} data: {1}".format(addr, pkt) r = sendSock.sendto(pkt, addr) while select.select([recvSock], [], [], 0)[0]: data, fromAddr = recvSock.recvfrom(1024) print "RECV from: {0} data: {1}".format(fromAddr, data) time.sleep(2) I've attempted to recvSock.bind(("127.0.0.1", addr[1])), but that prevents the socket from receiving any multicast traffic. Is there a proper way to configure recvSock to only accept multicast packets from the 127/24 network, or do I need to test the address of each received packet?

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  • Combining SQL Rows

    - by lumberjack4
    I've got SQL Compact Database that contains a table of IP Packet Headers. The Table looks like this: Table: PacketHeaders ID SrcAddress SrcPort DestAddress DestPort Bytes 1 10.0.25.1 255 10.0.25.50 500 64 2 10.0.25.50 500 10.0.25.1 255 80 3 10.0.25.50 500 10.0.25.1 255 16 4 75.48.0.25 387 74.26.9.40 198 72 5 74.26.9.40 198 75.48.0.25 387 64 6 10.0.25.1 255 10.0.25.50 500 48 I need to perform a query to show 'conversations' going on across a local network. Packets going from A - B is part of the same conversations as packets going from B - A. I need to perform a query to show the on going conversations. Basically what I need is something that looks like this: Returned Query: SrcAddress SrcPort DestAddress DestPort TotalBytes BytesA->B BytesB->A 10.0.25.1 255 10.0.25.50 500 208 112 96 75.48.0.25 387 74.26.9.40 198 136 72 64 As you can see I need the query (or series of queries) to recognize that A-B is the same as B-A and break up the byte counts accordingly. I'm not a SQL guru by any means but any help on this would be greatly appreciated.

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  • Testing a patch to the Rails mysql adapter

    - by Sleepycat
    I wrote a little monkeypatch to the Rails MySQLAdapter and want to package it up to use it in my other projects. I am trying to write some tests for it but I am still new to testing and I am not sure how to test this. Can someone help get me started? Here is the code I want to test: unless RAILS_ENV == 'production' module ActiveRecord module ConnectionAdapters class MysqlAdapter < AbstractAdapter def select_with_explain(sql, name = nil) explanation = execute_with_disable_logging('EXPLAIN ' + sql) e = explanation.all_hashes.first exp = e.collect{|k,v| " | #{k}: #{v} "}.join log(exp, 'Explain') select_without_explain(sql, name) end def execute_with_disable_logging(sql, name = nil) #:nodoc: #Run a query without logging @connection.query(sql) rescue ActiveRecord::StatementInvalid => exception if exception.message.split(":").first =~ /Packets out of order/ raise ActiveRecord::StatementInvalid, "'Packets out of order' error was received from the database. Please update your mysql bindings (gem install mysql) and read http://dev.mysql.com/doc/mysql/en/password-hashing.html for more information. If you're on Windows, use the Instant Rails installer to get the updated mysql bindings." else raise end end alias_method_chain :select, :explain end end end end Thanks.

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  • Altruistic network connection bandwidth estimation

    - by datenwolf
    Assume two peers Alice and Bob connected over a IP network. Alice and Bob are exchanging packets of lossy compressed data which are generated and to be consumes in real time (think a VoIP or video chat application). The service is designed to cope with as little bandwidth available, but relies on low latencies. Alice and Bob would mark their connection with an apropriate QoS profile. Alice and Bob want use a variable bitrate compression and would like to consume all of the leftover bandwidth available for the connection between them, but would voluntarily reduce the consumed bitrate depending on the state of the network. However they'd like to retain a stable link, i.e. avoid interruptions in their decoded data stream caused by congestion and the delay until the bandwidth got adjusted. However it is perfectly possible for them to loose a few packets. TL;DR: Alice and Bob want to implement a VoIP protocol from scratch, and are curious about bandwidth and congestion control. What papers and resources do you suggest for Alice and Bob to read? Mainly in the area of bandwidth estimation and congestion control.

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  • error detection/correction/recovery in serial protocols

    - by Jason S
    I have some designing to do for a serial protocol and am running into some questions that I figure must have been considered elsewhere. So I'm wondering if there are some recommendations for best practices in designing serial protocols. (Please either state a fact that is easily verifiable, or cite a reputable source if you make a claim.) General recommendations for websites/books are also welcome. In particular I have to deal with issues like parsing a stream of bytes into packets verifying a packet is correct (easy with a CRC, for instance) identifying reasonable types of errors that can occur (e.g. in a point-to-point serial stream, sporadic single bit errors, and dropped series of bytes, are both likely, but extra phantom bytes are unlikely; whereas with a record stored in flash memory or on a disk drive the types of errors that predominate are different) error correction or recovery (if I detect an error in a packet, can I correct it? If not, can I resync to the boundary of the next packet?) how to make variable-length packets robust to error correction / recovery. Any suggestions?

