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• Base 128 or 256 Encoding for the Binary Lexical Octet Adhoc Transport Protocol?

  - by Randolpho

  I'm in the process of implementing a network driver for the Binary Lexical Octet Adhoc Transport (BLOAT) protocols in the hopes of replacing the TCP/UDP/IP stack with a much more flexible XML structure. BLOAT is detailed in RFC 3252, so if you're unfamiliar with the protocol I highly recommend you read the entire RFC before providing any comments. Don't worry, it's short and sweet; you might even enjoy it. Anyway, my problem is this: BLOAT requires that the payload be Base64 encoded which doesn't make sense to me. I mean, sure, it's the internet standard for binary payloads, but there are better, more efficient encodings available: Base128 and Base256, for example. That the RFC requires Base64 and doesn't allow for any other payload encoding really bothers me. To that end, I'm considering a small optional change to the protocol. Embrace and extend, right? Anyway, I'd like to modify the payload element to accept an encoding attribute, which can extend the encoding to Base128 or Base256, or even to other encodings I can't conceive of at the moment. If the encoding attribute isn't present, Base64 would be assumed. So my question is this: should I? I mean, BLOAT is an accepted standard, even if it isn't exactly omnipresent. If I make this change, will there be compatibility issues? I don't foresee any, but perhaps you, oh great Stack Overflow Community, can? If I do implement this change, should I contact the original RFC author? Should I offer a supplemental RFC?

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• Problem with non blocking fifo in bash

  - by timdel

  Hi! I'm running a few Team Fortress 2 servers and I want to write a little management script. Basically the TF2 servers are a fg process which provides a server console, so I can start the server, type status and get an answer from it: ***@purple:~/tf2$ ./start_server_testing Auto detecting CPU Using AMD Optimised binary. Server will auto-restart if there is a crash. Console initialized. [bla bla bla] Connection to Steam servers successful. VAC secure mode is activated. status hostname: Team Fortress version : 1.0.6.1/15 3883 secure udp/ip : ***.***.133.31:27600 map : ctf_2fort at: 0 x, 0 y, 0 z players : 0 (2 max) # userid name uniqueid connected ping loss state adr Great, now I want to create a script which sends the command sm_reloadadmins to all my servers. The best way I found to do this is using a fifo named pipe. Now what I want to do is having this pipe readonly and non blocking to the server process, so I can write into the pipe and the server executes it, but still I want to write via console one the server, so if I switch back to the fg process of the server and I type status I want an answer printed. I tried this (assuming serverfifo is mkfifo serverfifo): ./start_server_testing < serverfifo Not working, the server won't start until something is written to the pipe. ./start_server_testing <> serverfifo Thats actually working pretty good, I can see the console output of the server and I can write to the fifo and the server executes the commands, but I can't write via console to the server anymore. Also, if I write 'exit' to the pipe (which should end the server) and I'm running it in a screen the screen window is getting killed for some reason (wtf why?). I only need the server to read the fifo without blocking AND all my keyboard input on the server itself should be send to the server AND all server ouput should be written to the console. Is that possible? If yes, how?

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• what is the best way to stream a audio file to website users/listners

  - by Naveen Chamikara Gamage

  I'm developing a music site which will stream audio files stored in a server to users, audio files will be played through flash player placed in a webpage.. As I heard I need to use a streaming media server for streaming audio files ( like 2mb to 3mb in size).. Do I need to use one? I found some streaming media server softwares like http://www.icecast.org - but as in their documentation, It is used for streaming radio stations and live streaming purposes, but I just need to stream audio files faster and in low size (low bandwidth) with good quality.. I heard I need to encode the audio files first and then send them to listeners and in their end audio files need to be decoded again. Is that true? How can I do that? if I need to use a special web server, where should I host my files? Any good hosting providers? if I host audio files in a normal web server, they will use HTTP or TCP to deliver my audio files to users/ listners but I found that HTTP and TCP are not good ways to use for multi media purposes like streaming audio and video files, and they are used for delivering HTML and stuff. I found I should use RSTP or UDP for streaming audio files.. What should I use? I know that .MP3 files has much better quality than the other formats but it also gives huge size to the audio files.. which format should I use for audio files? Most of the best quality audio files are more than 7mb so I'm planning to convert them my self using a software so I could get low size files with some level of good quality. If I'm converting my audio files what is the good BITRATE I should use for my files? Any known best softwares for converting audio files while keeping quality in a good level? Note** - I know that I will not need complex requirements at the beginning of the site but I wanted to what are the best ways like they are using for soundcloud.com

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• Experiences with (free) embedded TCP / IP stacks?

