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• Openswan + xl2tpd connections time out after a while

  - by Halfgaar

  I have a non-NATed Openswan+xl2tpd server (Ubuntu 12.04), to which I connect with a Windows 8 behind NAT. The client loses its connection after a while of doing nothing (between 30 and 60 minutes, but I didn't time it). The client doesn't have enabled that it should kill inactive connections. Nor does it ever go into sleep mode. I also tried setting the kill-after-time to 24 hours, but that didn't help. The NAT router behind which the client located is Debian Linux, and its router is a Cisco which connects us directly to the data center where the server is. None of our other connections, like SSH, get dropped with inactivity (because of cheap routers). I did however try turning on the keepalives in /etc/ipsec.conf: config setup (...snip...) nat_traversal=yes force_keepalive=yes keep_alive=10 but that didn't help. As you can see in the config later, dead peer detection's action is clear. That would be a first suggestion to fix, but I need clear, because people will be connecting from everwhere but the kitchen sink. Besides, as I said, in the test setup I have now, I can't see any device killing its connection. (edit: 'restart' also has the same effect) These are of one time it happened: Jul 18 16:18:06 host xl2tpd[1918]: Maximum retries exceeded for tunnel 49070. Closing. Jul 18 16:18:06 host xl2tpd[1918]: Terminating pppd: sending TERM signal to pid 18359 Jul 18 16:18:06 host xl2tpd[1918]: Connection 4 closed to 89.188.x.y, port 1701 (Timeout) Jul 18 16:18:11 host xl2tpd[1918]: Unable to deliver closing message for tunnel 49070. Destroying anyway. and these on another: Jul 18 17:44:39 host xl2tpd[1918]: udp_xmit failed to 89.188.x.y:1701 with err=-1:Operation not permitted Jul 18 17:44:43 xl2tpd[1918]: last message repeated 4 times Jul 18 17:44:43 host xl2tpd[1918]: Maximum retries exceeded for tunnel 10918. Closing. Jul 18 17:44:43 host xl2tpd[1918]: udp_xmit failed to 89.188.x.y:1701 with err=-1:Operation not permitted Jul 18 17:44:43 host xl2tpd[1918]: Terminating pppd: sending TERM signal to pid 26338 Jul 18 17:44:43 host xl2tpd[1918]: Connection 6 closed to 89.188.x.y, port 1701 (Timeout) Jul 18 17:44:44 host xl2tpd[1918]: udp_xmit failed to 89.188.x.y:1701 with err=-1:Operation not permitted Jul 18 17:44:48 xl2tpd[1918]: last message repeated 3 times Jul 18 17:44:48 host xl2tpd[1918]: Unable to deliver closing message for tunnel 10918. Destroying anyway. Jul 18 17:44:59 host xl2tpd[1918]: Can not find tunnel 10918 (refhim=0) Jul 18 17:44:59 host xl2tpd[1918]: network_thread: unable to find call or tunnel to handle packet. call = 0, tunnel = 10918 Dumping. Jul 18 17:45:09 host xl2tpd[1918]: Can not find tunnel 10918 (refhim=0) Jul 18 17:45:09 host xl2tpd[1918]: network_thread: unable to find call or tunnel to handle packet. call = 0, tunnel = 10918 Dumping. Jul 18 17:45:19 host xl2tpd[1918]: Can not find tunnel 10918 (refhim=0) Jul 18 17:45:19 host xl2tpd[1918]: network_thread: unable to find call or tunnel to handle packet. call = 0, tunnel = 10918 Dumping. Jul 18 17:45:29 host xl2tpd[1918]: Can not find tunnel 10918 (refhim=0) Jul 18 17:45:29 host xl2tpd[1918]: network_thread: unable to find call or tunnel to handle packet. call = 0, tunnel = 10918 Dumping. Jul 18 17:45:39 host xl2tpd[1918]: Can not find tunnel 10918 (refhim=0) Jul 18 17:45:39 host xl2tpd[1918]: network_thread: unable to find call or tunnel to handle packet. call = 0, tunnel = 10918 Dumping. Jul 18 17:45:49 host xl2tpd[1918]: Can not find tunnel 10918 (refhim=0) Jul 18 17:45:49 host xl2tpd[1918]: network_thread: unable to find call or tunnel to handle packet. call = 0, tunnel = 10918 Dumping. Versions: Ubuntu 12.04 Openswan: 2.6.37-1 xl2tpd: 3.1+dfsg-1 kernel: 3.2.0-49-generic configs: /etc/ipsec.conf: version 2.0 # conforms to second version of ipsec.conf specification config setup nat_traversal=yes virtual_private=%v4:10.0.0.0/8,%v4:192.168.0.0/16,%v4:172.16.0.0/12,%v4:!10.152.2.0/24 oe=off protostack=netkey force_keepalive=yes keep_alive=10 conn L2TP-PSK-NAT rightsubnet=vhost:%priv also=L2TP-PSK-noNAT conn L2TP-PSK-noNAT authby=secret pfs=no auto=add keyingtries=2 rekey=no dpddelay=30 dpdtimeout=120 dpdaction=clear ikelifetime=8h keylife=1h type=transport left=%defaultroute leftprotoport=17/1701 right=%any rightprotoport=17/%any /etc/xl2tpd/xl2tpd.conf [global] ipsec saref = no [lns default] ip range = 10.152.2.2-10.152.2.254 local ip = 10.152.2.1 refuse chap = yes refuse pap = yes require authentication = yes ppp debug = no pppoptfile = /etc/ppp/options.xl2tpd length bit = yes /etc/ppp/options.xl2tpd: require-mschap-v2 refuse-mschap ms-dns 10.152.2.1 asyncmap 0 auth crtscts idle 1800 mtu 1200 mru 1200 lock hide-password local #debug name l2tpd proxyarp lcp-echo-interval 30 lcp-echo-failure 4

