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  • Handling large numbers of sockets with .NET

    - by Dreaddan
    I'm looking at writing a application that need to be able to handle in the region of 200 connections / sec and was wondering if C# and .NET will handle this or if I need to really be looking at C++ to do this? It looks like SocketAsyncEventArgs may be the way to go but thought id check before I plough in to it. Each transaction should only last less than a second but could take up to 15 seconds each if that makes any difference.

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  • Does anyone really understand how HFSC scheduling in Linux/BSD works?

    - by Mecki
    I read the original SIGCOMM '97 PostScript paper about HFSC, it is very technically, but I understand the basic concept. Instead of giving a linear service curve (as with pretty much every other scheduling algorithm), you can specify a convex or concave service curve and thus it is possible to decouple bandwidth and delay. However, even though this paper mentions to kind of scheduling algorithms being used (real-time and link-share), it always only mentions ONE curve per scheduling class (the decoupling is done by specifying this curve, only one curve is needed for that). Now HFSC has been implemented for BSD (OpenBSD, FreeBSD, etc.) using the ALTQ scheduling framework and it has been implemented Linux using the TC scheduling framework (part of iproute2). Both implementations added two additional service curves, that were NOT in the original paper! A real-time service curve and an upper-limit service curve. Again, please note that the original paper mentions two scheduling algorithms (real-time and link-share), but in that paper both work with one single service curve. There never have been two independent service curves for either one as you currently find in BSD and Linux. Even worse, some version of ALTQ seems to add an additional queue priority to HSFC (there is no such thing as priority in the original paper either). I found several BSD HowTo's mentioning this priority setting (even though the man page of the latest ALTQ release knows no such parameter for HSFC, so officially it does not even exist). This all makes the HFSC scheduling even more complex than the algorithm described in the original paper and there are tons of tutorials on the Internet that often contradict each other, one claiming the opposite of the other one. This is probably the main reason why nobody really seems to understand how HFSC scheduling really works. Before I can ask my questions, we need a sample setup of some kind. I'll use a very simple one as seen in the image below: Here are some questions I cannot answer because the tutorials contradict each other: What for do I need a real-time curve at all? Assuming A1, A2, B1, B2 are all 128 kbit/s link-share (no real-time curve for either one), then each of those will get 128 kbit/s if the root has 512 kbit/s to distribute (and A and B are both 256 kbit/s of course), right? Why would I additionally give A1 and B1 a real-time curve with 128 kbit/s? What would this be good for? To give those two a higher priority? According to original paper I can give them a higher priority by using a curve, that's what HFSC is all about after all. By giving both classes a curve of [256kbit/s 20ms 128kbit/s] both have twice the priority than A2 and B2 automatically (still only getting 128 kbit/s on average) Does the real-time bandwidth count towards the link-share bandwidth? E.g. if A1 and B1 both only have 64kbit/s real-time and 64kbit/s link-share bandwidth, does that mean once they are served 64kbit/s via real-time, their link-share requirement is satisfied as well (they might get excess bandwidth, but lets ignore that for a second) or does that mean