Search Results

Search found 1705 results on 69 pages for 'syn packet'.

Page 22/69 | < Previous Page | 18 19 20 21 22 23 24 25 26 27 28 29  | Next Page >

• Application stuck in TCP retransmit

  - by SandeepJ

  I am running Linux kernel 3.13 (Ubuntu 14.04) on two Virtual Machines each of which operates inside two different servers running ESXi 5.1. There is a zeromq client-server application running between the two VMs. After running for about 10-30 minutes, this application consistently hangs due to inability to retransmit a lost packet. When I run the same setup over Ubuntu 12.04 (Linux 3.11), the application never fails If you notice below, "ss" (socket statistics) shows 1 packet lost, sk_wmem_queued of 14110 (i.e. w14110) and a high rto (120000). State Recv-Q Send-Q Local Address:Port Peer Address:Port ESTAB 0 12350 192.168.2.122:41808 192.168.2.172:55550 timer:(on,16sec,10) uid:1000 ino:35042 sk:ffff880035bcb100 <- skmem:(r0,rb648720,t0,tb1164800,f2274,w14110,o0,bl0) ts sack cubic wscale:7,7 rto:120000 rtt:7.5/3 ato:40 mss:8948 cwnd:1 ssthresh:21 send 9.5Mbps unacked:1 retrans:1/10 lost:1 rcv_rtt:1476 rcv_space:37621 Since this has happened so consistently, I was able to capture the TCP log in wireshark. I found that the packet which is lost does get retransmitted and even acknowledged by the TCP in the other OS (the sequence number is seen in the ACK), but the sender doesn't seem to understand this ACK and continues retransmitting. MTU is 9000 on both virtual machines and througout the route. The packets being sent are large in size. As I said earlier, this does not happen on Ubuntu 12.04 (kernel 3.11). So I did a diff on the TCP config options (seen via "sysctl -a |grep tcp ") between 14.04 and 12.04 and found the following differences. I also noticed that net.ipv4.tcp_mtu_probing=0 in both configurations. Left side is 3.11, right side is 3.13 <<net.ipv4.tcp_abc = 0 <<net.ipv4.tcp_cookie_size = 0 <<net.ipv4.tcp_dma_copybreak = 4096 14c11 << net.ipv4.tcp_early_retrans = 2 --- >> net.ipv4.tcp_early_retrans = 3 17c14 << net.ipv4.tcp_fastopen = 0 >> net.ipv4.tcp_fastopen = 1 20d16 << net.ipv4.tcp_frto_response = 0 26,27c22 << net.ipv4.tcp_max_orphans = 16384 << net.ipv4.tcp_max_ssthresh = 0 >> net.ipv4.tcp_max_orphans = 4096 29,30c24,25 << net.ipv4.tcp_max_tw_buckets = 16384 << net.ipv4.tcp_mem = 94377 125837 188754 >> net.ipv4.tcp_max_tw_buckets = 4096 >> net.ipv4.tcp_mem = 23352 31138 46704 34a30 >> net.ipv4.tcp_notsent_lowat = -1 My question to the networking experts on this forum : Are there any other debugging tools or options I can install/enable to dig further into why this TCP retransmit failure is occurring so consistently ? Are there any configuration changes which might account for this weird behaviour.

  Read the article

• 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() }

  Read the article

• 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() }

  Read the article

• Odd tcp deadlock under windows

  - by John Robertson

  We are moving large amounts of data on a LAN and it has to happen very rapidly and reliably. Currently we use windows TCP as implemented in C++. Using large (synchronous) sends moves the data much faster than a bunch of smaller (synchronous) sends but will frequently deadlock for large gaps of time (.15 seconds) causing the overall transfer rate to plummet. This deadlock happens in very particular circumstances which makes me believe it should be preventable altogether. More importantly if we don't really know the cause we don't really know it won't happen some time with smaller sends anyway. Can anyone explain this deadlock? Deadlock description (OK, zombie-locked, it isn't dead, but for .15 or so seconds it stops, then starts again) The receiving side sends an ACK. The sending side sends a packet containing the end of a message (push flag is set) The call to socket.recv takes about .15 seconds(!) to return About the time the call returns an ACK is sent by the receiving side The the next packet from the sender is finally sent (why is it waiting? the tcp window is plenty big) The odd thing about (3) is that typically that call doesn't take much time at all and receives exactly the same amount of data. On a 2Ghz machine that's 300 million instructions worth of time. I am assuming the call doesn't (heaven forbid) wait for the received data to be acked before it returns, so the ack must be waiting for the call to return, or both must be delayed by something else. The problem NEVER happens when there is a second packet of data (part of the same message) arriving between 1 and 2. That part very clearly makes it sound like it has to do with the fact that windows TCP will not send back a no-data ACK until either a second packet arrives or a 200ms timer expires. However the delay is less than 200 ms (its more like 150 ms). The third unseemly character (and to my mind the real culprit) is (5). Send is definitely being called well before that .15 seconds is up, but the data NEVER hits the wire before that ack returns. That is the most bizarre part of this deadlock to me. Its not a tcp blockage because the TCP window is plenty big since we set SO_RCVBUF to something like 500*1460 (which is still under a meg). The data is coming in very fast (basically there is a loop spinning out data via send) so the buffer should fill almost immediately. According to msdn the buffer being full and at least one pending send should cause the data to be sent (though in another place it mentions that there various "heuristics" used in deciding when a send hits the wire). Anway, why the sender doesn't actually send more data during that .15 second pause is the most bizarre part to me. The information above was captured on the receiving side via wireshark (except of course the socket.recv return times which were logged in a text file). We tried changing the send buffer to zero and turning off Nagle on the sender (yes, I know Nagle is about not sending small packets - but we tried turning Nagle off in case that was part of the unstated "heuristics" affecting whether the message would be posted to the wire. Technically microsoft's Nagle is that a small packet isn't sent if the buffer is full and there is an outstanding ACK, so it seemed like a possibility).