  - by Dan

  Does anyone have especially good (or bad) experiences with any of the following embedded TCP / IP stacks? uIP lwIP Bentham's TCP/IP Lean implementation The TCP/IP stack from this book My needs are for a solid, easy-to-port stack. Code size isn't terribly important, performance is relatively important, but ease of use & porting is very important. The system will probably use an RTOS, that hasn't been decided, but in my experience most stacks can be used with or without an RTOS. Most likely the platform will be an ARM variant (ARM7 or CM3 in all likelihood). Not too concerned about bolting the stack to the Ethernet driver, so that isn't a big priority in the selection. I'm not terribly interested in extracting a stack out of an OS, such as Linux, RTEMS, etc. I'm also not interested in commercial offerings such as Interniche, Micrium, etc... The stack doesn't need all sorts of bells & whistles, doesn't need IPv6, and I don't need any stuff on top of it (web servers, FTP servers, etc..) In fact it's possible that I'll only use UDP, although I can envision a couple scenarios where TCP would be preferable. Experiences with other stacks I've missed are of course also very much of interest. Thanks for your time & input.

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• Android Hashtable Serialization

  - by Nsyed

  Hi All, I am having a weird issue with serialization of a Hashtable. I have made a Server, Client app. Where server(PC/MAC) is serializing a Hashtable and sending it to Client(Android) through UDP. The data is sent/read correctly but I get a bunch of these messages below on LogCat. 04-12 11:19:43.059: DEBUG/dalvikvm(407): GetFieldID: unable to find field Ljava/util/Hashtable;.loadFactor:F Occasionally, I would see these 04-12 11:21:19.150: DEBUG/dalvikvm(407): GC freed 10814 objects / 447184 bytes in 97ms The app would run for 2-3 mins and then crash. Interestingly enough I do not see the Loadfactor errors on SDK 1.5. But I do see the GC Free xxxx objects, quiet often. After debugging I have found that the issue is with de-serialization and the error/warning are coming from following code Code: ByteArrayInputStream bis = new ByteArrayInputStream(bytes); ObjectInputStream ois = new ObjectInputStream(bis); object = ois.readObject(); at Code: object = ois.readObject(); on the client. My server is serializing code is the following. Code: ByteArrayOutputStream bos = new ByteArrayOutputStream(); ObjectOutputStream oos = new ObjectOutputStream(bos); oos.writeObject(obj); Any idea what is going on? Thanks for the Help!

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• moving audio over a local network using GStreamer

  - by James Turner

  I need to move realtime audio between two Linux machines, which are both running custom software (of mine) which builds on top of Gstreamer. (The software already has other communication between the machines, over a separate TCP-based protocol - I mention this in case having reliable out-of-band data makes a difference to the solution). The audio input will be a microphone / line-in on the sending machine, and normal audio output as the sink on the destination; alsasrc and alsasink are the most likely, though for testing I have been using the audiotestsrc instead of a real microphone. GStreamer offers a multitude of ways to move data round over networks - RTP, RTSP, GDP payloading, UDP and TCP servers, clients and sockets, and so on. There's also many examples on the web of streaming both audio and video - but none of them seem to work for me, in practice; either the destination pipeline fails to negotiate caps, or I hear a single packet and then the pipeline stalls, or the destination pipeline bails out immediately with no data available. In all cases, I'm testing on the command-line just gst-launch. No compression of the audio data is required - raw audio, or trivial WAV, uLaw or aLaw encoding is fine; what's more important is low-ish latency.

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• Android Multiple Handlers Design Question

  - by Soumya Simanta

  This question is related to an existing question I asked. I though I'll ask a new question instead of replying back to the other question. Cannot "comment" on my previous question because of a word limit. Marc wrote - I've more than one Handlers in an Activity." Why? If you do not want a complicated handleMessage() method, then use post() (on Handler or View) to break the logic up into individual Runnables. Multiple Handlers makes me nervous. I'm new to Android. Is having multiple handlers in a single activity a bad design ? I'm new to Android. My question is - is having multiple handlers in a single activity a bad design ? Here is the sketch of my current implementation. I've a mapActivity that creates a data thread (a UDP socket that listens for data). My first handler is responsible for sending data from the data thread to the activity. On the map I've a bunch of "dynamic" markers that are refreshed frequently. Some of these markers are video markers i.e., if the user clicks a video marker, I add a ViewView that extends a android.opengl.GLSurfaceView to my map activity and display video on this new vide. I use my second handler to send information about the marker that the user tapped on ItemizedOverlay onTap(int index) method. The user can close the video view by tapping on the video view. I use my third handler for this. I would appreciate if people can tell me what's wrong with this approach and suggest better ways to implement this. Thanks.