they get another 64 kbit/s via link-share? So does each class has a bandwidth "requirement" of real-time plus link-share? Or does a class only have a higher requirement than the real-time curve if the link-share curve is higher than the real-time curve (current link-share requirement equals specified link-share requirement minus real-time bandwidth already provided to this class)? Is upper limit curve applied to real-time as well, only to link-share, or maybe to both? Some tutorials say one way, some say the other way. Some even claim upper-limit is the maximum for real-time bandwidth + link-share bandwidth? What is the truth? Assuming A2 and B2 are both 128 kbit/s, does it make any difference if A1 and B1 are 128 kbit/s link-share only, or 64 kbit/s real-time and 128 kbit/s link-share, and if so, what difference? If I use the seperate real-time curve to increase priorities of classes, why would I need "curves" at all? Why is not real-time a flat value and link-share also a flat value? Why are both curves? The need for curves is clear in the original paper, because there is only one attribute of that kind per class. But now, having three attributes (real-time, link-share, and upper-limit) what for do I still need curves on each one? Why would I want the curves shape (not average bandwidth, but their slopes) to be different for real-time and link-share traffic? According to the little documentation available, real-time curve values are totally ignored for inner classes (class A and B), they are only applied to leaf classes (A1, A2, B1, B2). If that is true, why does the ALTQ HFSC sample configuration (search for 3.3 Sample configuration) set real-time curves on inner classes and claims that those set the guaranteed rate of those inner classes? Isn't that completely pointless? (note: pshare sets the link-share curve in ALTQ and grate the real-time curve; you can see this in the paragraph above the sample configuration). Some tutorials say the sum of all real-time curves may not be higher than 80% of the line speed, others say it must not be higher than 70% of the line speed. Which one is right or are they maybe both wrong? One tutorial said you shall forget all the theory. No matter how things really work (schedulers and bandwidth distribution), imagine the three curves according to the following "simplified mind model": real-time is the guaranteed bandwidth that this class will always get. link-share is the bandwidth that this class wants to become fully satisfied, but satisfaction cannot be guaranteed. In case there is excess bandwidth, the class might even get offered more bandwidth than necessary to become satisfied, but it may never use more than upper-limit says. For all this to work, the sum of all real-time bandwidths may not be above xx% of the line speed (see question above, the percentage varies). Question: Is this more or less accurate or a total misunderstanding of HSFC? And if assumption above is really accurate, where is prioritization in that model? E.g. every class might have a real-time bandwidth (guaranteed), a link-share bandwidth (not guaranteed) and an maybe an upper-limit, but still some classes have higher priority needs than other classes. In that case I must still prioritize somehow, even among real-time traffic of those classes. Would I prioritize by the slope of the curves? And if so, which curve? The real-time curve? The link-share curve? The upper-limit curve? All of them? Would I give all of them the same slope or each a different one and how to find out the right slope? I still haven't lost hope that there exists at least a hand full of people in this world that really understood HFSC and are able to answer all these questions accurately. And doing so without contradicting each other in the answers would be really nice ;-)