  Read the article

• 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() }

  Read the article

• Incorrect data when passing pointer a list of pointers to a function. (C++)

  - by Phil Elm

  I'm writing code for combining data received over multiple sources. When the objects received (I'll call them MyPacket for now), they are stored in a standard list. However, whenever I reference the payload size of a partial MyPacket, the value shows up as 1 instead of the intended size. Here's the function code: MyPacket* CombinePackets(std::list<MyPacket*>* packets, uint8* current_packet){ uint32 total_payload_size = 0; if(packets->size() <= 0) return NULL; //For now. std::list<MyPacket*>::iterator it = packets->begin(); //Some minor code here, not relevant to the problem. for(uint8 index = 0; index < packets->size(); index++){ //(*it)->GetPayloadSize() returns 1 when it should show 1024. I've tried directly accessing the variable and more, but I just can't get it to work. total_payload_size += (*it)->GetPayloadSize(); cout << "Adding to total payload size value: " << (*it)->GetPayloadSize() << endl; std::advance(it,1); } MyPacket* packet = new MyPacket(); //Byte is just a typedef'd unsigned char. packet->payload = (byte) calloc(total_payload_size, sizeof(byte)); packet->payload_size = total_payload_size; it = packets->begin(); //Go back to the beginning again. uint32 big_payload_index = 0; for(uint8 index = 0; index < packets->size(); index++){ if(current_packet != NULL) *current_packet = index; for(uint32 payload_index = 0; payload_index < (*it)->GetPayloadSize(); payload_index++){ packet->payload[big_payload_index] = (*it)->payload[payload_index]; big_payload_index++; } std::advance(it,1); } return packet; } //Calling code std::list<MyPacket*> received = std::list<MyPacket*>(); //The code that fills it is here. std::list<MyPacket*>::iterator it = received.begin(); cout << (*it)->GetPayloadSize() << endl; // Outputs 1024 correctly! MyPacket* final = CombinePackets(&received,NULL); cout << final->GetPayloadSize() << endl; //Outputs 181, which happens to be the number of elements in the received list. So, as you can see above, when I reference (*it)-GetPayloadSize(), it returns 1 instead of the intended 1024. Can anyone see the problem and if so, do you have an idea on how to fix this? I've spent 4 hours searching and trying new solutions, but they all keep returning 1... EDIT:

  Read the article

• pfSense: How to route traffic out the WAN port?

  - by Ian Boyd

  Expert version i want to create a route in pfSense that will send traffic out the physical WAN port, not the PPPoE WAN port. i want to talk to talk to the web-server on my DSL modem, but it doesn't see packets wrapped in a PPPoE header. Long version My pfSense router is responsible for setting up the PPPoE connection over DSL to my ISP. When a machine on the LAN wants to sent packets to the internet, the default route sends packets out over the PPPoE connection. Those packets, wrapped in a PPPoE header, are sent on the ethernet cable to my DSL modem. From there they are sent the ISP, and the internet at large. i want a way to send a packet out the WAN port itself - not the PPPoE WAN port. My modem is sitting out there, with a http interface where i can monitor connection speed signal-to-noise ratio bandwidth connection time Whenever i try to set a route for destination of 192.168.2.1 (the IP that the modem will listen to for HTTP requests) to go out the WAN port, they instead end up going out the PPPoE port. The difference being that they're wrapped in a PPPoE protocol packet, and the modem isn't being sent the packet, it's being delivered to the ISP. Given that pfSense has no ability to direct traffic out the physical WAN port: how can i direct traffic out the physical WAN port on pfSense?