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• Advice on designing and building distributed application to track vehicles

  - by dario-g

  I'm working on application for tracking vehicles. There will be about 10k or more vehicles. Each will be sending ~250bytes in each minute. Data contains gps location and everything from CAN Bus (every data that we can read from vehicle computer and dashboard). Data are sent by GSM/GPRS (using UDP protocol). Estimated rows with this data per day is ~2000k. I see there 3 main blocks. 1. Multithreaded Socket Server (MSS) - I have it. MSS stores received data to the queue (using NServiceBus). 2. Rule Processor Server (RPS) - this is core of this system. This block is responsible for parsing received data, storing in the database, processing rules, sending messages to Notifier Server (this will be sending e-mails/sms texts). Rule example. As I said between received bytes there will be information about current speed. When speed will be above 120 then: show alert in web application for specified users, send e-mail, send sms text. (There can be more than one instance of RPS). 3. Web application - allows reporting and defining rules by users, monitoring alerts, etc. I'm looking for advice how to design communication between RPS and Web application. Some questions: - Should Web application and RPS have separated databases or one central database will be enough? I have one domain model in web application. If there will be one central database then can I use the same model (objects) on RPS? So, how to send changed rules to RPS? I try to decouple this blocks as much as possible. I'm planning to create different instance of application for each client (each client will have separated database). One client will be have 10k vehicles, others only 100 vehicles.

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• Is there the equivalent of cloud computing for modems?

  - by morpheous

  I asked this question on SF, and someone recommended that I ask it here - (I don't think I have enough points to move a question from SF to SO - and in any case, I don't know how to do it - so here is the question again): I am interested in the concept of PAAS (platform as a service). However, all talk about SAAS/PAAS seems to focus on only the computer itself - not its peripherals. Is it possible to 'outsource' modems as a resource - so that an app running remotely can pump data to a modem in the cloud? As a bit of background to the question, a group of us are thinking of starting a company that offers similar services to companies like twilio etc - but I want to 'outsource' both the computing hardware (thats PAAS - the easy bit) and the modems (thats what I cant seem to find any info on). Does anyone know if modems can be bundled as part of a PAAS service? - alternatively, is there a way that an application running on one computer can communicate (i.e. pump data) to a remote modem residing on another machine?. I assume I can come up with some protocol over UDP or TCP - but there is no point reinventing the wheel - if such a protocol like that already exists (or if it some open source software allows one to do this). Any suggestions on how to solve this problem?

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• How to properly close a UDT server in Netty 4

  - by Steffen

  I'm trying to close my UDT server (Netty 4.0.5.Final) with shutDownGracefully() and reopen it on the same port. Unfortunately, I always get the socket exception below although it waits until the future has completed. I also added the socket option SO_REUSEADDR. What is the proper way to do this? Exception in thread "main" com.barchart.udt.ExceptionUDT: UDT Error : 5011 : another socket is already listening on the same UDP port : listen0:listen [id: 0x323d3939] at com.barchart.udt.SocketUDT.listen0(Native Method) at com.barchart.udt.SocketUDT.listen(SocketUDT.java:1136) at com.barchart.udt.net.NetServerSocketUDT.bind(NetServerSocketUDT.java:66) at io.netty.channel.udt.nio.NioUdtAcceptorChannel.doBind(NioUdtAcceptorChannel.java:71) at io.netty.channel.AbstractChannel$AbstractUnsafe.bind(AbstractChannel.java:471) at io.netty.channel.DefaultChannelPipeline$HeadHandler.bind(DefaultChannelPipeline.java:1006) at io.netty.channel.DefaultChannelHandlerContext.invokeBind(DefaultChannelHandlerContext.java:504) at io.netty.channel.DefaultChannelHandlerContext.bind(DefaultChannelHandlerContext.java:487) at io.netty.channel.ChannelDuplexHandler.bind(ChannelDuplexHandler.java:38) at io.netty.handler.logging.LoggingHandler.bind(LoggingHandler.java:254) at io.netty.channel.DefaultChannelHandlerContext.invokeBind(DefaultChannelHandlerContext.java:504) at io.netty.channel.DefaultChannelHandlerContext.bind(DefaultChannelHandlerContext.java:487) at io.netty.channel.DefaultChannelPipeline.bind(DefaultChannelPipeline.java:848) at io.netty.channel.AbstractChannel.bind(AbstractChannel.java:193) at io.netty.bootstrap.AbstractBootstrap$2.run(AbstractBootstrap.java:321) at io.netty.util.concurrent.SingleThreadEventExecutor.runAllTasks(SingleThreadEventExecutor.java:354) at io.netty.channel.nio.NioEventLoop.run(NioEventLoop.java:366) at io.netty.util.concurrent.SingleThreadEventExecutor$2.run(SingleThreadEventExecutor.java:101) at java.lang.Thread.run(Thread.java:724) A small test program demonstration the problem: public class MsgEchoServer { public static class MsgEchoServerHandler extends ChannelInboundHandlerAdapter { } public void run() throws Exception { final ThreadFactory acceptFactory = new UtilThreadFactory("accept"); final ThreadFactory connectFactory = new UtilThreadFactory("connect"); final NioEventLoopGroup acceptGroup = new NioEventLoopGroup(1, acceptFactory, NioUdtProvider.MESSAGE_PROVIDER); final NioEventLoopGroup connectGroup = new NioEventLoopGroup(1, connectFactory, NioUdtProvider.MESSAGE_PROVIDER); try { final ServerBootstrap boot = new ServerBootstrap(); boot.group(acceptGroup, connectGroup) .channelFactory(NioUdtProvider.MESSAGE_ACCEPTOR) .option(ChannelOption.SO_BACKLOG, 10) .option(ChannelOption.SO_REUSEADDR, true) .handler(new LoggingHandler(LogLevel.INFO)) .childHandler(new ChannelInitializer<UdtChannel>() { @Override public void initChannel(final UdtChannel ch) throws Exception { ch.pipeline().addLast(new MsgEchoServerHandler()); } }); final ChannelFuture future = boot.bind(1234).sync(); } finally { acceptGroup.shutdownGracefully().syncUninterruptibly(); connectGroup.shutdownGracefully().syncUninterruptibly(); } new MsgEchoServer().run(); } public static void main(final String[] args) throws Exception { new MsgEchoServer().run(); } }

