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  • Cisco ASA site-to-site vpn not initiating phase 1 (not sending udp 500 packets)

    - by Sean Steadman
    I am hoping someone here can help me with my problem. I am trying to setup an IPSEC site-to-site VPN between two cisco ASA 5520's in GNS3 (both using 8.4.2). I have been unsuccesful in getting the tunnel up and it appears neither ASA is sending packets out,in regards to phase 1 and phase 2 (tested by using wireshark and seeing NO udp 500 packets). Doing show ipsec sa and such shows nothing. CALIFORNIA(config)# show ipsec sa There are no ipsec sas FLA-ASA# show ipsec sa There are no ipsec sas I will attach both configurations in two different pastebin files as to keep this post a bit cleaner. Essentially California side has 172.20.1.0/24 and Florida side has 10.10.10.0/24. California ASA config: http://pastebin.com/v0pngYzF Florida ASA config: http://pastebin.com/E2geybta Please let me know if there is any other vital information that could help. I have gotten IPSEC tunnels to work using openSwan (linux) and cisco routers but cannot for the life of me get ASA IPSEC tunnels to work. The ASDM is out of the question I only use cli. Thanks for any useful help!

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  • Android stream to Wowza

    - by Curtis Kiu
    I feel very confused about Android streaming to wowza. I am doing a video conference using rtmp cross-platform, but Android doesn't eat RTMP. Therefore I need to find another way to do it. Upstreaming I found a new open-source app called spydroid-ipcamera. It is using rtp, sending udp packets to computer, and opens it in vlc using the following sdp v=0 s=Unnamed m=video 5006 RTP/AVP 96 a=rtpmap:96 H264/90000 a=fmtp:96 packetization-mode=1;profile-level-id=420016;sprop-parameter-sets=Z0IAFukBQHsg,aM4BDyA=; But it can't work. Then I follow wowza tutorial and stream to it and then play again in VLC. That works! I wrote it in http://code.google.com/p/spydroid-ipcamera/issues/detail?id=2 However when I want to add audio in the packet, it fails to work. I change to code in http://code.google.com/p/spydroid-ipcamera/source/browse/trunk/src/net/mkp/spydroid/CameraStreamer.java mr.setAudioSource(MediaRecorder.AudioSource.MIC); mr.setVideoSource(MediaRecorder.VideoSource.CAMERA); mr.setOutputFormat(MediaRecorder.OutputFormat.MPEG_4); mr.setVideoFrameRate(20); mr.setVideoSize(640, 480); mr.setAudioEncoder(MediaRecorder.AudioEncoder.AAC); mr.setVideoEncoder(MediaRecorder.VideoEncoder.H264); mr.setPreviewDisplay(holder.getSurface()); Then I thought that the problem should be in sdp, but I don't know how to due with sdp. I am streaming H.264/AAC with Mp4 Second I don't understand sdp. So how can I make video conference upstreaming part using this apps. Android ----(UDP Port:5006)----> PC (SDP file) and then Wowza read the SDP file ------> VLC I think in this way the system cannot handle more than 1 client. sdp can only hold 1 port, any idea or actually it wont' work? Also Wowza need to set the stream before we stream it, so does it mean that I should not follow this way to do it? Sorry my English is poor, I hope you guys understand.

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  • Broadcasting

    - by Avik
    Could anyone please provide me with the code or link to send and receive broadcast messages if possible using UDP? I have been stuck in a problem and hope if u guys could help me resolve it. Thanks

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  • [Android] SocketException (The system call was cancelled)

    - by cat
    Hi. I am working on a android project and I need to connect the gPhone with the PC using UDP socket. The same operation, sending a message to from gPhone to the PC, returns different results in different time: Sometimes succeeded and sometimes throws a SocketExcepton(The system call was cancelled). Can anyone tell me what possibly results in this SocketException? Thanks in advance!

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  • What is the best way to do Bit Field manipulation in Python?