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  • libpcap read packet size

    - by spicyramen
    I started to write an application which will read RTP/H.264 video packets from an existing .pcap file, I need to read the packet size. I tried to use packet-len or header-len, but it never displays the right number of bytes for packets (I'm using wireshark to verify packet size - under Length column). How to do it? This is part of my code: while (packet = pcap_next(handle,&header)) { u_char *pkt_ptr = (u_char *)packet; struct ip *ip_hdr = (struct ip *)pkt_ptr; //point to an IP header structure struct pcap_pkthdr *pkt_hdr =(struct pcap_pkthdr *)packet; unsigned int packet_length = pkt_hdr->len; unsigned int ip_length = ntohs(ip_hdr->ip_len); printf("Packet # %i IP Header length: %d bytes, Packet length: %d bytes\n",pkt_counter,ip_length,packet_length); Packet # 0 IP Header length: 180 bytes, Packet length: 104857664 bytes Packet # 1 IP Header length: 52 bytes, Packet length: 104857600 bytes Packet # 2 IP Header length: 100 bytes, Packet length: 104857600 bytes Packet # 3 IP Header length: 100 bytes, Packet length: 104857664 bytes Packet # 4 IP Header length: 52 bytes, Packet length: 104857600 bytes Packet # 5 IP Header length: 100 bytes, Packet length: 104857600 bytes Another option I tried is to use: pkt_ptr- I get: read_pcapfile.c:67:43: error: request for member ‘len’ in something not a structure or union

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  • RPC for java/python with rest support, HTML monitoring and goodies

    - by Ran
    Here's my set of requirements: I'm looking for an RPC framework such as thrift, avro, protobuf (when adding services to it) which supports: Easy and intuitive IDL. No serial numbers, no manual versioning, simple... avro is a good example for this. Works with Java and Python Supports both fast binary prorocol, as well as HTTP based restful style. I'd like to be able to use it for both backend-to-backend communication (java-java or python-java) as well as frontend-to-backend communication (javascript to java). The rest support needs to include &param=value input as get/post requests (configurable per request) and output in three possible formats: json, jsonp, XML. Compact, fast, backward compatible, easy to upgrade etc... Provides some nice monitoring interfaces such as: JMX, web page status reports (e.g. packets in, packets out, error rate etc) Ops friendly... no need to take the whole site down to release new versions Both sync and asyc communication ... other goodies are welcome... Is there something out there? So far I've looked at thrift and avro and they are both nice in some ways, but don't check all my list. Thanks

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  • Text piped to PowerShell.exe isn't recieved when using [Console]::ReadLine()

    - by crtracy
    I'm getting itermittent data loss when calling .NET [Console]::ReadLine() to read piped input to PowerShell.exe: >ping localhost | powershell -NonInteractive -NoProfile -C "do {$line = [Console]::ReadLine(); ('' + (Get-Date -f 'HH:mm :ss') + $line) | Write-Host; } while ($line -ne $null)" 23:56:45time<1ms 23:56:45 23:56:46time<1ms 23:56:46 23:56:47time<1ms 23:56:47 23:56:47 Normally 'ping localhost' from Vista64 looks like this, so there is a lot of data missing from the output above: Pinging WORLNTEC02.bnysecurities.corp.local [::1] from ::1 with 32 bytes of data: Reply from ::1: time<1ms Reply from ::1: time<1ms Reply from ::1: time<1ms Reply from ::1: time<1ms Ping statistics for ::1: Packets: Sent = 4, Received = 4, Lost = 0 (0% loss), Approximate round trip times in milli-seconds: Minimum = 0ms, Maximum = 0ms, Average = 0ms But using the same API from C# recieves all the data sent to the process (excluding some newline differences). Code: namespace ConOutTime { class Program { static void Main (string[] args) { string s; while ((s = Console.ReadLine ()) != null) { if (s.Length > 0) // don't write time for empty lines Console.WriteLine("{0:HH:mm:ss} {1}", DateTime.Now, s); } } } } Output: 00:44:30 Pinging WORLNTEC02.bnysecurities.corp.local [::1] from ::1 with 32 bytes of data: 00:44:30 Reply from ::1: time<1ms 00:44:31 Reply from ::1: time<1ms 00:44:32 Reply from ::1: time<1ms 00:44:33 Reply from ::1: time<1ms 00:44:33 Ping statistics for ::1: 00:44:33 Packets: Sent = 4, Received = 4, Lost = 0 (0% loss), 00:44:33 Approximate round trip times in milli-seconds: 00:44:33 Minimum = 0ms, Maximum = 0ms, Average = 0ms So, if calling the same API from PowerShell instead of C# many parts of StdIn get 'eaten'. Is the PowerShell host reading string from StdIn even though I didn't use 'PowerShell.exe -Command -'?