  Read the article

• Snort/Barnyard2 Logging

  - by Eric

  I need some help with my Snort/Barnyard2 setup. My goal is to have Snort send unified2 logs to Barnyard2 and then have Barnyard2 send the data to other locations. Here is my currrent setup. OS Scientific Linux 6 Snort Version 2.9.2.3 Barnyard2 Version 2.1.9 Snort command snort -c /etc/snort/snort.conf -i eth2 & Barnyard2 command /usr/local/bin/barnyard2 -c /etc/snort/barnyard2.conf -d /var/log/snort -f snort.log -w /var/log/snort/barnyard.waldo & snort.conf output unified2: filename snort.log, limit 128 barnyard2.conf output alert_syslog: host=127.0.0.1 output database: log, mysql, user=snort dbname=snort password=password host=localhost With this setup, barnyard2 is showing all of the correct information in the database and I'm using BASE to view it on the web GUI. I was hoping to be able to send the full packet data to syslog with barnyard2 but after reading around, it seems that it is impossible to do that. So I then started trying to modify the snort.conf file and add lines like "output alert_full: alert.full". This definitely gave me a lot more information but still not the full packet data like I want. So my question is, is there anyway I can use barnyard2 to send the full packet data of alerts to a human readable file? Since I can't send it directly to syslog, I can create another process to take the data from that file and ship it off to another server. If not, what flags and/or snort.conf configuration would you recommend to get the most data possible but still be able to handle quite a bit of traffic? In the end of it all, these alerts will be shipped to a central server via a SSH tunnel. I'm trying to stay away from databases.

  Read the article

• ipv6 port 445 does not accept the request from a global type address

  - by blacktea

  I want to scan the port 445 in windows server 2003, but my scanner only have one type ipv6 address which is global not link-local. When I do this,I find that I can't find port 445 open. But I use the command "netstat -an" to assure the port 445 is listening. Finally I find this confusing phenomenon: 1.when I set a link-local ddress in my scanner, then it will work in scanning port 445. 2.when I only set a global address in my scanner, it doed not work. This means if a host with a link-local address use socket to send a syn packet to the port 445 in server 2003, it will receive a ack packet. But if with a global address it will receive a rst packet. Thus, I can't scan the port 445 in server 2003 with a global address. I need to know why? Can anybody help? And I use the netsh-firewall to check the exception and netsh-interface-ipv6 to turn off the firewall on the specific interface. Still can't establish the connection with port 445, do you have any ideal about this ?

  Read the article

• DHCP Server on local machine

  - by EralpB

  Hello I am trying to setup a dhcp3-server on Ubuntu. But my question is more generic, if dhcp server is in a blockbox and all clients are connected to it I think I get what is going on but when dhcp server is installed on one of the "clients" that confuses me. When I send a dhcp packet from that client to the dhcp server, will my ethernet card read and write at the same time? Or will it handle it internally without writing any data to ethernet cable. It's the first time I am encountering these network things so I am a little bit confused. Also I wonder If I am in a big network with lan IP let's say 192.168.0.100 and I install a dhcp server to my computer, can any other computers accidentally get IP from my dhcp server? Every computer has one ethernet card (if that matters?). And every computer is connected to one router. I guess the answer is no because the broadcast message won't reach to my computer since when router receives a dhcp search packet it will answer and it won't let other computers know about it because they don't need to. And without router sending that packet one by one, it cannot travel further. I'd be glad if someone enlightens me. Thank you very much.

  Read the article

• VMware ARP/Mac Networking

  - by Ross Wilson

  Hi Guys, I am very interested in how VMware networking works. I have scoured the VMware website and read their data sheets, this has given me some basic knowledge. I now have some questions. Lets assume that we have a physical server running the VMware hypervisor. The physical server is running a Virtual Machine. The physical box has one physical NIC. The NIC is connected to a switch, as so is a desktop client. Now, this is where my first question lies. The VM has an IP address: 192.168.1.1. How do desktop clients on the network communicate with this VM? So, the client pings 192.168.1.1. The ping packet is sent to the switch. The switch checks its MAC address table and sees that 192.168.1.1 is associated with the MAC address of the physical NIC. Correct? I then assume that the ping packet is sent to the server's physical NIC, where the hypervisor routes the packet to the VM thats using 192.168.1.1? Please could you give me a run down as to how VM networking works? Many thanks, Ross

  Read the article

• Only one domains not resolving via Windows DNS server at multiple locations, but is at others