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• Cross-Platform Language + GUI Toolkit for Prototyping Multimedia Applications

  - by msutherl

  I'm looking for a language + GUI toolkit for rapidly prototyping utility applications for multimedia installations. I've been working with Max/MSP/Jitter for many years, but I'd like to add a text-based language to my 'arsenal' for tasks apart from 'content production'. (When it comes to actual media synthesis, my choices are clear [SuperCollider + MSP for audio, Jitter + Quartz + openFrameworks for video]). I'm looking for something that maintains some of the advantages of Max, but is lower-level, faster, more cross-platfrom (Linux support), and text-based. Integration with powerful sound/video libraries is not a requirement. Some requirements: Cross-platform (at least OSX and Linux, Windows is a plus) Fast and easy cross-platform GUIs with no platform-specific modification GUI code separated from backend code as much as possible Good for interfacing with external serial devices (micro-controllers) Good network support (UDP/TCP) Good libraries for multi-media (video, sound, OSC) are a plus Asynchronous synchronous UNIX integration is a plus The options that come to mind: AS3/Flex (not a fan of AS3 or the idea of running in the Flash Player) openFrameworks (C++ framework, perhaps a bit too low level [looking for fast development time] and biased toward video work) Java w/ Processing libraries (like openFrameworks, just slower) Python + Qt (is Qt appropriate for rapid prototyping?) Python + Another GUI toolkit SuperCollider + Swing (yucky GUI development) Java w/ SWT Any other options? What do you recommend?

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• help on ejb stateless datagram and message driven beans

  - by Kemmal

  i have a client thats sending a message to the ejbserver using UDP, i want the server(stateless bean) to echo back this message to the client but i cant seem to do this. or can i implement the same logic by using JMS? please help and enlighten. this is just a test, in the end i want a midp to be sending the message to the ejb using datagrams. here is my code. @Stateless public class SessionFacadeBean implements SessionFacadeRemote { public SessionFacadeBean() { } public static void main(String[] args) { DatagramSocket aSocket = null; byte[] buffer = null; try { while(true) { DatagramPacket request = new DatagramPacket(buffer, buffer.length); aSocket.receive(request); DatagramPacket reply = new DatagramPacket(request.getData(), request.getLength(), request.getAddress(), request.getPort()); aSocket.send(reply); } } catch (SocketException e) { System.out.println("Socket: " + e.getMessage()); } catch (IOException e) { System.out.println("IO: " + e.getMessage()); } finally { if(aSocket != null) aSocket.close(); } } } and the client: public static void main(String[] args) { DatagramSocket aSocket = null; try { aSocket = new DatagramSocket(); byte [] m = "Test message!".getBytes(); InetAddress aHost = InetAddress.getByName("localhost"); int serverPort = 6789; DatagramPacket request = new DatagramPacket(m, m.length, aHost, serverPort); aSocket.send(request); byte[] buffer = new byte[1000]; DatagramPacket reply = new DatagramPacket(buffer, buffer.length); aSocket.receive(reply); System.out.println("Reply: " + new String(reply.getData())); } catch (SocketException e) { System.out.println("Socket: " + e.getMessage()); } catch (IOException e) { System.out.println("IO: " + e.getMessage()); } finally { if(aSocket != null) aSocket.close(); } } please help.