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  • Does anyone really understand how HFSC scheduling in Linux/BSD works?

    - by Mecki
    I read the original SIGCOMM '97 PostScript paper about HFSC, it is very technically, but I understand the basic concept. Instead of giving a linear service curve (as with pretty much every other scheduling algorithm), you can specify a convex or concave service curve and thus it is possible to decouple bandwidth and delay. However, even though this paper mentions to kind of scheduling algorithms being used (real-time and link-share), it always only mentions ONE curve per scheduling class (the decoupling is done by specifying this curve, only one curve is needed for that). Now HFSC has been implemented for BSD (OpenBSD, FreeBSD, etc.) using the ALTQ scheduling framework and it has been implemented Linux using the TC scheduling framework (part of iproute2). Both implementations added two additional service curves, that were NOT in the original paper! A real-time service curve and an upper-limit service curve. Again, please note that the original paper mentions two scheduling algorithms (real-time and link-share), but in that paper both work with one single service curve. There never have been two independent service curves for either one as you currently find in BSD and Linux. Even worse, some version of ALTQ seems to add an additional queue priority to HSFC (there is no such thing as priority in the original paper either). I found several BSD HowTo's mentioning this priority setting (even though the man page of the latest ALTQ release knows no such parameter for HSFC, so officially it does not even exist). This all makes the HFSC scheduling even more complex than the algorithm described in the original paper and there are tons of tutorials on the Internet that often contradict each other, one claiming the opposite of the other one. This is probably the main reason why nobody really seems to understand how HFSC scheduling really works. Before I can ask my questions, we need a sample setup of some kind. I'll use a very simple one as seen in the image below: Here are some questions I cannot answer because the tutorials contradict each other: What for do I need a real-time curve at all? Assuming A1, A2, B1, B2 are all 128 kbit/s link-share (no real-time curve for either one), then each of those will get 128 kbit/s if the root has 512 kbit/s to distribute (and A and B are both 256 kbit/s of course), right? Why would I additionally give A1 and B1 a real-time curve with 128 kbit/s? What would this be good for? To give those two a higher priority? According to original paper I can give them a higher priority by using a curve, that's what HFSC is all about after all. By giving both classes a curve of [256kbit/s 20ms 128kbit/s] both have twice the priority than A2 and B2 automatically (still only getting 128 kbit/s on average) Does the real-time bandwidth count towards the link-share bandwidth? E.g. if A1 and B1 both only have 64kbit/s real-time and 64kbit/s link-share bandwidth, does that mean once they are served 64kbit/s via real-time, their link-share requirement is satisfied as well (they might get excess bandwidth, but lets ignore that for a second) or does that mean they get another 64 kbit/s via link-share? So does each class has a bandwidth "requirement" of real-time plus link-share? Or does a class only have a higher requirement than the real-time curve if the link-share curve is higher than the real-time curve (current link-share requirement equals specified link-share requirement minus real-time bandwidth already provided to this class)? Is upper limit curve applied to real-time as well, only to link-share, or maybe to both? Some tutorials say one way, some say the other way. Some even claim upper-limit is the maximum for real-time bandwidth + link-share bandwidth? What is the truth? Assuming A2 and B2 are both 128 kbit/s, does it make any difference if A1 and B1 are 128 kbit/s link-share only, or 64 kbit/s real-time and 128 kbit/s link-share, and if so, what difference? If I use the seperate real-time curve to increase priorities of classes, why would I need "curves" at all? Why is not real-time a flat value and link-share also a flat value? Why are both curves? The need for curves is clear in the original paper, because there is only one attribute of that kind per class. But now, having three attributes (real-time, link-share, and upper-limit) what for do I still need curves on each one? Why would I want the curves shape (not average bandwidth, but their slopes) to be different for real-time and link-share traffic? According to the little documentation available, real-time curve values are totally ignored for inner classes (class A and B), they are only applied to leaf classes (A1, A2, B1, B2). If that is true, why does the ALTQ HFSC sample configuration (search for 3.3 Sample configuration) set real-time curves on inner classes and claims that those set the guaranteed rate of those inner classes? Isn't that completely pointless? (note: pshare sets the link-share curve in ALTQ and grate the real-time curve; you can see this in the paragraph above the sample configuration). Some tutorials say the sum of all real-time curves may not be higher than 80% of the line speed, others say it must not be higher than 70% of the line speed. Which one is right or are they maybe both wrong? One tutorial said you shall forget all the theory. No matter how things really work (schedulers and bandwidth distribution), imagine the three curves according to the following "simplified mind model": real-time is the guaranteed bandwidth that this class will always get. link-share is the bandwidth that this class wants to become fully satisfied, but satisfaction cannot be guaranteed. In case there is excess bandwidth, the class might even get offered more bandwidth than necessary to become satisfied, but it may never use more than upper-limit says. For all this to work, the sum of all real-time bandwidths may not be above xx% of the line speed (see question above, the percentage varies). Question: Is this more or less accurate or a total misunderstanding of HSFC? And if assumption above is really accurate, where is prioritization in that model? E.g. every class might have a real-time bandwidth (guaranteed), a link-share bandwidth (not guaranteed) and an maybe an upper-limit, but still some classes have higher priority needs than other classes. In that case I must still prioritize somehow, even among real-time traffic of those classes. Would I prioritize by the slope of the curves? And if so, which curve? The real-time curve? The link-share curve? The upper-limit curve? All of them? Would I give all of them the same slope or each a different one and how to find out the right slope? I still haven't lost hope that there exists at least a hand full of people in this world that really understood HFSC and are able to answer all these questions accurately. And doing so without contradicting each other in the answers would be really nice ;-)