  - by Brett G

  I'm having quite a weird issue. Had mail delivery issues to a specific domain. After looking closer, I realized that the DNS for that domain isn't resolving via the in-house Windows 2003 SP2 DNS server. C:\>nslookup foodmix.net Server: DC.DOMAIN.com Address: 10.1.1.1 DNS request timed out. timeout was 2 seconds. DNS request timed out. timeout was 2 seconds. *** Request to DC.DOMAIN.com timed-out (DC.DOMAIN.com and 10.1.1.1 are generic values to replace the actual ones) Even if I run this nslookup from the DC.DOMAIN.com server, I get the same result. However, all other requests are working as they should. I tried it on severs at completely separate organizations on different networks(Windows 2003 AD servers). The weird thing is some of these were having the same exact issue. However using public DNS servers work. I have tried clearing the DNS cache, restarting the server, restarting the services, etc. Nothing has worked. One weird event I noticed in the DNS Server Event Logs that might be related is an event ID of 5504 with the following description: The DNS server encountered an invalid domain name in a packet from 192.33.4.12. The packet will be rejected. The event data contains the DNS packet. For more information, see Help and Support Center at http://go.microsoft.com/fwlink/events.asp. In the data section below, I can see the following mentioned: ns2.webhostingstar.com Which happens to be the nameserver for the domain in question. Several discussion threads and a MS KB have pointed to disabling EDNS. I have done this via "dnscmd /config /enableednsprobes 0" and it has not fixed the issue.

  Read the article

• Make isolinux 4.0.3 chainload itself in VMWare

  - by chainloader

  I have a bootable iso which boots into isolinux 4.0.3 and I want to make it chainload itself (my actual goal is to chainload isolinux.bin v4.0.1-debian, which should start up the Ubuntu10.10 Live CD, but for now I just want to make it chainload itself). I can't get isolinux to chainload any isolinux.bin, no matter what version. It either freezes or shows a "checksum error" message. I'm using VMWare to test the iso. Things I have tried: .com32 /boot/isolinux/chain.c32 /boot/isolinux/isolinux-debug.bin (chainload self) this shows Loading the boot file... Booting... ISOLINUX 4.03 2010-10-22 Copyright (C) 1994-2010 H. Peter Anvin et al isolinux: Starting up, DL = 9F isolinux: Loaded spec packet OK, drive = 9F isolinux: Main image LBA = 53F00100 ...and the machine freezes. Then I've tried this (chainload GRUB4DOS 0.4.5b) chainloader /boot/isolinux/isolinux-debug.bin Result: Error 13: Invalid or unsupported executable format Next try: (chainload GRUB4DOS 0.4.5b) chainloader --force /boot/isolinux/isolinux-debug.bin boot Result: ISOLINUX 4.03 2010-10-22 Copyright (C) 1994-2010 H. Peter Anvin et al isolinux: Starting up, DL = 9F isolinux: Loaded spec packet OK, drive = 9F isolinux: No boot info table, assuming single session disk... isolinux: Spec packet missing LBA information, trying to wing it... isolinux: Main image LBA = 00000686 isolinux: Image checksum error, sorry... Boot failed: press a key to retry... I have tried other things, but all of them failed miserably. Any suggestions?

  Read the article

• Do all routers really must know all routes to every router?

  - by Philipili

  This is my complicated and long question. First let's talk about the context. Network topology: PC A --- RT A --- RT C --- RT B --- PC B (RT C has a WAN NIC connected to "the cloud") With this situation : PC A must send a packet to PC B Default routes direct packets to the cloud We haven't access to RT C's configuration RT C only knows how to join network A, not network B RT A knows about network B RT B knows about network A RT C's routing table: Destination NIC Gateway 0.0.0.0 WAN Cloud Network A LAN A RT A's WAN RT A's routing table: Destination NIC Gateway 0.0.0.0 WAN LAN A Network B WAN LAN A RT B's routing table: Destination NIC Gateway 0.0.0.0 WAN LAN B Network A WAN LAN B I would like to permit PC A and PC B to communicate, but I don't have access to RT C. Networks B and BC are new. Can PC A send a packet to RT B's WAN NIC (which is possible) and "ask RT B to direct the packet to PC B" ? I believe replacing RT B with a VPN server should do the trick, but I would like to know if it is possible to make it without establishing a new connection.