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• Netty options for real-time distribution of small messages to a large number of clients?

  - by user439407

  I am designing a (near) real-time Netty server to distribute a large number of very small messages to a large number of clients across the internet. In internal, go as fast as you can testing, I found that I could do 10k clients no sweat, but now that we are trying to go across the internet, where the latency, bandwidth etc varies pretty wildly we are running into the dreaded outOfMemory issues, even with 2 gigs of RAM. I have tried various workarounds(setting the socket stack sizes smaller, setting high and low water marks, cancelling things that are too old), and they help a little, but they seem to only help a little bit. What would some good ways to optimize Netty for sending large #s of small messages without significant delays? Also, the bulk of the message only consists of one kind of message that I don't particularly care if it doesn't arrive. I would use UDP but because we don't control the client, thats not really a possibility. Is it possible to set a separate timeout solely for this kind of message without affecting the other messages? Any insight you could offer would be greatly appreciated.

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• How does the socket API accept() function work?

  - by Eli Bendersky

  The socket API is the de-facto standard for TCP/IP and UDP/IP communications (that is, networking code as we know it). However, one of its core functions, accept() is a bit magical. To borrow a semi-formal definition: accept() is used on the server side. It accepts a received incoming attempt to create a new TCP connection from the remote client, and creates a new socket associated with the socket address pair of this connection. In other words, accept returns a new socket through which the server can communicate with the newly connected client. The old socket (on which accept was called) stays open, on the same port, listening for new connections. How does accept work? How is it implemented? There's a lot of confusion on this topic. Many people claim accept opens a new port and you communicate with the client through it. But this obviously isn't true, as no new port is opened. You actually can communicate through the same port with different clients, but how? When several threads call recv on the same port, how does the data know where to go? I guess it's something along the lines of the client's address being associated with a socket descriptor, and whenever data comes through recv it's routed to the correct socket, but I'm not sure. It'd be great to get a thorough explanation of the inner-workings of this mechanism.

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• Accessing functions in a swf-file through javascript [chatroulette.com]

  - by RadiantHeart

  Lately I have been interested in the code behind chatroulette.com. As you probably know it is a peer-to-peer webcam-chat-service written in actionscript, as I understand. What i have been wondering about is weather its possible to extract the ip-address of whomever you are currently communicating with. I have seen services that do that, but they require that you install a program that runs alongside on your computer sniffing UDP-packages. I was wondering if there was a simpler method. What I do know is that the javascript on the page communicates with the application via "ExternalInterface". On this area I am pretty much a novice but according to my limited understanding you cant get information from the flash-application unless you have configured a listener for a call from javascript and then attach a callback to that event. Is this correct or can you access public functions and variables directly through javascript? There is for example a public function like this: public function get outgoingAddress():String{ return (this.__info.outgoingAddress); } Can it be accessed directly through javascript? If it cant be done so easily, is it possible to decompile the .swf-file, change it (add some functions) and recompile it and run it instead? I am hoping someone can satisfy my curiosity here. Here are two links to a decompiled version of the swf-file. The first with line numbering and one without. With line numbering ˜ 3.5 Mbyte Without line numbering ˜ 2.1 Mbyte

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• A local error has occurred while connecting to AD in Windows 2008 server

  - by Sara

  There's Active directory on windows 2000 advance server, I have a web server on Windows 2008 server Enterprise Edition, the following code works fine in Winsows 2003 server but when I installed Win 2008 server, it gives me the following error, the webserver is not subdomain of the AD server. but they have the same range IP address. A local error has occurred.\r\n"} System.Exception system.DirectoryServices.DirectoryServicesCOMException} I want to Authenticate Via AD from my webserver, I even test the port 389 and it was open(by telnet), I even added port 389 UDP and TCP to firewall of webserver to be sure it is open, even I turned the firewall off but nothing changed. I don't know what's wrong with Windows 2008 server that cannot run my code, I search Internet but I found nothing. any solution would be helpful. Thank you public bool IsAuthenticated(string username, string pwd,string group) { string domainAndUsername = "LDAP://192.xx.xx.xx:389/DC=test,DC=oc,DC=com" ; string usr="CN=" + username + ",CN=" + group; DirectoryEntry entry = new DirectoryEntry(domainAndUsername, usr, pwd, AuthenticationTypes.Secure ); try { DirectorySearcher search = new DirectorySearcher(entry); search.Filter = "(SAMAccountName=" + username + ")"; SearchResult result = search.FindOne(); if (result == null) { return false; } } catch (Exception ex) { return false; } return true; }