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  • UDP sockets in ad hoc network (Ubuntu 9.10)

    - by Ekhiotz
    Hi! I am using BSD sockets in Ubuntu 9.10 to send UDP packets in broadcast with the following code: sock_fd = socket(PF_INET,SOCK_DGRAM,IPPROTO_UDP); //sock_fd=socket(AF_INET,SOCK_DGRAM,0); receiver_addr.sin_family = PF_INET; //does not send with broadcast in ad hoc receiver_addr.sin_addr.s_addr = htonl(INADDR_BROADCAST); inet_aton("169.254.255.255",&receiver_addr.sin_addr); receiver_addr.sin_port = htons(port); int broadcast = 1; // this call is what allows broadcast packets to be sent: if (setsockopt(sock_fd, SOL_SOCKET, SO_BROADCAST, &broadcast, sizeof broadcast) == -1) { perror("setsockopt (SO_BROADCAST)"); exit(1); } ret=sendto(sock_fd, packet, size, 0,(struct sockaddr*)&receiver_addr,sizeof(receiver_addr)); Note that is not all the code, it is only to have an idea. The program sends all the data with INADDR_BROADCAST if I am connected to an infrastructure wireless network. However, if my laptop is connected to an ad-hoc network, it is able to receive all the data, but not to send it. I have solved the problem using the 169.254.255.255 broadcast address, but I would like to know what is going on. Thank you in advance!

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  • Cannot connect to Github?

    - by user2973438
    so I tried to push some updates onto my repo on github via terminal on Mac OSX 10.8.4 and it doesn't work. I've been getting the same error many times: Lillys-MacBook-Air:Yuewei Lilly$ git push origin master error: Failed connect to github.com:443; Operation timed out while accessing https://github.com/lillybeans/Yuewei.git/info/refs?service=git-receive-pack fatal: HTTP request failed Some background: I've pushed many projects onto github before using terminal (when I was in Canada). I am currently in Shanghai, China, could it be the GFW? But when I was in Beijing, I was able to push projects onto github still. when I do ping github.com: Lillys-MacBook-Air:Yuewei Lilly$ ping github.com PING github.com (192.30.252.131): 56 data bytes Request timeout for icmp_seq 0 Request timeout for icmp_seq 1 ping: sendto: No route to host Request timeout for icmp_seq 2 ping: sendto: Host is down Request timeout for icmp_seq 3 ping: sendto: Host is down Request timeout for icmp_seq 4 ping: sendto: Host is down Request timeout for icmp_seq 5 ping: sendto: Host is down Request timeout for icmp_seq 6 ping: sendto: Host is down Request timeout for icmp_seq 7 ^C --- github.com ping statistics --- 9 packets transmitted, 0 packets received, 100.0% packet loss Lillys-MacBook-Air:Yuewei Lilly$ I have ShadowSocks (proxy) turned on. Without it I can't access github.com via browser, with it, I can. also when I do "git remote -v" I see both my pull and push remote repos correctly listed. Thank you in advance!

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  • What's the cleanest way to do byte-level manipulation?

    - by Jurily
    I have the following C struct from the source code of a server, and many similar: // preprocessing magic: 1-byte alignment typedef struct AUTH_LOGON_CHALLENGE_C { // 4 byte header uint8 cmd; uint8 error; uint16 size; // 30 bytes uint8 gamename[4]; uint8 version1; uint8 version2; uint8 version3; uint16 build; uint8 platform[4]; uint8 os[4]; uint8 country[4]; uint32 timezone_bias; uint32 ip; uint8 I_len; // I_len bytes uint8 I[1]; } sAuthLogonChallenge_C; // usage (the actual code that will read my packets): sAuthLogonChallenge_C *ch = (sAuthLogonChallenge_C*)&buf[0]; // where buf is a raw byte array These are TCP packets, and I need to implement something that emits and reads them in C#. What's the cleanest way to do this? My current approach involves [StructLayout(LayoutKind.Sequential, Pack = 1)] unsafe struct foo { ... } and a lot of fixed statements to read and write it, but it feels really clunky, and since the packet itself is not fixed length, I don't feel comfortable using it. Also, it's a lot of work. However, it does describe the data structure nicely, and the protocol may change over time, so this may be ideal for maintenance. What are my options? Would it be easier to just write it in C++ and use some .NET magic to use that?