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  • Why does async BeginReceiveFrom never time out on a raw socket?

    - by James Hugard
    Writing an asynchronous Ping using Raw Sockets in F#, to enable parallel requests using as few threads as possible. Not using "System.Net.NetworkInformation.Ping", because it appears to allocate one thread per request. Am also interested in using F# async workflows. The synchronous version below correctly times out when the target host does not exist/respond, but the asynchronous version hangs. Both work when the host does respond. Not sure if this is a .NET issue, or an F# one... Any ideas? (note: the process must run as Admin to allow Raw Socket access) This throws a timeout: let result = Ping.Ping ( IPAddress.Parse( "192.168.33.22" ), 1000 ) However, this hangs: let result = Ping.AsyncPing ( IPAddress.Parse( "192.168.33.22" ), 1000 ) |> Async.RunSynchronously Here's the code... module Ping open System open System.Net open System.Net.Sockets open System.Threading //---- ICMP Packet Classes type IcmpMessage (t : byte) = let mutable m_type = t let mutable m_code = 0uy let mutable m_checksum = 0us member this.Type with get() = m_type member this.Code with get() = m_code member this.Checksum = m_checksum abstract Bytes : byte array default this.Bytes with get() = [| m_type m_code byte(m_checksum) byte(m_checksum >>> 8) |] member this.GetChecksum() = let mutable sum = 0ul let bytes = this.Bytes let mutable i = 0 // Sum up uint16s while i < bytes.Length - 1 do sum <- sum + uint32(BitConverter.ToUInt16( bytes, i )) i <- i + 2 // Add in last byte, if an odd size buffer if i <> bytes.Length then sum <- sum + uint32(bytes.[i]) // Shuffle the bits sum <- (sum >>> 16) + (sum &&& 0xFFFFul) sum <- sum + (sum >>> 16) sum <- ~~~sum uint16(sum) member this.UpdateChecksum() = m_checksum <- this.GetChecksum() type InformationMessage (t : byte) = inherit IcmpMessage(t) let mutable m_identifier = 0us let mutable m_sequenceNumber = 0us member this.Identifier = m_identifier member this.SequenceNumber = m_sequenceNumber override this.Bytes with get() = Array.append (base.Bytes) [| byte(m_identifier) byte(m_identifier >>> 8) byte(m_sequenceNumber) byte(m_sequenceNumber >>> 8) |] type EchoMessage() = inherit InformationMessage( 8uy ) let mutable m_data = Array.create 32 32uy do base.UpdateChecksum() member this.Data with get() = m_data and set(d) = m_data <- d this.UpdateChecksum() override this.Bytes with get() = Array.append (base.Bytes) (this.Data) //---- Synchronous Ping let Ping (host : IPAddress, timeout : int ) = let mutable ep = new IPEndPoint( host, 0 ) let socket = new Socket( AddressFamily.InterNetwork, SocketType.Raw, ProtocolType.Icmp ) socket.SetSocketOption( SocketOptionLevel.Socket, SocketOptionName.SendTimeout, timeout ) socket.SetSocketOption( SocketOptionLevel.Socket, SocketOptionName.ReceiveTimeout, timeout ) let packet = EchoMessage() let mutable buffer = packet.Bytes try if socket.SendTo( buffer, ep ) <= 0 then raise (SocketException()) buffer <- Array.create (buffer.Length + 20) 0uy let mutable epr = ep :> EndPoint if socket.ReceiveFrom( buffer, &epr ) <= 0 then raise (SocketException()) finally socket.Close() buffer //---- Entensions to the F# Async class to allow up to 5 paramters (not just 3) type Async with static member FromBeginEnd(arg1,arg2,arg3,arg4,beginAction,endAction,?cancelAction): Async<'T> = Async.FromBeginEnd((fun (iar,state) -> beginAction(arg1,arg2,arg3,arg4,iar,state)), endAction, ?cancelAction=cancelAction) static member FromBeginEnd(arg1,arg2,arg3,arg4,arg5,beginAction,endAction,?cancelAction): Async<'T> = Async.FromBeginEnd((fun (iar,state) -> beginAction(arg1,arg2,arg3,arg4,arg5,iar,state)), endAction, ?cancelAction=cancelAction) //---- Extensions to the Socket class to provide async SendTo and ReceiveFrom type System.Net.Sockets.Socket with member this.AsyncSendTo( buffer, offset, size, socketFlags, remoteEP ) = Async.FromBeginEnd( buffer, offset, size, socketFlags, remoteEP, this.BeginSendTo, this.EndSendTo ) member this.AsyncReceiveFrom( buffer, offset, size, socketFlags, remoteEP ) = Async.FromBeginEnd( buffer, offset, size, socketFlags, remoteEP, this.BeginReceiveFrom, (fun asyncResult -> this.EndReceiveFrom(asyncResult, remoteEP) ) ) //---- Asynchronous Ping let AsyncPing (host : IPAddress, timeout : int ) = async { let ep = IPEndPoint( host, 0 ) use socket = new Socket( AddressFamily.InterNetwork, SocketType.Raw, ProtocolType.Icmp ) socket.SetSocketOption( SocketOptionLevel.Socket, SocketOptionName.SendTimeout, timeout ) socket.SetSocketOption( SocketOptionLevel.Socket, SocketOptionName.ReceiveTimeout, timeout ) let packet = EchoMessage() let outbuffer = packet.Bytes try let! result = socket.AsyncSendTo( outbuffer, 0, outbuffer.Length, SocketFlags.None, ep ) if result <= 0 then raise (SocketException()) let epr = ref (ep :> EndPoint) let inbuffer = Array.create (outbuffer.Length + 256) 0uy let! result = socket.AsyncReceiveFrom( inbuffer, 0, inbuffer.Length, SocketFlags.None, epr ) if result <= 0 then raise (SocketException()) return inbuffer finally socket.Close() }

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  • How to detect a timeout when using asynchronous Socket.BeginReceive?