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• protocol parsing in c

  - by nomad.alien

  I have been playing around with trying to implement some protocol decoders, but each time I run into a "simple" problem and I feel the way I am solving the problem is not optimal and there must be a better way to do things. I'm using C. Currently I'm using some canned data and reading it in as a file, but later on it would be via TCP or UDP. Here's the problem. I'm currently playing with a binary protocol at work. All fields are 8 bits long. The first field(8bits) is the packet type. So I read in the first 8 bits and using a switch/case I call a function to read in the rest of the packet as I then know the size/structure of it. BUT...some of these packets have nested packets inside them, so when I encounter that specific packet I then have to read another 8-16 bytes have another switch/case to see what the next packet type is and on and on. (Luckily the packets are only nested 2 or 3 deep). Only once I have the whole packet decoded can I handle it over to my state machine for processing. I guess this can be a more general question as well. How much data do you have to read at a time from the socket? As much as possible? As much as what is "similar" in the protocol headers? So even though this protocol is fairly basic, my code is a whole bunch of switch/case statements and I do a lot of reading from the file/socket which I feel is not optimal. My main aim is to make this decoder as fast as possible. To the more experienced people out there, is this the way to go or is there a better way which I just haven't figured out yet? Any elegant solution to this problem?

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• What library can I use to do simple, lightweight message passing?

  - by Mike

  I will be starting a project which requires communication between distributed nodes(the project is in C++). I need a lightweight message passing library to pass very simple messages(basically just strings of text) between nodes. The library must have the following characteristics: No external setup required. I need to be able to get everything up-and-running in my code - I don't want to require the user to install any packages or edit any configuration files(other than a list of IP addresses and ports to connect to). The underlying protocol which the library uses must be TCP(or if it is UDP, the library must guarantee the eventual receipt of the message). The library must be able to send and receive arbitrarily large strings(think up to 3GB+). The library needn't support any security mechanisms, fault tolerance, or encryption - I just need it to be fast, simple, and easy to use. I've considered MPI, but concluded it would require too much setup on the user's machine for my project. What library would you recommend for such a project? I would roll my own, but due to time constraints, I don't think that will be feasible.

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• C#/.NET: FInd out whether a server exists, Query DNS for SVC records.

  - by TomTom

  Writing a cleint/Server tool I am tasked trying to find a server to connect to. I would love to make things as easy as possible for the user. As such, my idea is to: CHeck whether specific servers (coded by name) exist (like "mail.xxx" for a mail server, for example - my exampüle is not a mail server;) Query otherwise for DNS SVC records, allowing the admin to configure a server location for a specific serivce (that the client connects to). The result is that the user may have to enter only a domain name, possibly even not even that (using the registered standard domain of the computer in a LAN environment). Anyone ideas how: To find out whether a server exists and answers (i.e. is online) in the fastest way? TCP can take a long time if the server is not there. A UDP style ping sounds like a good idea to me. PING itself may be unavailable. Anyonw knows how to ask from withint .NET best for a SVC record in a specific (the default) domain?

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• sendto is returning ENOBUF

  - by user338159

  Hi, I am currently running an old system on Tru64 which involves lots of UDP sockets using the sendto() function. The sockets are used in our code to send messages to/from various processes and then eventually on to a thick client app that is connected remotely. Occasionally the socket to the thick client gets stuck, this can cause some of these messages to get built up. My question is how can I determine the current buffer size, and how do I determine the maximum message buffer. The code below gives a snippet of how I set up the port and use the sendto function. /* need to adjust the maximum size we can send on this / / as it needs to be able to cope with the biggest / / messages we send / lenlen = sizeof(len) ; / allow double for when the system is under load */ len = 2 * C_MAX_MESSAGE_DATA_SIZE ; lpos_setsockopt(FATAL, msg_socket,SOL_SOCKET, SO_SNDBUF, &len, lenlen, &error_no) ; result = sendto( msg_socket, (char *)message, (int)message_len, flags, dest_addr, addrlen); Note. We have ported this application to Linux and the problem does not seem to appear there. Any help would be greatly appreciated. Regards

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• Audio Streaming Latency

  - by killianmcc

  I'm writing a UDP local area network video chat system and have got the video and audio streams working. However I'm experiencing a little latency (about half a second) in the audio and was wondering what codecs would provide the least latency. I'm using NAudio (http://naudio.codeplex.com/) which provides me access to the following codecs for streaming; Speex Narrow Band (VBR) Speex Wide Band (16kHz)(VBR) Speex Ultra Wide Band (32kHz)(VBR) DSP Group TrueSpeech (8.5kbps) GSM 6.10 (13kbps) Microsoft ADPCM (32.8kbps) G.711 a-law (64kbps) G.722 16kHz (64kbps) G.711 mu-law (64kbps) PCM 8kHz 16 bit uncompressed (128kbps) I've tried them out and I'm not noticing much difference. Is there any others that I should download and try to reduce latency? I'm only going to be sending voice over the connection but I'm not really worried about quality or background noises too much. UPDATE I'm sending the audio in blocks like so; waveIn = new WaveIn(); waveIn.BufferMilliseconds = 50; waveIn.DeviceNumber = inputDeviceNumber; waveIn.WaveFormat = codec.RecordFormat; waveIn.DataAvailable += waveIn_DataAvailable; void waveIn_DataAvailable(object sender, WaveInEventArgs e) { if (connected) { byte[] encoded = codec.Encode(e.Buffer, 0, e.BytesRecorded); udpSender.Send(encoded, encoded.Length); } }