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  • Where are possible locations of queueing/buffering delays in Linux multicast?

    - by Matt
    We make heavy use of multicasting messaging across many Linux servers on a LAN. We are seeing a lot of delays. We basically send an enormous number of small packages. We are more concerned with latency than throughput. The machines are all modern, multi-core (at least four, generally eight, 16 if you count hyperthreading) machines, always with a load of 2.0 or less, usually with a load less than 1.0. The networking hardware is also under 50% capacity. The delays we see look like queueing delays: the packets will quickly start increasing in latency, until it looks like they jam up, then return back to normal. The messaging structure is basically this: in the "sending thread", pull messages from a queue, add a timestamp (using gettimeofday()), then call send(). The receiving program receives the message, timestamps the receive time, and pushes it in a queue. In a separate thread, the queue is processed, analyzing the difference between sending and receiving timestamps. (Note that our internal queues are not part of the problem, since the timestamps are added outside of our internal queuing.) We don't really know where to start looking for an answer to this problem. We're not familiar with Linux internals. Our suspicion is that the kernel is queuing or buffering the packets, either on the send side or the receive side (or both). But we don't know how to track this down and trace it. For what it's worth, we're using CentOS 4.x (RHEL kernel 2.6.9).

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  • Which are the most useful techniques for faster Bluetooth?

    - by Mike Howard
    Hi. I'm adding peer-to-peer bluetooth using GameKit to an iPhone shoot-em-up, so speed is vital. I'm sending about 40 messages a second each way, most of them with the faster GKSendDataUnreliable, all serializing with NSCoding. In testing between a 3G and 3GS, this is slowing the 3G down a lot more than I'd like. I'm wondering where I should concentrate my efforts to speed it up. How much slower is GKSendDataReliable? For the few packets that have to get there, would it be faster to send a GKSendDataUnreliable and have the peer send an acknowledgement so I can send again if I don't get the Ack within, say, 100ms? How much faster would it be to create the NSData instance using a regular C array rather than archiving with the NSCoding protocol? Is this serialization process (for about a dozen floats) just as slow as you'd expect from an object creation/deallocation overhead, or is something particularly slow happening? I heard that (for example) sending four seperate sets of data is much, much slower, than sending one piece of data four times the size. Would I make a significant saving by sending separate packets of data that wouldn't always go together in the same packet when they happen at the same time? Are there any other bluetooth performance secrets I've missed? Thanks for your help.

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  • iPhone: Which are the most useful techniques for faster Bluetooth?

    - by Mike Howard
    Hi. I'm adding peer-to-peer bluetooth using GameKit to an iPhone shoot-em-up, so speed is vital. I'm sending about 40 messages a second each way, most of them with the faster GKSendDataUnreliable, all serializing with NSCoding. In testing between a 3G and 3GS, this is slowing the 3G down a lot more than I'd like. I'm wondering where I should concentrate my efforts to speed it up. How much slower is GKSendDataReliable? For the few packets that have to get there, would it be faster to send a GKSendDataUnreliable and have the peer send an acknowledgement so I can send again if I don't get the Ack within, say, 100ms? How much faster would it be to create the NSData instance using a regular C array rather than archiving with the NSCoding protocol? Is this serialization process (for about a dozen floats) just as slow as you'd expect from an object creation/deallocation overhead, or is something particularly slow happening? I heard that (for example) sending four seperate sets of data is much, much slower, than sending one piece of data four times the size. Would I make a significant saving by sending separate packets of data that wouldn't always go together in the same packet when they happen at the same time? Are there any other bluetooth performance secrets I've missed? Thanks for your help.