    - by James Hugard
    Writing an asynchronous Ping using Raw Sockets in F#, to enable parallel requests using as few threads as possible. Not using "System.Net.NetworkInformation.Ping", because it appears to allocate one thread per request. Am also interested in using F# async workflows. The synchronous version below correctly times out when the target host does not exist/respond, but the asynchronous version hangs. Both work when the host does respond. Not sure if this is a .NET issue, or an F# one... Any ideas? (note: the process must run as Admin to allow Raw Socket access) This throws a timeout: let result = Ping.Ping ( IPAddress.Parse( "192.168.33.22" ), 1000 ) However, this hangs: let result = Ping.AsyncPing ( IPAddress.Parse( "192.168.33.22" ), 1000 ) |> Async.RunSynchronously Here's the code... module Ping open System open System.Net open System.Net.Sockets open System.Threading //---- ICMP Packet Classes type IcmpMessage (t : byte) = let mutable m_type = t let mutable m_code = 0uy let mutable m_checksum = 0us member this.Type with get() = m_type member this.Code with get() = m_code member this.Checksum = m_checksum abstract Bytes : byte array default this.Bytes with get() = [| m_type m_code byte(m_checksum) byte(m_checksum >>> 8) |] member this.GetChecksum() = let mutable sum = 0ul let bytes = this.Bytes let mutable i = 0 // Sum up uint16s while i < bytes.Length - 1 do sum <- sum + uint32(BitConverter.ToUInt16( bytes, i )) i <- i + 2 // Add in last byte, if an odd size buffer if i <> bytes.Length then sum <- sum + uint32(bytes.[i]) // Shuffle the bits sum <- (sum >>> 16) + (sum &&& 0xFFFFul) sum <- sum + (sum >>> 16) sum <- ~~~sum uint16(sum) member this.UpdateChecksum() = m_checksum <- this.GetChecksum() type InformationMessage (t : byte) = inherit IcmpMessage(t) let mutable m_identifier = 0us let mutable m_sequenceNumber = 0us member this.Identifier = m_identifier member this.SequenceNumber = m_sequenceNumber override this.Bytes with get() = Array.append (base.Bytes) [| byte(m_identifier) byte(m_identifier >>> 8) byte(m_sequenceNumber) byte(m_sequenceNumber >>> 8) |] type EchoMessage() = inherit InformationMessage( 8uy ) let mutable m_data = Array.create 32 32uy do base.UpdateChecksum() member this.Data with get() = m_data and set(d) = m_data <- d this.UpdateChecksum() override this.Bytes with get() = Array.append (base.Bytes) (this.Data) //---- Synchronous Ping let Ping (host : IPAddress, timeout : int ) = let mutable ep = new IPEndPoint( host, 0 ) let socket = new Socket( AddressFamily.InterNetwork, SocketType.Raw, ProtocolType.Icmp ) socket.SetSocketOption( SocketOptionLevel.Socket, SocketOptionName.SendTimeout, timeout ) socket.SetSocketOption( SocketOptionLevel.Socket, SocketOptionName.ReceiveTimeout, timeout ) let packet = EchoMessage() let mutable buffer = packet.Bytes try if socket.SendTo( buffer, ep ) <= 0 then raise (SocketException()) buffer <- Array.create (buffer.Length + 20) 0uy let mutable epr = ep :> EndPoint if socket.ReceiveFrom( buffer, &epr ) <= 0 then raise (SocketException()) finally socket.Close() buffer //---- Entensions to the F# Async class to allow up to 5 paramters (not just 3) type Async with static member FromBeginEnd(arg1,arg2,arg3,arg4,beginAction,endAction,?cancelAction): Async<'T> = Async.FromBeginEnd((fun (iar,state) -> beginAction(arg1,arg2,arg3,arg4,iar,state)), endAction, ?cancelAction=cancelAction) static member FromBeginEnd(arg1,arg2,arg3,arg4,arg5,beginAction,endAction,?cancelAction): Async<'T> = Async.FromBeginEnd((fun (iar,state) -> beginAction(arg1,arg2,arg3,arg4,arg5,iar,state)), endAction, ?cancelAction=cancelAction) //---- Extensions to the Socket class to provide async SendTo and ReceiveFrom type System.Net.Sockets.Socket with member this.AsyncSendTo( buffer, offset, size, socketFlags, remoteEP ) = Async.FromBeginEnd( buffer, offset, size, socketFlags, remoteEP, this.BeginSendTo, this.EndSendTo ) member this.AsyncReceiveFrom( buffer, offset, size, socketFlags, remoteEP ) = Async.FromBeginEnd( buffer, offset, size, socketFlags, remoteEP, this.BeginReceiveFrom, (fun asyncResult -> this.EndReceiveFrom(asyncResult, remoteEP) ) ) //---- Asynchronous Ping let AsyncPing (host : IPAddress, timeout : int ) = async { let ep = IPEndPoint( host, 0 ) use socket = new Socket( AddressFamily.InterNetwork, SocketType.Raw, ProtocolType.Icmp ) socket.SetSocketOption( SocketOptionLevel.Socket, SocketOptionName.SendTimeout, timeout ) socket.SetSocketOption( SocketOptionLevel.Socket, SocketOptionName.ReceiveTimeout, timeout ) let packet = EchoMessage() let outbuffer = packet.Bytes try let! result = socket.AsyncSendTo( outbuffer, 0, outbuffer.Length, SocketFlags.None, ep ) if result <= 0 then raise (SocketException()) let epr = ref (ep :> EndPoint) let inbuffer = Array.create (outbuffer.Length + 256) 0uy let! result = socket.AsyncReceiveFrom( inbuffer, 0, inbuffer.Length, SocketFlags.None, epr ) if result <= 0 then raise (SocketException()) return inbuffer finally socket.Close() }

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  • Why does BeginReceiveFrom never time out?