  Read the article

• Make isolinux 4.0.3 chainload itself

  - by chainloader

  I have a bootable iso which boots into isolinux 4.0.3 and I want to make it chainload itself (my actual goal is to chainload isolinux.bin v4.0.1-debian, which should start up the Ubuntu10.10 Live CD, but for now I just want to make it chainload itself). I can't get isolinux to chainload any isolinux.bin, no matter what version. It either freezes or shows a "checksum error" message. I'm using VMWare to test the iso. Things I have tried: .com32 /boot/isolinux/chain.c32 /boot/isolinux/isolinux-debug.bin (chainload self) this shows Loading the boot file... Booting... ISOLINUX 4.03 2010-10-22 Copyright (C) 1994-2010 H. Peter Anvin et al isolinux: Starting up, DL = 9F isolinux: Loaded spec packet OK, drive = 9F isolinux: Main image LBA = 53F00100 ...and the machine freezes. Then I've tried this (chainload GRUB4DOS 0.4.5b) chainloader /boot/isolinux/isolinux-debug.bin Result: Error 13: Invalid or unsupported executable format Next try: (chainload GRUB4DOS 0.4.5b) chainloader --force /boot/isolinux/isolinux-debug.bin boot Result: ISOLINUX 4.03 2010-10-22 Copyright (C) 1994-2010 H. Peter Anvin et al isolinux: Starting up, DL = 9F isolinux: Loaded spec packet OK, drive = 9F isolinux: No boot info table, assuming single session disk... isolinux: Spec packet missing LBA information, trying to wing it... isolinux: Main image LBA = 00000686 isolinux: Image checksum error, sorry... Boot failed: press a key to retry... I have tried other things, but all of them failed miserably. Any suggestions?

  Read the article

• Prevent outgoing traffic unless OpenVPN connection is active using pf.conf on Mac OS X

  - by Nick

  I've been able to deny all connections to external networks unless my OpenVPN connection is active using pf.conf. However, I lose Wi-Fi connectivity if the connection is broken by closing and opening the laptop lid or toggling Wi-Fi off and on again. I'm on Mac OS 10.8.1. I connect to the Web via Wi-Fi (from varying locations, including Internet cafés). The OpenVPN connection is set up with Viscosity. I have the following packet filter rules set up in /etc/pf.conf # Deny all packets unless they pass through the OpenVPN connection wifi=en1 vpn=tun0 block all set skip on lo pass on $wifi proto udp to [OpenVPN server IP address] port 443 pass on $vpn I start the packet filter service with sudo pfctl -e and load the new rules with sudo pfctl -f /etc/pf.conf. I have also edited /System/Library/LaunchDaemons/com.apple.pfctl.plist and changed the line <string>-f</string> to read <string>-ef</string> so that the packet filter launches at system startup. This all seems to works great at first: applications can only connect to the web if the OpenVPN connection is active, so I'm never leaking data over an insecure connection. But, if I close and reopen my laptop lid or turn Wi-Fi off and on again, the Wi-Fi connection is lost, and I see an exclamation mark in the Wi-Fi icon in the status bar. Clicking the Wi-Fi icon shows an "Alert: No Internet connection" message: To regain the connection, I have to disconnect and reconnect Wi-Fi, sometimes five or six times, before the "Alert: No Internet connection" message disappears and I'm able to open the VPN connection again. Other times, the Wi-Fi alert disappears of its own accord, the exclamation mark clears, and I'm able to connect again. Either way, it can take five minutes or more to get a connection again, which can be frustrating. Why does Wi-Fi report "No internet connection" after losing connectivity, and how can I diagnose this issue and fix it?

  Read the article

• Server 2012, Jumbo Frames - should I expect problems?

  - by TomTom

  Ok, this sound might stupid - but is there any negative on just enabling jumbo frames in practice? From what I understand: Any switch or ethernet adapter that sees a jumbo frame it can not handle will just drop it. TCP is not a problem as max frame size is negotiated in the setinuo phase. UCP is a theoretical problem as a server may just send a LARGE UDP packet that gets dropped on the way. Practically though, as UDP is packet based, I do not really think any software WOULD send a UDP packet larger than 1500 bytes net without app level configuration changes - at least this is how I do my programming, as it is quite hard to get a decent MTU size for that without testing yourself, so you fall back in programming to max 1500 packets. The network in question is a standard small business network - we upgraded now from a non managed 24 port switch to a 52 port switch with 4 10g ports (netgear - quite cheap) and will mov a file server to 10g for also ISCSI serving. All my equipment on the Ethernet level can handle minimum 9000 bytes and due to local firewalls I really want to get packets larger (less firewall processing), but the network is also NAT'ed to the internet. On top, different machines move around (download) large files (multi gigabyte area) quite often for processing. The question is - can I expect problems when I just enable jumbo frames? Again, this is not totally ignorance - I just don't see programs sending more than 1500 byte UDP packets (if that is a practical problem please tell me) and for TCP the MTU is negotiated anyway. if there is a problem I can move to a dedicated VLAN, but this has it's own shares of problems as basically most workstations must then be on both VLAN's.

  Read the article

• Why Wireshark does not recognize this HTTP response?