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• How often should network traffic/collisions cause SNMP Sets to fail?

  - by A. Levy

  My team has a situation where an SNMP SET will fail once every two weeks or so. Since this set happens automatically, we don't necessarily notice it immediately when it fails, and this can result in an inconsistent configuration and associated wailing and gnashing of teeth. The plan is to fix this by having our software automatically retry the SET when it fails. The problem is, we aren't sure why the failure is happening. My (extremely limited) knowledge of SNMP isn't particularly helpful in diagnosing this problem, so I thought I'd ask StackOverflow for some advice. We think that every so often a spike in network traffic will cause the SET to fail. Since SNMP uses UDP for communication, I would think it would be relatively easy for a command to be drowned out if traffic was high for a short period of time. However, I have no idea how common this is. We have a small network with a single cisco router and there are less than a dozen SNMP controlled devices on that network. In addition to the SNMP traffic, there are some status web pages being loaded from the various devices. In case it makes a difference, I believe we are using the AdventNet SNMP API version 4.0.4 for Java. Does it sound reasonable that there will be some SET commands dropped occasionally, or should we be looking for other causes?

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• Perl, Net::Traceroute::PurePerl return value

  - by John R

  This is a sub routine that I copied from CPAN. It works fine as it is when I run it from the command line. I have a similar function from Net::Traceroute that also works fine AND allows me to return the string with a SOAP call. The problem comes when I try to return the ~string(?) from the function below with a SOAP call. sub tr { use Net::Traceroute::PurePerl; my $t = new Net::Traceroute::PurePerl( backend => 'PurePerl', # this optional host => 'www.whatever.com', debug => 0, max_ttl => 30, query_timeout => 2, packetlen => 40, protocol => 'udp', # Or icmp ); $t->traceroute; $t->pretty_print; return $t; #print $t; } The output looks like a string except the last part of the string looks like this: 28 * * * 29 * * * 30 * * * Net::Traceroute::PurePerl=HASH(0x11fa6bf0) I don't know what is different about Net::Traceroute::PurePerl that won't allow me to return the value with SOAP since the Net::Traceroute version does allow me to return it with SOAP.

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• Java loop to collect the second and third elements every three in an array

  - by mhollander38

  I have a file with data in the form timestamp, coordinate, coordinate, seperated by spaces, as here; 14:25:01.215 370.0 333.0 I need to loop through and add the coordinates only to an array. The data from the file is read in and put into as String[] called info, from split(" "). I have two problems, I think the end of the file has a extra " " which I need to lose appropriately and I also want confirmation/suggestions of my loop, at the moment I am getting sporadic out of bounds exceptions. My loop is as follows; String[] info; info = dataHolder.split(" "); ArrayList<String> coOrds1 = new ArrayList<String>(); for (int counter = 0; counter < info.length; counter = counter+3) { coOrds1.add(info[counter+1]); coOrds1.add(info[counter+2]); } Help and suggestions appreciated. The text file is here but the class receives in a UDP packet from another class so I am unsure if this potentially adds " " at the end or not.

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• Encouter error "Linux ip -6 addr add failed" while setting up OpenVPN client