    - by James Hugard
    I am writing an asynchronous Ping using Raw Sockets in F#, to enable parallel requests using as few threads as possible ("System.Net.NetworkInformation.Ping" appears to use one thread per request, but have not tested this... also am interested in using F# async workflows). The synchronous version below correctly times out when the target host does not exist/respond, but the asynchronous version hangs. Both work when the host does respond... Any ideas? (note: the process must run as Admin for this code to work) This throws a timeout: let result = Ping.Ping ( IPAddress.Parse( "192.168.33.22" ), 1000 ) However, this hangs: let result = Ping.PingAsync ( IPAddress.Parse( "192.168.33.22" ), 1000 ) |> Async.RunSynchronously Here's the code... module Ping open System open System.Net open System.Net.Sockets open System.Threading //---- ICMP Packet Classes type IcmpMessage (t : byte) = let mutable m_type = t let mutable m_code = 0uy let mutable m_checksum = 0us member this.Type with get() = m_type member this.Code with get() = m_code member this.Checksum = m_checksum abstract Bytes : byte array default this.Bytes with get() = [| m_type m_code byte(m_checksum) byte(m_checksum >>> 8) |] member this.GetChecksum() = let mutable sum = 0ul let bytes = this.Bytes let mutable i = 0 // Sum up uint16s while i < bytes.Length - 1 do sum <- sum + uint32(BitConverter.ToUInt16( bytes, i )) i <- i + 2 // Add in last byte, if an odd size buffer if i <> bytes.Length then sum <- sum + uint32(bytes.[i]) // Shuffle the bits sum <- (sum >>> 16) + (sum &&& 0xFFFFul) sum <- sum + (sum >>> 16) sum <- ~~~sum uint16(sum) member this.UpdateChecksum() = m_checksum <- this.GetChecksum() type InformationMessage (t : byte) = inherit IcmpMessage(t) let mutable m_identifier = 0us let mutable m_sequenceNumber = 0us member this.Identifier = m_identifier member this.SequenceNumber = m_sequenceNumber override this.Bytes with get() = Array.append (base.Bytes) [| byte(m_identifier) byte(m_identifier >>> 8) byte(m_sequenceNumber) byte(m_sequenceNumber >>> 8) |] type EchoMessage() = inherit InformationMessage( 8uy ) let mutable m_data = Array.create 32 32uy do base.UpdateChecksum() member this.Data with get() = m_data and set(d) = m_data <- d this.UpdateChecksum() override this.Bytes with get() = Array.append (base.Bytes) (this.Data) //---- Synchronous Ping let Ping (host : IPAddress, timeout : int ) = let mutable ep = new IPEndPoint( host, 0 ) let socket = new Socket( AddressFamily.InterNetwork, SocketType.Raw, ProtocolType.Icmp ) socket.SetSocketOption( SocketOptionLevel.Socket, SocketOptionName.SendTimeout, timeout ) socket.SetSocketOption( SocketOptionLevel.Socket, SocketOptionName.ReceiveTimeout, timeout ) let packet = EchoMessage() let mutable buffer = packet.Bytes try if socket.SendTo( buffer, ep ) <= 0 then raise (SocketException()) buffer <- Array.create (buffer.Length + 20) 0uy let mutable epr = ep :> EndPoint if socket.ReceiveFrom( buffer, &epr ) <= 0 then raise (SocketException()) finally socket.Close() buffer //---- Entensions to the F# Async class to allow up to 5 paramters (not just 3) type Async with static member FromBeginEnd(arg1,arg2,arg3,arg4,beginAction,endAction,?cancelAction): Async<'T> = Async.FromBeginEnd((fun (iar,state) -> beginAction(arg1,arg2,arg3,arg4,iar,state)), endAction, ?cancelAction=cancelAction) static member FromBeginEnd(arg1,arg2,arg3,arg4,arg5,beginAction,endAction,?cancelAction): Async<'T> = Async.FromBeginEnd((fun (iar,state) -> beginAction(arg1,arg2,arg3,arg4,arg5,iar,state)), endAction, ?cancelAction=cancelAction) //---- Extensions to the Socket class to provide async SendTo and ReceiveFrom type System.Net.Sockets.Socket with member this.AsyncSendTo( buffer, offset, size, socketFlags, remoteEP ) = Async.FromBeginEnd( buffer, offset, size, socketFlags, remoteEP, this.BeginSendTo, this.EndSendTo ) member this.AsyncReceiveFrom( buffer, offset, size, socketFlags, remoteEP ) = Async.FromBeginEnd( buffer, offset, size, socketFlags, remoteEP, this.BeginReceiveFrom, (fun asyncResult -> this.EndReceiveFrom(asyncResult, remoteEP) ) ) //---- Asynchronous Ping let PingAsync (host : IPAddress, timeout : int ) = async { let ep = IPEndPoint( host, 0 ) use socket = new Socket( AddressFamily.InterNetwork, SocketType.Raw, ProtocolType.Icmp ) socket.SetSocketOption( SocketOptionLevel.Socket, SocketOptionName.SendTimeout, timeout ) socket.SetSocketOption( SocketOptionLevel.Socket, SocketOptionName.ReceiveTimeout, timeout ) let packet = EchoMessage() let outbuffer = packet.Bytes try let! result = socket.AsyncSendTo( outbuffer, 0, outbuffer.Length, SocketFlags.None, ep ) if result <= 0 then raise (SocketException()) let epr = ref (ep :> EndPoint) let inbuffer = Array.create (outbuffer.Length + 256) 0uy let! result = socket.AsyncReceiveFrom( inbuffer, 0, inbuffer.Length, SocketFlags.None, epr ) if result <= 0 then raise (SocketException()) return inbuffer finally socket.Close() }