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  • Securing an ASP.NET MVC 2 Application

    - by rajbk
    This post attempts to look at some of the methods that can be used to secure an ASP.NET MVC 2 Application called Northwind Traders Human Resources.  The sample code for the project is attached at the bottom of this post. We are going to use a slightly modified Northwind database. The screen capture from SQL server management studio shows the change. I added a new column called Salary, inserted some random salaries for the employees and then turned off AllowNulls.   The reporting relationship for Northwind Employees is shown below.   The requirements for our application are as follows: Employees can see their LastName, FirstName, Title, Address and Salary Employees are allowed to edit only their Address information Employees can see the LastName, FirstName, Title, Address and Salary of their immediate reports Employees cannot see records of non immediate reports.  Employees are allowed to edit only the Salary and Title information of their immediate reports. Employees are not allowed to edit the Address of an immediate report Employees should be authenticated into the system. Employees by default get the “Employee” role. If a user has direct reports, they will also get assigned a “Manager” role. We use a very basic empId/pwd scheme of EmployeeID (1-9) and password test$1. You should never do this in an actual application. The application should protect from Cross Site Request Forgery (CSRF). For example, Michael could trick Steven, who is already logged on to the HR website, to load a page which contains a malicious request. where without Steven’s knowledge, a form on the site posts information back to the Northwind HR website using Steven’s credentials. Michael could use this technique to give himself a raise :-) UI Notes The layout of our app looks like so: When Nancy (EmpID 1) signs on, she sees the default page with her details and is allowed to edit her address. If Nancy attempts to view the record of employee Andrew who has an employeeID of 2 (Employees/Edit/2), she will get a “Not Authorized” error page. When Andrew (EmpID 2) signs on, he can edit the address field of his record and change the title and salary of employees that directly report to him. Implementation Notes All controllers inherit from a BaseController. The BaseController currently only has error handling code. When a user signs on, we check to see if they are in a Manager role. We then create a FormsAuthenticationTicket, encrypt it (including the roles that the employee belongs to) and add it to a cookie. private void SetAuthenticationCookie(int employeeID, List<string> roles) { HttpCookiesSection cookieSection = (HttpCookiesSection) ConfigurationManager.GetSection("system.web/httpCookies"); AuthenticationSection authenticationSection = (AuthenticationSection) ConfigurationManager.GetSection("system.web/authentication"); FormsAuthenticationTicket authTicket = new FormsAuthenticationTicket( 1, employeeID.ToString(), DateTime.Now, DateTime.Now.AddMinutes(authenticationSection.Forms.Timeout.TotalMinutes), false, string.Join("|", roles.ToArray())); String encryptedTicket = FormsAuthentication.Encrypt(authTicket); HttpCookie authCookie = new HttpCookie(FormsAuthentication.FormsCookieName, encryptedTicket); if (cookieSection.RequireSSL || authenticationSection.Forms.RequireSSL) { authCookie.Secure = true; } HttpContext.Current.Response.Cookies.Add(authCookie); } We read this cookie back in Global.asax and set the Context.User to be a new GenericPrincipal with the roles we assigned earlier. protected void Application_AuthenticateRequest(Object sender, EventArgs e){ if (Context.User != null) { string cookieName = FormsAuthentication.FormsCookieName; HttpCookie authCookie = Context.Request.Cookies[cookieName]; if (authCookie == null) return; FormsAuthenticationTicket authTicket = FormsAuthentication.Decrypt(authCookie.Value); string[] roles = authTicket.UserData.Split(new char[] { '|' }); FormsIdentity fi = (FormsIdentity)(Context.User.Identity); Context.User = new System.Security.Principal.GenericPrincipal(fi, roles); }} We ensure that a user has permissions to view a record by creating a custom attribute AuthorizeToViewID that inherits from ActionFilterAttribute. public class AuthorizeToViewIDAttribute : ActionFilterAttribute{ IEmployeeRepository employeeRepository = new EmployeeRepository(); public override void OnActionExecuting(ActionExecutingContext filterContext) { if (filterContext.ActionParameters.ContainsKey("id") && filterContext.ActionParameters["id"] != null) { if (employeeRepository.IsAuthorizedToView((int)filterContext.ActionParameters["id"])) { return; } } throw new UnauthorizedAccessException("The record does not exist or you do not have permission to access it"); }} We add the AuthorizeToView attribute to any Action method that requires authorization. [HttpPost][Authorize(Order = 1)]//To prevent CSRF[ValidateAntiForgeryToken(Salt = Globals.EditSalt, Order = 2)]//See AuthorizeToViewIDAttribute class[AuthorizeToViewID(Order = 3)] [ActionName("Edit")]public ActionResult Update(int id){ var employeeToEdit = employeeRepository.GetEmployee(id); if (employeeToEdit != null) { //Employees can edit only their address //A manager can edit the title and salary of their subordinate string[] whiteList = (employeeToEdit.IsSubordinate) ? new string[] { "Title", "Salary" } : new string[] { "Address" }; if (TryUpdateModel(employeeToEdit, whiteList)) { employeeRepository.Save(employeeToEdit); return RedirectToAction("Details", new { id = id }); } else { ModelState.AddModelError("", "Please correct the following errors."); } } return View(employeeToEdit);} The Authorize attribute is added to ensure that only authorized users can execute that Action. We use the TryUpdateModel with a white list to ensure that (a) an employee is able to edit only their Address and (b) that a manager is able to edit only the Title and Salary of a subordinate. This works in conjunction with the AuthorizeToViewIDAttribute. The ValidateAntiForgeryToken attribute is added (with a salt) to avoid CSRF. The Order on the attributes specify the order in which the attributes are executed. The Edit View uses the AntiForgeryToken helper to render the hidden token: ......<% using (Html.BeginForm()) {%><%=Html.AntiForgeryToken(NorthwindHR.Models.Globals.EditSalt)%><%= Html.ValidationSummary(true, "Please correct the errors and try again.") %><div class="editor-label"> <%= Html.LabelFor(model => model.LastName) %></div><div class="editor-field">...... The application uses View specific models for ease of model binding. public class EmployeeViewModel{ public int EmployeeID; [Required] [DisplayName("Last Name")] public string LastName { get; set; } [Required] [DisplayName("First Name")] public string FirstName { get; set; } [Required] [DisplayName("Title")] public string Title { get; set; } [Required] [DisplayName("Address")] public string Address { get; set; } [Required] [DisplayName("Salary")] [Range(500, double.MaxValue)] public decimal Salary { get; set; } public bool IsSubordinate { get; set; }} To help with displaying readonly/editable fields, we use a helper method. //Simple extension method to display a TextboxFor or DisplayFor based on the isEditable variablepublic static MvcHtmlString TextBoxOrLabelFor<TModel, TProperty>(this HtmlHelper<TModel> htmlHelper, Expression<Func<TModel, TProperty>> expression, bool isEditable){ if (isEditable) { return htmlHelper.TextBoxFor(expression); } else { return htmlHelper.DisplayFor(expression); }} The helper method is used in the view like so: <%=Html.TextBoxOrLabelFor(model => model.Title, Model.IsSubordinate)%> As mentioned in this post, there is a much easier way to update properties on an object. Download Demo Project VS 2008, ASP.NET MVC 2 RTM Remember to change the connectionString to point to your Northwind DB NorthwindHR.zip Feedback and bugs are always welcome :-)