  - by Alois Mahdal

  I have a trivial CGI script that outputs simple text content. It's written in Perl and using CGI module and it specifies only the most basic headers: print $q->header( -type => 'text/plain', -Content_length => $length, ); print $stuff; There's no apparent issue with functionality, but I'm confused about the fact that Wireshark does not recognize the HTTP response as HTTP--it's marked as TCP. Here is request and response: GET /cgi-bin/memfile/memfile.pl?mbytes=1 HTTP/1.1 Host: 10.6.130.38 User-Agent: Mozilla/5.0 (Windows NT 6.1; WOW64; rv:11.0) Gecko/20100101 Firefox/11.0 Accept: text/html,application/xhtml+xml,application/xml;q=0.9,*/*;q=0.8 Accept-Language: cs,en-us;q=0.7,en;q=0.3 Accept-Encoding: gzip, deflate Connection: keep-alive HTTP/1.1 200 OK Date: Thu, 05 Apr 2012 18:52:23 GMT Server: Apache/2.2.15 (Win32) mod_ssl/2.2.15 OpenSSL/0.9.8m Content-length: 1048616 Keep-Alive: timeout=5, max=100 Connection: Keep-Alive Content-Type: text/plain; charset=ISO-8859-1 XXXXXXXX... And here is the packet overview (Full packet is here on pastebin) No. Time Source srcp Destination dstp Protocol Info tcp.stream abstime 5 0.112749 10.6.130.38 80 10.6.130.53 48072 TCP [TCP segment of a reassembled PDU] 0 20:52:23.228063 Frame 5: 1514 bytes on wire (12112 bits), 1514 bytes captured (12112 bits) Ethernet II, Src: Dell_97:29:ac (00:1e:4f:97:29:ac), Dst: Dell_3b:fe:70 (00:24:e8:3b:fe:70) Internet Protocol Version 4, Src: 10.6.130.38 (10.6.130.38), Dst: 10.6.130.53 (10.6.130.53) Transmission Control Protocol, Src Port: http (80), Dst Port: 48072 (48072), Seq: 1, Ack: 330, Len: 1460 Now when I see this in Wireshark: there's usual TCP handshake then the GET request shown as HTTP with preview then the next packet contains the response, but is not marked as an HTTP response--just a generic "[TCP segment of a reassembled PDU]", and is not caught by "http.response" filter. Can somebody explain why Wireshark does not recognize it? Is there something wrong with the response?

  Read the article

• Router vs switch in a LAN [closed]

  - by servernewbie

  If I have a LAN and and connect it with a switch, I understand it uses a CAM table to route packets in layer 2 (by saving mac to port relations). So far all good. However, when using a router for a LAN (ONLY for a LAN, not to connect it to "the outside" WAN/internet/etc) I get a bit confused as to how it internally processes packets. I would first split this into two router scenarios: Router with buit-in switch In this scenario, I would expect that it will act exactly as a switch with a CAM table internally. This would probably benefit a bit in speed (guessing here?) compared to the next option. Router without built-in switch Here is where I get confused. If hostA wants to send a packet to hostB, it will ARP to find hostB's MAC address and send it there. Now, if we had a switch (above scenario) this would be easy. But how does it work now in a router WITHOUT a switch? If I would guess, hostA would send an Ethernet frame with hostB's MAC address to the line. The router would fetch the packet (even though the router has another MAC address, it would still fetch this packet even if it only contains hostB's MAC address). It would strip the Ethernet frame header and check the IP, and then check its own internal ARP table again for the MAC address. Now, this would seem like a waste of resources compared to a router with a built-in switch. But maybe it does not work like that at all. Does it also contain a CAM table? If that would be true, what would then the difference between these two routers really be?

  Read the article

• How to prevent asymmetric routing with multiple eBGP routers?

  - by Andy Shinn

  I have 2 routers announcing a /22 subnet to different providers (one providers connects to each of the 2 routers). I have split the /22 in two /23 to announce one /23 on each of the routers plus the /22 (the providers will take the more specific route). This allows me to fail over and keep traffic inside the /23 in and out the same provider. What are other ways in which I could announce just the /22 with both routers and have packets from servers on the network behind the routers go back out the same router in which they came in from? EDIT: The main problem I come across, which end users and clients complain about the most, is that the least hop route is sometimes not the "optimal" route. In my case, I know that Provider B may have better latency to X nation. But when packets come in from provider B, they may go out Provider A or provider B. The reverse is also true. If I send a packet to X nation out provider A, even though it may have more hops back, the packet will likely come in from Provider B (which may have higher latency, packet loss, etc. to this nation)

  Read the article

• NAT and P2P router crash

  - by returnFromException

  So..i had this argument with my networks teacher. He said that some people complains about router crashes due to many entrys on NAT tables on a router. I didnt understand and i asked: "If the application uses the same port, why does the router crash?. It should have only one entry (pc-ip,pcport;public-ip,public-port)". And he said: "it doesnt matter its using the same port". I got the idea that NAT creates an entry for every packet that passes trought it. Iam assuming NAT with overloading as you might have guessed. So the questions are: 1-How does nat entrys are created? On a packet basis or connection basis? I mean: suppose i send a udp packet..does the router create an entry? 2-When i start a TCP connection, does the router create a persistant nat entry until the connection closes? 3-Was my teacher right? The NAT table can overload assuming an aplication on the same port sending packets? Thanks in advance.