    - by ZebZiggle
    I'm reading some MPEG Transport Stream protocol over UDP and it has some funky bitfields in it (length 13 for example). I'm using the "struct" library to do the broad unpacking, but is there a simple way to say "Grab the next 13 bits" rather than have to hand-tweak the bit manipulation? I'd like something like the way C does bit fields (without having to revert to C). Suggestions?

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  • C# UDP Socket taking time to send data to unknown IP

    - by Mohsan
    Hi. i am sending data to UDP socket using this code Socket udpClient = new Socket(AddressFamily.InterNetwork, SocketType.Dgram, ProtocolType.Udp); IPEndPoint ipEndPoint = new IPEndPoint(IPAddress.Parse(obj.destAddress), obj.destPort); byte[] buf = new byte[obj.length]; Array.Copy((byte[])obj.data, buf, obj.length); int n = udpClient.SendTo(buf, ipEndPoint); udpClient.Close(); this code works fine when IP exists in current network, but it takes 3-5 seconds when I send data to unknown IP address. This causes main application to hang for 3-5 seconds.. What could be the reason behind this problem..

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  • No network packets sent immediately after quick physical disconnect and reconnect.

    - by Hans
    I am using Boost's ASIO libraries to establish a UDP connection to a remote server. To make sure the connection is active, every second a keep-alive message is sent to the server. I have noticed that if I unplug the network cable and reinsert it quickly, the first 2 or 3 keep-alive messages after the reinsert are never sent. I tested this by running wire-shark on the server. I have seen it take up to 5 seconds before the client starts sending out network traffic again. The client is running under Linux (2.6.2), if that helps.

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  • Android Broadcast Address

    - by Eef
    Hey, I am making a Client Server application for my Android phone. I have created a UDP Server in Python which sits and listens for connections. I can put either the server IP address in directly like 192.169.0.100 and it sends data fine. I can also put in 192.168.0.255 and it find the server on 192.169.0.100. Is it possible to get the broadcast address of the network my Android phone is connected to? I am only ever going to use this application on my Wifi network or other Wifi networks. Cheers

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  • Pinging CS Servers

    - by Zubair1
    Hello, This has been bothering me for awhile, can some one show me how to ping a counter strike server. I just want to ping the server and see if it is online, thats all. I found many small snippets online that were using fsock and UDP to do this but none of them actually did the job i wanted it to do. Most of the ones i found were showing offline servers as online. I would really really appreciate if some one could provide me with this useful information (code). Thank you in advance ^_^

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  • How do I write my own desktop sharing application in java ?

    - by nomemory
    Hello I want write my own desktop sharing application in Java. The application should have some very default features: Capture screen; Allow a remote connected user to click / edit fields. I was thinking to use Java Robot class for mouse movements / key pressing. The problem is i don't know what screen capture strategy to use. Should I make sequentially screen captures (on the hosting computer) every second, and send those captures with UDP via network, so that the clients can intercept the data-grams ? Isn't this a little overkill for the network ? What other strategies are available ? (Except trying an already existing app). PS: If necessary I can even write native code using JNI (still that's the last thing I planning to do).

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  • Discovering maximum packet size

    - by ereOn
    I'm working on a network-related project and I am using DTLS (TLS/UDP) to secure communications. Reading the specifications for DTLS, I've noted that DTLS requires the DF flag (Don't Fragment) to be set. On my local network if I try to send a message bigger than 1500 bytes, nothing is sent. That makes perfect sense. On Windows the sendto() reports a success but nothing is sent. I obviously cannot unset the DF flag manually since it is mandatory for DTLS and i'm not sure whether the 1500 bytes limit (MTU ?) could change in some situations. I guess it can. So, my question is : "Is there a way to discover this limit ?" If not, what would be the lowest possible value ? My software runs under UNIX (Linux/MAC OSX) and Windows OSes so different solutions for each OS are welcome ;) Many thanks.

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  • Count of memory copies in *nix systems between packet at NIC and user application?