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  • dhcp-snooping option 82 drops valid dhcp requests on 2610 series Procurve switches

    - by kce
    We are slowly starting to implement dhcp-snooping on our HP ProCurve 2610 series switches, all running the R.11.72 firmware. I'm seeing some strange behavior where dhcp-request or dhcp-renew packets are dropped when originating from "downstream" switches due "untrusted relay information from client". The full error: Received untrusted relay information from client <mac-address> on port <port-number> In more detail we have a 48 port HP2610 (Switch A) and a 24 port HP2610 (Switch B). Switch B is "downstream" of Switch A by virtue of a DSL connection to one of Switch A ports. The dhcp server is connected to Switch A. The relevant bits are as follows: Switch A dhcp-snooping dhcp-snooping authorized-server 192.168.0.254 dhcp-snooping vlan 1 168 interface 25 name "Server" dhcp-snooping trust exit Switch B dhcp-snooping dhcp-snooping authorized-server 192.168.0.254 dhcp-snooping vlan 1 interface Trk1 dhcp-snooping trust exit The switches are set to trust BOTH the port the authorized dhcp server is attached to and its IP address. This is all well and good for the clients attached to Switch A, but the clients attached to Switch B get denied due to the "untrusted relay information" error. This is odd for a few reasons 1) dhcp-relay is not configured on either switch, 2) the Layer-3 network here is flat, same subnet. DHCP packets should not have a modified option 82 attribute. dhcp-relay does appear to be enabled by default however: SWITCH A# show dhcp-relay DHCP Relay Agent : Enabled Option 82 : Disabled Response validation : Disabled Option 82 handle policy : append Remote ID : mac Client Requests Server Responses Valid Dropped Valid Dropped ---------- ---------- ---------- ---------- 0 0 0 0 SWITCH B# show dhcp-relay DHCP Relay Agent : Enabled Option 82 : Disabled Response validation : Disabled Option 82 handle policy : append Remote ID : mac Client Requests Server Responses Valid Dropped Valid Dropped ---------- ---------- ---------- ---------- 40156 0 0 0 And interestingly enough the dhcp-relay agent seems very busy on Switch B, but why? As far as I can tell there is no reason why dhcp requests need a relay with this topology. And furthermore I can't tell why the upstream switch is dropping legitimate dhcp requests for untrusted relay information when the relay agent in question (on Switch B) isn't modifying the option 82 attributes anyway. Adding the no dhcp-snooping option 82 on Switch A allows the dhcp traffic from Switch B to be approved by Switch A, by virtue of just turning off that feature. What are the repercussions of not validating option 82 modified dhcp traffic? If I disable option 82 on all my "upstream" switches - will they pass dhcp traffic from any downstream switch regardless of that traffic's legitimacy? This behavior is client operating system agnostic. I see it with both Windows and Linux clients. Our DHCP servers are either Windows Server 2003 or Windows Server 2008 R2 machines. I see this behavior regardless of the DHCP servers' operating system. Can anyone shed some light on what's happening here and give me some recommendations on how I should proceed with configuring the option 82 setting? I feel like i just haven't completely grokked dhcp-relaying and option 82 attributes.