  Read the article

• Wireshark WPA 4-way handshake

  - by cYrus

  From this wiki page: WPA and WPA2 use keys derived from an EAPOL handshake to encrypt traffic. Unless all four handshake packets are present for the session you're trying to decrypt, Wireshark won't be able to decrypt the traffic. You can use the display filter eapol to locate EAPOL packets in your capture. I've noticed that the decryption works with (1, 2, 4) too, but not with (1, 2, 3). As far as I know the first two packets are enough, at least for what concern unicast traffic. Can someone please explain exactly how does Wireshark deal with that, in other words why does only the former sequence work, given that the fourth packet is just an acknowledgement? Also, is it guaranteed that the (1, 2, 4) will always work when (1, 2, 3, 4) works? Test case This is the gzipped handshake (1, 2, 4) and an ecrypted ARP packet (SSID: SSID, password: password) in base64 encoding: H4sICEarjU8AA2hhbmRzaGFrZS5jYXAAu3J400ImBhYGGPj/n4GhHkhfXNHr37KQgWEqAwQzMAgx 6HkAKbFWzgUMhxgZGDiYrjIwKGUqcW5g4Ldd3rcFQn5IXbWKGaiso4+RmSH+H0MngwLUZMarj4Rn S8vInf5yfO7mgrMyr9g/Jpa9XVbRdaxH58v1fO3vDCQDkCNv7mFgWMsAwXBHMoEceQ3kSMZbDFDn ITk1gBnJkeX/GDkRjmyccfus4BKl75HC2cnW1eXrjExNf66uYz+VGLl+snrF7j2EnHQy3JjDKPb9 3fOd9zT0TmofYZC4K8YQ8IkR6JaAT0zIJMjxtWaMmCEMdvwNnI5PYEYJYSTHM5EegqhggYbFhgsJ 9gJXy42PMx9JzYKEcFkcG0MJULYE2ZEGrZwHIMnASwc1GSw4mmH1JCCNQYEF7C7tjasVT+0/J3LP gie59HFL+5RDIdmZ8rGMEldN5s668eb/tp8vQ+7OrT9jPj/B7425QIGJI3Pft72dLxav8BefvcGU 7+kfABxJX+SjAgAA Decode with: $ base64 -d | gunzip > handshake.cap Run tshark to see if it correctly decrypt the ARP packet: $ tshark -r handshake.cap -o wlan.enable_decryption:TRUE -o wlan.wep_key1:wpa-pwd:password:SSID It should print: 1 0.000000 D-Link_a7:8e:b4 - HonHaiPr_22:09:b0 EAPOL Key 2 0.006997 HonHaiPr_22:09:b0 - D-Link_a7:8e:b4 EAPOL Key 3 0.038137 HonHaiPr_22:09:b0 - D-Link_a7:8e:b4 EAPOL Key 4 0.376050 ZyxelCom_68:3a:e4 - HonHaiPr_22:09:b0 ARP 192.168.1.1 is at 00:a0:c5:68:3a:e4

  Read the article

• Openvpn - stuck on Connecting

  - by user224277

  I've got a problem with openvpn server... every time when I trying to connect to the VPN , I am getting a window with login and password box, so I typed my login and password (login = Common Name (user1) and password is from a challenge password from the client certificate. Logs : Jun 7 17:03:05 test ovpn-openvpn[5618]: Authenticate/Decrypt packet error: packet HMAC authentication failed Jun 7 17:03:05 test ovpn-openvpn[5618]: TLS Error: incoming packet authentication failed from [AF_INET]80.**.**.***:54179 Client.ovpn : client #dev tap dev tun #proto tcp proto udp remote [Server IP] 1194 resolv-retry infinite nobind persist-key persist-tun ca ca.crt cert user1.crt key user1.key <tls-auth> -----BEGIN OpenVPN Static key V1----- d1e0... -----END OpenVPN Static key V1----- </tls-auth> ns-cert-type server cipher AES-256-CBC comp-lzo yes verb 0 mute 20 My openvpn.conf : port 1194 #proto tcp proto udp #dev tap dev tun #dev-node MyTap ca /etc/openvpn/keys/ca.crt cert /etc/openvpn/keys/VPN.crt key /etc/openvpn/keys/VPN.key dh /etc/openvpn/keys/dh2048.pem server 10.8.0.0 255.255.255.0 ifconfig-pool-persist ipp.txt #push „route 192.168.5.0 255.255.255.0? #push „route 192.168.10.0 255.255.255.0? keepalive 10 120 tls-auth /etc/openvpn/keys/ta.key 0 #cipher BF-CBC # Blowfish #cipher AES-128-CBC # AES #cipher DES-EDE3-CBC # Triple-DES comp-lzo #max-clients 100 #user nobody #group nogroup persist-key persist-tun status openvpn-status.log #log openvpn.log #log-append openvpn.log verb 3 sysctl : net.ipv4.ip_forward=1