    - by Michael_73
    Hi there, This is just a general question relating to some high-performance computing I've been wondering about. A certain low-latency messaging vendor speaks in its supporting documentation about using raw sockets to transfer the data directly from the network device to the user application and in so doing it speaks about reducing the messaging latency even further than it does anyway (in other admittedly carefully thought-out design decisions). My question is therefore to those that grok the networking stacks on Unix or Unix-like systems. How much difference are they likely to be able to realise using this method? Feel free to answer in terms of memory copies, numbers of whales rescued or areas the size of Wales ;) Their messaging is UDP-based, as I understand it, so there's no problem with establishing TCP connections etc. Any other points of interest on this topic would be gratefully thought about! Best wishes, Mike

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  • How can I diagnose "Cannot determine peer address" in my Perl TCP script?

    - by MadBoy
    I've this little script which does it's job pretty well but sometimes it tends to fail. It fails in 2 cases: with error send: Cannot determine peer address at ./tcp-new.pl line 52 with no output or anything, it just fails to deliver what it got to connected Tcp Client. Usually it happens after I disconnect from server, go home and connect it again. To fix this restart is required and it starts working. Sometimes this problem is followed by problem mentioned in point 1. Note: it's not problem when I disconnect and reconnect to it again within short amount of time (unless error nr 1 happens). So can anyone help me make this code be a bit more stable so I don't have to restart it every day? #!/usr/bin/perl use strict; use warnings; use IO::Socket; use IO::Select; my $tcp_port = "10008"; my $udp_port = "2099"; my $tcp_socket = IO::Socket::INET->new( Listen => SOMAXCONN, LocalPort => $tcp_port, Proto => 'tcp', ReuseAddr => 1, ); my $udp_socket = IO::Socket::INET->new( LocalPort => $udp_port, Proto => 'udp', ); my $read_select = IO::Select->new(); my $write_select = IO::Select->new(); $read_select->add($tcp_socket); $read_select->add($udp_socket); while (1) { my @read = $read_select->can_read(); foreach my $read (@read) { if ($read == $tcp_socket) { my $new_tcp = $read->accept(); $write_select->add($new_tcp); } elsif ($read == $udp_socket) { my $recv_buffer; $udp_socket->recv($recv_buffer, 1024, undef); my @write = $write_select->can_write(); foreach my $write (@write) { $write->send($recv_buffer); } } } }

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  • C - dns query to structure

    - by Bibo
    I have these structures: typedef struct dnsQuery { char header[12]; struct dnsQuerySection *querySection; } TdnsQuery; typedef struct dnsQuerySection { unsigned char *name; struct dnsQueryQuestion *question; } TdnsQuerySection; typedef struct dnsQueryQuestion { unsigned short qtype; unsigned short qclass; } TdnsQueryQuestion; and I have dns query in byte array from recvfrom. I am trying to get structure from byte array like this: TdnsQuery* dnsQuery = (TdnsQuery*)buf; When I tried to access qtype like this: printf("%u", dnsQuery->querySection->question.qtype); I get seg fault 11. Can someone help me with these structures? What´s wrong with them? I tried to add structure: typedef struct udpPacket { char header[8]; structr dnsQuery query; } and mapped this structure from byte array but it didn´t help. Can someone help me with these structures? How they should look like for dns query with udp protocol?

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  • Timer C#. Start, stop, and get the amount of time between the calls

    - by user1886060
    I'm writing UDP chat with reliable data transfer. I need to start a timer when a packet is sent, and stop it as soon it receives an answer from the server(ACK- acknowledgment). Here is my code: private void sendButton_Click(object sender, EventArgs e) { Packet snd = new Packet(ack, textBox1.Text.Trim()); textBox1.Text = string.Empty; Smsg = snd.GetDataStream();//convert message into array of bytes to send. while (true) { try { // Here I need to Start a timer! clientSock.SendTo(Smsg, servEP); clientSock.ReceiveFrom(Rmsg, ref servEP); //Here I need to stop a timer and get elapsed amount of time. Packet rcv = new Packet(Rmsg); if (Rmsg != null && rcv.ACK01 != ack) continue; if (Rmsg != null && rcv.ACK01 == ack) { this.displayMessageDelegate("ack is received :"+ack); ChangeAck(ack); break; } Thank you.