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  • Update php 5.0 to 5.2.4 with aptitude

    - by Kiva
    Hi guy, I would like to update my php 5 in my server. At this moment, I use php 5.0 so I want to update it to php 5.2.4 (not php 5.3). I tried to do this: aptitude update aptitude upgrade 63 packets were updated but not php which is always in 5.0 How can I update my php please ?

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  • Anti-DDoS Question

    - by Andre
    Our company´s main owner (telecon group) wants us to deploy anti-DDoS mechanisms, such as Arbor Pravail, which is a great idea. Although... I have a question... If our main ISP Backbone provider have no anti-DDoS mechanism, means that there is no point we get the Arbor Pravail? An DDoS attack can make damage uniquely the destination IP or to the whole network that the DDoS packets go through? Regards,

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  • How do I enable multicast routing in Windows XP

    - by Simon Richter
    I have successfully set up a Windows XP machine as an IPv6 router using netsh, that is, it announces prefixes and forwards packets on two interfaces, as verified by pinging. Now I'd like to forward multicast frames between both subnets; hosts on both sides are properly sending out multicast listener reports, so all it would take would be for the router to process these and start forwarding datagrams. How can I enable IPv6 multicast routing between two interfaces?

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  • openstack, bridging, netfilter and dnat

    - by Craig Sanders
    In a recent upgrade (from Openstack Diablo on Ubuntu Lucid to Openstack Essex on Ubuntu Precise), we found that DNS packets were frequently (almost always) dropped on the bridge interface (br100). For our compute-node hosts, that's a Mellanox MT26428 using the mlx4_en driver module. We've found two workarounds for this: Use an old lucid kernel (e.g. 2.6.32-41-generic). This causes other problems, in particular the lack of cgroups and the old version of the kvm and kvm_amd modules (we suspect the kvm module version is the source of a bug we're seeing where occasionally a VM will use 100% CPU). We've been running with this for the last few months, but can't stay here forever. With the newer Ubuntu Precise kernels (3.2.x), we've found that if we use sysctl to disable netfilter on bridge (see sysctl settings below) that DNS started working perfectly again. We thought this was the solution to our problem until we realised that turning off netfilter on the bridge interface will, of course, mean that the DNAT rule to redirect VM requests for the nova-api-metadata server (i.e. redirect packets destined for 169.254.169.254:80 to compute-node's-IP:8775) will be completely bypassed. Long-story short: with 3.x kernels, we can have reliable networking and broken metadata service or we can have broken networking and a metadata service that would work fine if there were any VMs to service. We haven't yet found a way to have both. Anyone seen this problem or anything like it before? got a fix? or a pointer in the right direction? Our suspicion is that it's specific to the Mellanox driver, but we're not sure of that (we've tried several different versions of the mlx4_en driver, starting with the version built-in to the 3.2.x kernels all the way up to the latest 1.5.8.3 driver from the mellanox web site. The mlx4_en driver in the 3.5.x kernel from Quantal doesn't work at all) BTW, our compute nodes have supermicro H8DGT motherboards with built-in mellanox NIC: 02:00.0 InfiniBand: Mellanox Technologies MT26428 [ConnectX VPI PCIe 2.0 5GT/s - IB QDR / 10GigE] (rev b0) we're not using the other two NICs in the system, only the Mellanox and the IPMI card are connected. Bridge netfilter sysctl settings: net.bridge.bridge-nf-call-arptables = 0 net.bridge.bridge-nf-call-iptables = 0 net.bridge.bridge-nf-call-ip6tables = 0 Since discovering this bridge-nf sysctl workaround, we've found a few pages on the net recommending exactly this (including Openstack's latest network troubleshooting page and a launchpad bug report that linked to this blog-post that has a great description of the problem and the solution)....it's easier to find stuff when you know what to search for :), but we haven't found anything on the DNAT issue that it causes.