  Read the article

• IPv6 works only after ping to routing box

  - by Ficik

  Situation: There is ipv4 only router in network and every computer is connected to it (wifi or cable). Server with ipv4 and ipv6 is connected to this router as well. Server has configured tunnelbrokers 6to4 tunnel and radvd. Clients in network has right prefix and can ping each other. But they can't ping to internet until they ping Server (the one with tunnel). I found somewhere that it's icmp problem, but I couldn't find solution. Is it problem that there is ipv4 only router? server and client runs linux router runs dd-wrt without ipv6 support :( Ping try: standa@standa-laptop:~$ ping6 ipv6.google.com PING ipv6.google.com(2a00:1450:8007::69) 56 data bytes ^C --- ipv6.google.com ping statistics --- 29 packets transmitted, 0 received, 100% packet loss, time 28223ms standa@standa-laptop:~$ ping6 2001:470:XXXX:XXXX:21c:c0ff:fe2b:6478 PING 2001:470:XXXX:XXXX:21c:c0ff:fe2b:6478(2001:470:XXXX:XXXX:21c:c0ff:fe2b:6478) 56 data bytes 64 bytes from 2001:470:XXXX:XXXX:21c:c0ff:fe2b:6478: icmp_seq=1 ttl=64 time=3.55 ms 64 bytes from 2001:470:XXXX:XXXX:21c:c0ff:fe2b:6478: icmp_seq=2 ttl=64 time=0.311 ms 64 bytes from 2001:470:XXXX:XXXX:21c:c0ff:fe2b:6478: icmp_seq=3 ttl=64 time=0.269 ms 64 bytes from 2001:470:XXXX:XXXX:21c:c0ff:fe2b:6478: icmp_seq=4 ttl=64 time=0.292 ms ^C --- 2001:470:XXXX:XXXX:21c:c0ff:fe2b:6478 ping statistics --- 4 packets transmitted, 4 received, 0% packet loss, time 3000ms rtt min/avg/max/mdev = 0.269/1.107/3.559/1.415 ms standa@standa-laptop:~$ ping6 ipv6.google.com PING ipv6.google.com(2a00:1450:8007::69) 56 data bytes 64 bytes from 2a00:1450:8007::69: icmp_seq=1 ttl=57 time=20.7 ms 64 bytes from 2a00:1450:8007::69: icmp_seq=2 ttl=57 time=20.2 ms 64 bytes from 2a00:1450:8007::69: icmp_seq=3 ttl=57 time=23.4 ms ^C --- ipv6.google.com ping statistics --- 3 packets transmitted, 3 received, 0% packet loss, time 2001ms rtt min/avg/max/mdev = 20.267/21.479/23.413/1.392 ms

  Read the article

• Only one domain is not resolving via Windows DNS server at multiple locations, but is at others

  - by Brett G

  I'm having quite a weird issue. Had mail delivery issues to a specific domain. After looking closer, I realized that the DNS for that domain isn't resolving via the in-house Windows 2003 SP2 DNS server. C:\>nslookup foodmix.net Server: DC.DOMAIN.com Address: 10.1.1.1 DNS request timed out. timeout was 2 seconds. DNS request timed out. timeout was 2 seconds. *** Request to DC.DOMAIN.com timed-out (DC.DOMAIN.com and 10.1.1.1 are generic values to replace the actual ones) Even if I run this nslookup from the DC.DOMAIN.com server, I get the same result. However, all other requests are working as they should. I had a sysadmin friend try this DNS lookup on servers at several companies that he consults for (which are also Windows 2003 AD servers). The weird thing is some of these were having the same exact issue. However using public DNS servers work. I have tried clearing the DNS cache, restarting the server, restarting the services, etc. Nothing has worked. One weird event I noticed in the DNS Server Event Logs that might be related is an event ID of 5504 with the following description: The DNS server encountered an invalid domain name in a packet from 192.33.4.12. The packet will be rejected. The event data contains the DNS packet. For more information, see Help and Support Center at http://go.microsoft.com/fwlink/events.asp. In the data section below, I can see the following mentioned: ns2.webhostingstar.com Which happens to be the nameserver for the domain in question. Several discussion threads and a MS KB have pointed to disabling EDNS. I have done this via "dnscmd /config /enableednsprobes 0" and it has not fixed the issue.

  Read the article

< Previous Page | 18 19 20 21 22 23 24 25 26 27 28 29  | Next Page >