  - by Mickel

  I am trying to set up my router to use OpenVPN and have gotten quite far (I think), but something seems to be missing and I am not sure what. Here is my configuration for the client: client dev tun proto udp remote ovpn.azirevpn.net 1194 remote-random resolv-retry infinite auth-user-pass /tmp/password.txt nobind persist-key persist-tun ca /tmp/AzireVPN.ca.crt remote-cert-tls server reneg-sec 0 verb 3 OpenVPN client log: Nov 8 15:45:13 rc_service: httpd 15776:notify_rc start_vpnclient1 Nov 8 15:45:14 openvpn[27196]: OpenVPN 2.3.2 arm-unknown-linux-gnu [SSL (OpenSSL)] [LZO] [EPOLL] [MH] [IPv6] built on Nov 1 2013 Nov 8 15:45:14 openvpn[27196]: NOTE: the current --script-security setting may allow this configuration to call user-defined scripts Nov 8 15:45:14 openvpn[27196]: Socket Buffers: R=[116736->131072] S=[116736->131072] Nov 8 15:45:14 openvpn[27202]: UDPv4 link local: [undef] Nov 8 15:45:14 openvpn[27202]: UDPv4 link remote: [AF_INET]178.132.75.14:1194 Nov 8 15:45:14 openvpn[27202]: TLS: Initial packet from [AF_INET]178.132.75.14:1194, sid=44d80db5 8b36adf9 Nov 8 15:45:14 openvpn[27202]: WARNING: this configuration may cache passwords in memory -- use the auth-nocache option to prevent this Nov 8 15:45:14 openvpn[27202]: VERIFY OK: depth=1, C=RU, ST=Moscow, L=Moscow, O=Azire Networks, OU=VPN, CN=Azire Networks, name=Azire Networks, [email protected] Nov 8 15:45:14 openvpn[27202]: Validating certificate key usage Nov 8 15:45:14 openvpn[27202]: ++ Certificate has key usage 00a0, expects 00a0 Nov 8 15:45:14 openvpn[27202]: VERIFY KU OK Nov 8 15:45:14 openvpn[27202]: Validating certificate extended key usage Nov 8 15:45:14 openvpn[27202]: ++ Certificate has EKU (str) TLS Web Server Authentication, expects TLS Web Server Authentication Nov 8 15:45:14 openvpn[27202]: VERIFY EKU OK Nov 8 15:45:14 openvpn[27202]: VERIFY OK: depth=0, C=RU, ST=Moscow, L=Moscow, O=AzireVPN, OU=VPN, CN=ovpn, name=ovpn, [email protected] Nov 8 15:45:15 openvpn[27202]: Data Channel Encrypt: Cipher 'BF-CBC' initialized with 128 bit key Nov 8 15:45:15 openvpn[27202]: Data Channel Encrypt: Using 160 bit message hash 'SHA1' for HMAC authentication Nov 8 15:45:15 openvpn[27202]: Data Channel Decrypt: Cipher 'BF-CBC' initialized with 128 bit key Nov 8 15:45:15 openvpn[27202]: Data Channel Decrypt: Using 160 bit message hash 'SHA1' for HMAC authentication Nov 8 15:45:15 openvpn[27202]: Control Channel: TLSv1, cipher TLSv1/SSLv3 DHE-RSA-AES256-SHA, 2048 bit RSA Nov 8 15:45:15 openvpn[27202]: [ovpn] Peer Connection Initiated with [AF_INET]178.132.75.14:1194 Nov 8 15:45:17 openvpn[27202]: SENT CONTROL [ovpn]: 'PUSH_REQUEST' (status=1) Nov 8 15:45:17 openvpn[27202]: PUSH: Received control message: 'PUSH_REPLY,ifconfig-ipv6 2a03:8600:1001:4010::101f/64 2a03:8600:1001:4010::1,route-ipv6 2000::/3 2A03:8600:1001:4010::1,redirect-gateway def1 bypass-dhcp,dhcp-option DNS 194.1.247.30,tun-ipv6,route-gateway 178.132.77.1,topology subnet,ping 3,ping-restart 15,ifconfig 178.132.77.33 255.255.255.192' Nov 8 15:45:17 openvpn[27202]: OPTIONS IMPORT: timers and/or timeouts modified Nov 8 15:45:17 openvpn[27202]: OPTIONS IMPORT: --ifconfig/up options modified Nov 8 15:45:17 openvpn[27202]: OPTIONS IMPORT: route options modified Nov 8 15:45:17 openvpn[27202]: OPTIONS IMPORT: route-related options modified Nov 8 15:45:17 openvpn[27202]: OPTIONS IMPORT: --ip-win32 and/or --dhcp-option options modified Nov 8 15:45:17 openvpn[27202]: TUN/TAP device tun0 opened Nov 8 15:45:17 openvpn[27202]: TUN/TAP TX queue length set to 100 Nov 8 15:45:17 openvpn[27202]: do_ifconfig, tt->ipv6=1, tt->did_ifconfig_ipv6_setup=1 Nov 8 15:45:17 openvpn[27202]: /usr/sbin/ip link set dev tun0 up mtu 1500 Nov 8 15:45:18 openvpn[27202]: /usr/sbin/ip addr add dev tun0 178.132.77.33/26 broadcast 178.132.77.63 Nov 8 15:45:18 openvpn[27202]: /usr/sbin/ip -6 addr add 2a03:8600:1001:4010::101f/64 dev tun0 Nov 8 15:45:18 openvpn[27202]: Linux ip -6 addr add failed: external program exited with error status: 254 Nov 8 15:45:18 openvpn[27202]: Exiting due to fatal error Any ideas are most welcome!

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