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  • Receiving Multicast Messages on a Multihomed Windows PC

    - by Basti
    I'm developing a diagnostic tool on a PC with several Network Interfaces based on multicast/udp. The user can select a NIC, the application creates sockets, binds them to this NIC and adds them to the specific multicast group. The sending of multicast messages works fine. However receiving of messages only succeeds if I bind the sockets to a specific NIC of my PC. It almost looks like as there is a 'default' NIC for receiving multicast messages in Windows which is always the first NIC returned by the GetAdapterInfo function. I monitored the network with Wireshark and discovered that the "IGMP Join Group" message isn't sent from the NIC I bound the socket at, but by this 'default' NIC. If I disable this NIC (or remove the network cable), the next NIC of the list returned by GetAdapterInfo is used for receiving multicast messages. I was successful to change this 'default' NIC by adding an additional entry to the routing table of my PC, but I don't think this is a good solution of the problem. The problem also occurs with the code appended below. The join group messages isn't sent via 192.168.52 but via a different NIC. // socket_tst.cpp : Defines the entry point for the console application. // #include tchar.h #include winsock2.h #include ws2ipdef.h #include IpHlpApi.h #include IpTypes.h #include stdio.h int _tmain(int argc, _TCHAR* argv[]) { WSADATA m_wsaData; SOCKET m_socket; sockaddr_in m_sockAdr; UINT16 m_port = 319; u_long m_interfaceAdr = inet_addr("192.168.1.52"); u_long m_multicastAdr = inet_addr("224.0.0.107"); int returnValue = WSAStartup(MAKEWORD(2,2), &m_wsaData); if (returnValue != S_OK) { return returnValue; } // Create sockets if (INVALID_SOCKET == (m_socket = socket(AF_INET, SOCK_DGRAM, IPPROTO_UDP)) ) { return WSAGetLastError(); } int doreuseaddress = TRUE; if (setsockopt(m_socket,SOL_SOCKET,SO_REUSEADDR,(char*) &doreuseaddress,sizeof(doreuseaddress)) == SOCKET_ERROR) { return WSAGetLastError(); } // Configure socket addresses memset(&m_sockAdr,0,sizeof(m_sockAdr)); m_sockAdr.sin_family = AF_INET; m_sockAdr.sin_port = htons(m_port); m_sockAdr.sin_addr.s_addr = m_interfaceAdr; //bind sockets if ( bind( m_socket, (SOCKADDR*) &m_sockAdr, sizeof(m_sockAdr) ) == SOCKET_ERROR ) { return WSAGetLastError(); } // join multicast struct ip_mreq_source imr; memset(&imr,0,sizeof(imr)); imr.imr_multiaddr.s_addr = m_multicastAdr; // address of multicastgroup imr.imr_sourceaddr.s_addr = 0; // sourceaddress (not used) imr.imr_interface.s_addr = m_interfaceAdr; // interface address /* first join multicast group, then registerer selected interface as * multicast sending interface */ if( setsockopt( m_socket ,IPPROTO_IP ,IP_ADD_MEMBERSHIP ,(char*) &imr , sizeof(imr)) == SOCKET_ERROR) { return SOCKET_ERROR; } else { if( setsockopt(m_socket ,IPPROTO_IP ,IP_MULTICAST_IF ,(CHAR*)&imr.imr_interface.s_addr ,sizeof(&imr.imr_interface.s_addr)) == SOCKET_ERROR ) { return SOCKET_ERROR; } } printf("receiving msgs...\n"); while(1) { // get inputbuffer from socket int sock_return = SOCKET_ERROR; sockaddr_in socketAddress; char buffer[1500]; int addressLength = sizeof(socketAddress); sock_return = recvfrom(m_socket, (char*) &buffer, 1500, 0, (SOCKADDR*)&socketAddress, &addressLength ); if( sock_return == SOCKET_ERROR) { int wsa_error = WSAGetLastError(); return wsa_error; } else { printf("got message!\n"); } } return 0; } Thanks four your help!