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  • Two network adapters on Ubuntu Server 9.10 - Can't have both working at once?

    - by Rob
    I'm trying to set up two network adapters in Ubuntu (server edition) 9.10. One for the public internet, the other a private LAN. During the install, I was asked to pick a primary network adapter (eth0 or eth1). I chose eth0, gave the installer the details listed below in the contents of /etc/network/interfaces, and carried on. I've been using this adapter with these setting for the last few days, and every thing's been fine. Today, I decide it's time to set up the local adapter. I edit the /etc/network/interfaces to add the details for eth1 (see below), and restart networking with sudo /etc/init.d/networking restart. After this, attempting to ping the machine using it's external IP address fails, but I can ping it's local IP address. If I bring eth1 down using sudo ifdown eth1, I can successfully ping the machine via it's external IP address again (but obviously not it's internal IP address). Bringing eth1 back up returns us to the original problem state: external IP not working, internal IP working. Here's my /etc/network/interfaces (I've removed the external IP information, but these settings are unchanged from when it worked) rob@rhea:~$ cat /etc/network/interfaces # This file describes the network interfaces available on your system # and how to activate them. For more information, see interfaces(5). # The loopback network interface auto lo iface lo inet loopback # The primary (public) network interface auto eth0 iface eth0 inet static address xxx.xxx.xxx.xxx netmask xxx.xxx.xxx.xxx network xxx.xxx.xxx.xxx broadcast xxx.xxx.xxx.xxx gateway xxx.xxx.xxx.xxx # The secondary (private) network interface auto eth1 iface eth1 inet static address 192.168.99.4 netmask 255.255.255.0 network 192.168.99.0 broadcast 192.168.99.255 gateway 192.168.99.254 I then do this: rob@rhea:~$ sudo /etc/init.d/networking restart * Reconfiguring network interfaces... [ OK ] rob@rhea:~$ sudo ifup eth0 ifup: interface eth0 already configured rob@rhea:~$ sudo ifup eth1 ifup: interface eth1 already configured Then, from another machine: C:\Documents and Settings\Rob>ping [external ip] Pinging [external ip] with 32 bytes of data: Request timed out. Request timed out. Request timed out. Request timed out. Ping statistics for [external ip]: Packets: Sent = 4, Received = 0, Lost = 4 (100% loss), Back on the Ubuntu server in question: rob@rhea:~$ sudo ifdown eth1 ... and again on the other machine: C:\Documents and Settings\Rob>ping [external ip] Pinging [external ip] with 32 bytes of data: Reply from [external ip]: bytes=32 time<1ms TTL=63 Reply from [external ip]: bytes=32 time<1ms TTL=63 Reply from [external ip]: bytes=32 time<1ms TTL=63 Reply from [external ip]: bytes=32 time<1ms TTL=63 Ping statistics for [external ip]: Packets: Sent = 4, Received = 4, Lost = 0 (0% loss), Approximate round trip times in milli-seconds: Minimum = 0ms, Maximum = 0ms, Average = 0ms So... what am I doing wrong?

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  • Overhead of TLS/SSL on a TCP socket connection?

    - by TK Kocheran
    Is there any bandwidth overhead on using SSL on a TCP connection? I understand, of course, the processing/memory usage overhead in encrypting and decrypting packets, but as far as bandwidth is concerned, what is the difference, if any? For example, given a XML file which is 64KB, will there be any tangible difference in the transfer size of the file over HTTP vs. HTTPS? (Ignoring mod_deflate and mod_gzip, of course)

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  • checksum in raw sockets and pcap

    - by hero
    i am using pcap library to sniff some packets, change their tcp data , and then inject my packet on the network. my question is: if i changed in the tcp data, should i recalculate the length field in the tcp header? should i also change the checksum? i read in a page on how to create raw sockets that if you set the tcp_checksum to 0, the kernel will automatically calculate it and fill it, is this true for windows machines also?

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