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  • Incorporating GPL Code in my Open Source Project

    - by rutherford
    I have downloaded a currently inactive GPL project with a view to updating it and releasing the completed codebase as open source. I'm not really a fan of GPL though and would rather licence my project under BSD. What are my options? Is it just a case of keeping any existing non-touched code under the GPL and any updated stuff can be BSD (messy)? The source will essentially be the same codebase i.e. there is no logical separation between the two and they certainly can't be split into anything resembling different libraries. Are my only realistic options to either GPL the whole thing or seek the original author's permission to release everthing under BSD?

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  • Lightweight Linux distro that includes developer tools? (or, the most BSD-like Linux)

    - by RevAaron
    I cut my teeth on Minix and Slackware 1.1, but I've been in the OS X Wilderness for the last few years. I'm trying to standardize on a Linux distribution for personal and work-related use on less powerful laptops and under virtualization. So far, NetBSD and OpenBSD are the best fit for my purposes- but after plenty of frustration I've come to the conclusion that I need to stick with Linux to get the hardware and software support that comes with it. What I like about NetBSD/OpenBSD that I'd like to keep: X, but no default KDE, GNOME or XFCE! A sensible /etc and dot file setup- startx calls xinit, xinit looks for ~/.xinitrc; nothing more complicated than that is needed. Command line tools and file-based configuration: I shouldn't need a GUI to connect to a WAP. Decent selection of binary packages; building from source is OK, but nothing source-only like Gentoo. pkg_add (BSD) and apt-get both have treated me well in the past. Modest RAM and HDD requirements: boot + X + awesome+ two xterms takes up 80 MB on OpenBSD and 240 MB on Debian 5 and Crunchbang In my experience, most "lightweight" and Live CDs focus on a nice desktop environment crammed into a CD or USB stick; once you add build-essentials you end up with something just about as bloated as Ubuntu or Debian full install. Crunchbang is a great example. Thanks in advance for all suggestions!

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  • What Parallel computing APIs take good use of sockets?

    - by Ole Jak
    What Parallel computing APIs take good use of sockets? So my programm uses soskets, what Parallel computing APIs I can use that would help me but will not obligate me to go from sockets to anything else... I mean when we are on claster with some special, not socket infrastructure sistem that API emulates something like socket but uses that infrustructure (so programm peforms much faster then on sockets, but keeps having nice soskets API)

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  • what happens when I don't manage to call `recv` fast enough?

    - by amn
    Hi all, I want to account for a possible scenario where clients of my TCP/IP stream socket service send data to my service faster than it manages to move the data to its buffers (I am talking about application buffers, naturally) with recv and work with it. So basically, what happens in such scenarios? Obviously, some sort of service beneath my service which is a user application, has to receive incoming stream and store it somewhere until I issue 'recv', right? Most certainly the operating system. I don't want to re-open old questions, but I can't seem to find an answer to this seemingly obvious one?

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  • Is there a BSD equivalent to "!!"?

    - by CT
    I often find myself issuing a command that I do not have the proper elevated privileges for. On Ubuntu I could use sudo !! This would issue the same command with sudo privlidges. Is there an equivalent on OpenBSD? Edit: I should have been more specific on what version of OpenBSD. I am using OpenBSD 4.8 where sudo seems to be installed by default. I have already created a user besides root and edited my sudoers file to allow for that user to use sudo. My question is, is there already a built-in shortcut for the "!!" to use previous command.