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  • structure of ethernet frame (tcp/udp) [closed]

    - by rtmrtm2
    How is an ethernet-frame structured. is it: |MAC | |_______________| | |IP | | |___________| | |TCP | | |_______| | |HTTP| |__________|____| or the other way around? so in words: is the mac wrapped around the ip wrapped around the tcp wrapped arround the http? can someone post an image of the specific 'wrapping'? thanks in advance. regards, rtmrtm2

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  • Bridged network on OS X only gets UDP broadcast traffic

    - by a paid nerd
    I've created a bridged network Mac OS X 10.8.5 using ifconfig and TUNTAP for OS X to bridge my wireless connection, en0, with a virtual interface, tap0, which I can use for guest VMs: $ sudo sysctl -w net.inet.ip.forwarding=1 $ sudo sysctl -w net.link.ether.inet.proxyall=1 $ sudo sysctl -w net.inet.ip.fw.enable=1 $ sudo ifconfig bridge0 create $ sudo ifconfig bridge0 addm en0 addm tap0 $ sudo ifconfig bridge0 up $ ifconfig en0: flags=8863<UP,BROADCAST,SMART,RUNNING,SIMPLEX,MULTICAST> mtu 1500 ether 28:cf:xx:xx:xx:xx inet6 xxxx::xxxx:xxxx:xxxx:xxxx%en0 prefixlen 64 scopeid 0x4 inet 192.168.100.64 netmask 0xffffff00 broadcast 192.168.100.1 media: autoselect status: active bridge0: flags=8863<UP,BROADCAST,SMART,RUNNING,SIMPLEX,MULTICAST> mtu 1500 ether ac:de:xx:xx:xx:xx Configuration: priority 0 hellotime 0 fwddelay 0 maxage 0 ipfilter disabled flags 0x2 member: en0 flags=3<LEARNING,DISCOVER> port 4 priority 0 path cost 0 member: tap0 flags=3<LEARNING,DISCOVER> port 8 priority 0 path cost 0 tap0: flags=8943<UP,BROADCAST,RUNNING,PROMISC,SIMPLEX,MULTICAST> mtu 1500 ether ca:3d:xx:xx:xx:xx open (pid 88244) However, if I tcpdump -i tap0, I only see broadcast traffic. Shouldn't I see a mirror of everything on en0? (192.168.100.33, the host doing the broadcasting, is another unrelate, noisy server on my LAN.) (I asked a similar question here and will probably close it.)

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  • how to get a udp response with netcat

    - by dihlofos
    Hi, everyone I'm trying to do something like: echo "request" | nc -u 1.1.1.1 9999 > response.txt I can see that response is comming from server (with tcpdump) after executing this line. However, my response.txt stays empty. Is there a way to get it? Thanks in advance

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  • Windows XP/Firewall -- Can't open UDP port 9100 for JetDirect sharing

    - by Barton Chittenden
    I'm setting up my Dad's laptop (Vista) to print to an HP Deskjet printer attached to a Windows XP Desktop box. The XP box is running Windows firewall. Laptop sits on 192.168.1.66, Desktop 192.168.1.65. I set up logging in the firewall, and determined that it is dropping inbound packets on TCP port 9100, used by JetDirect, the printing protocol. I set up an exception on TCP port 9100, allowing access to the local subnet. There's also an entry under the 'allow applications' tab which was marked for the exact printer model. I selected this as well, then tried to print again... I see more dropped packets from 192.168.1.66 to 192.168.1.65 for TCP port 9100. What do I try next?

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  • redirecting arbitrary tcp/udp in kvm

    - by jbfink
    I've got a server with KVM on it, and multiple guest VMs. I'd like a way to redirect traffic from the host server to the VMs. Like, say, forward all traffic on port 2222 on the host to 22 on a guest VM for ssh. This would have to be done either through virt-manager or libvirt XML config files -- I've found multiple references to doing it through qemu (like http://forums.fedoraforum.org/showthread.php?t=237969) but absolutely nothing that I can see related to either libvirt or virt-manager. Do you know how I can do this?

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