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  • Android Design - Service vs Thread for Networking

    - by Nevyn
    I am writing an Android app, finally (yay me) and for this app I need persistant, but user closeable, network sockets (yes, more than one). I decided to try my hand at writing my own version of an IRC Client. My design issue however, is I'm not sure how to run the Socket connectivity itself. If I put the sockets at the Activity level, they keeps getting closed shortly after the Activity becomes non-visible (also a problem that needs solving...but I think i figured that one out)...but if I run a "connectivity service", I need to find out if I can have multiple instances of it running (the service, that is...one per server/socket). Either that or a I need a way to Thread the sockets themselves and have multiple threads running that I can still communicate with directly (ID system of some sort). Thus the question: Is it a 'better', or at least more "proper" design pattern, to put the Socket and networking in a service, and have the Activities consume said service...or should I tie the sockets directly to some Threaded Process owned by the UI Activity and not bother with the service implementation at all? I do know better than to put the networking directly on the UI thread, but that's as far as I've managed to get.

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  • TCP/IP & throughput between FreeNAS (BSD) server & other LAN machines

    - by Tim Dickerson
    I have got a question for someone that knows BSD a bit better than me that are in regards to my LAN setup at home/work here outside Chicago. I can't seem to fully optimize my network's (LAN) thoughput via my FreeNAS (BSD based) file server. It runs with the latest FreeBSD release which is modified to support several protocols for file transfers and more. Every machine that is behind my Smoothwall (Linux based) router is on the usual 192.168.0.x subnet and for most part works just fine. Behind the Smoothwall box, all machines are connected to a GB HP unmanaged switch. I host a large WISP here and have an OC-3 connection here at home/work and have no issues with downloading/uploading from/to the 'net'. My problem is with throughput. When I try and transfer large files...really any for that matter..between any of the machines to/and from the FreeNAS server via FTP, the max throughput I can achieve say between a Win 7 or a Linux box is ~65Mbit/sec. All machines are running Intel Pro 1000 GB NIC's and all cable is CAT6. Each is set to 'auto negotiation' and each shows 1500 MTU Full Duplex @1GB so I know the hardware is okay. I have not adjusted the MTU on any machine as I understand it to be pointless unless certain configurations are used (I assume I am not one of those). My settings for the FreeNAS machine are the following: # FreeNAS /etc/sysctl.conf - pertinent settings shown kern.ipc.maxsockbuf=262144 kern.ipc.nmbclusters=32768 kern.ipc.somaxconn=8192 kern.maxfiles=65536 kern.maxfilesperproc=32768 net.inet.tcp.delayed_ack=0 net.inet.tcp.inflight.enable=0 net.inet.tcp.path_mtu_discovery=0 net.inet.tcp.recvbuf_auto=1 net.inet.tcp.recvbuf_inc=524288 net.inet.tcp.recvbuf_max=16777216 net.inet.tcp.recvspace=65536 net.inet.tcp.rfc1323=1 net.inet.tcp.sendbuf_inc=16384 net.inet.tcp.sendbuf_max=16777216 net.inet.tcp.sendspace=65536 net.inet.udp.recvspace=65536 net.local.stream.recvspace=65536 net.local.stream.sendspace=65536 net.inet.tcp.hostcache.expire=1 From what I can tell, that looks to be a somewhat optimized profile for a typical BSD machine acting as a server for a LAN. I might be wrong and just wanted to find out from someone that knows BSD better than I do if indeed that is ok or if something is out of tune or what. Are there other ways I would find better for P2P file transfers? I honestly do not know what I SHOULD be looking for with respect to throughput between the NAS box and another client when xferring files via FTP, but I am told that what I get on average (40-70MB/sec) is too low for what it could be. I have thought about adding another NIC in the FreeNAS box as well as the Win7 machine and use a X-over cable via a static route, but wanted to check with someone first to see if that might be worth it or not. I don't know if doing that would bypass the HP GB switch and allow for a machine to machine xfer anyways. The FTP client I use is: Filezilla and have tried both active and passive modes with no real gain over each other. The NAS box runs ProFTPD.

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  • Open source license with backlink requirement

    - by KajMagnus
    I'm developing a Javascript library, and I'm thinking about releasing it under an open source license (e.g. GPL, BSD, MIT) — but that requires that websites that use the software link back to my website. Do you know about any such licenses? And how have they formulated the attribution part of the license text? Do you think this BSD-license would do what you think that I want? (I suppose it doesn't :-)) [...] 3. Each website that redistributes this work must include a visible rel=follow link to my-website.example.com, reachable via rel=follow links from each page where the software is being redistributed. (For example, you could have a link back to your homepage, and from your homepage to an About-Us section, which could link to a Credits section) I realize that some companies wouldn't want to use the library because of legal issues with interpreting non-standard licenses (have a look at this answer: http://programmers.stackexchange.com/a/156859/54906). — After half a year, or perhaps some years, I'd change the license to plain GPL + MIT.

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

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

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