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• ssh through a bastion machine works on someone else's desktop but not my own

  - by Terrence Brannon

  I have to ssh into a bastion (jump) server in order to get to the final server. On the jump server, my .ssh/config says: Host * ForwardAgent yes My co-worker uses PuTTy and Pageant. When I use a putty shell to connect from his desktop to the final server as root via the jump server, it works fine. At my desk I cannot connect to the final server, only the jump server. However, if I go to his desk, and successfully log into the final server via the jump server, I can then go back to my desk and also do so.... but after a certain amount of time, my shells revert to the original behavior of not connecting to final server via jump server. The entire transcript of ssh -v -v -v final_server is here The relevant part to me is when the public key is offered but then it says 'we did not send a packet': debug1: Offering public key: /home/CORP/t.brannon/.ssh/id_dsa debug3: send_pubkey_test debug2: we sent a publickey packet, wait for reply debug1: Authentications that can continue: publickey,password debug2: we did not send a packet, disable method debug3: authmethod_lookup password

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• Printing to shared printers across VPN

  - by CYMR0

  I have a program that prints labels at five remote sites. Two sites, aren't working, but the rest are with an identical (as far as I can tell) setup. Using Wireshark, I have determined that the handshaking all goes well, but after the "Open Print File Response" the packet that is sent from the server, doesn't reach the client. But I'm a bit at a loss as to where I go from here. I know the port the packet was sent on (445) isn't being blocked, the RST packet gets sent on the same port and that gets there fine. It's also weird that the three out of five sites are working fine. This has been up and running for years without issue, all that we have changed is our connectivity (from DSL to bonded DSL). But this traffic is over a VPN - so it can't be the ISP interfering either can it? I'm totally stuck, and any help would be much appreciated. Thanks!

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• Is SYN flooding still a threat?

  - by Rob

  Well recently I've been reading about different Denial of Service methods. One method that kind of stuck out was SYN flooding. I'm a member of some not-so-nice forums, and someone was selling a python script that would DoS a server using SYN packets with a spoofed IP address. However, if you sent a SYN packet to a server, with a spoofed IP address, the target server would return the SYN/ACK packet to the host that was spoofed. In which case, wouldn't the spoofed host return an RST packet, thus negating the 75 second long-wait, and ultimately failing in its attempt to DoS the server?

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• Windows 7 cannot join samba domain

  - by Antonis Christofides

  I have a 3.5.6 samba server with a LDAP backend (both on Debian 6.0). I've been successfully adding Windows XP machines to the domain for years. I now try to add Windows 7. I have made the recommended registry changes, but I don't have any success so far. Here is what happens: 1. I go to computer name, select "Domain" instead of "Workgroup", type in the domain name, click OK. It asks me for the username and password of an account that can add computers to the domain; I enter them. After about 40 seconds, I get the following message: The following error occurred attempting to join the domain "ITIA": The specified computer account could not be found. Contact an administrator to verify the account is in the domain. If the account has been deleted unjoin, reboot, and rejoin the domain. Despite this, the samba server successfully creates the computer account. 2. Therefore, if I try again a second time, without deleting the already created computer account, I get a different error: The following error occurred attempting to join the domain "ITIA": The specified account already exists. (Note that until a while ago samba wasn't configured to automatically create computer accounts. What I did whenever I wanted an XP to join was to manually create it. When I first attempted to solve the Windows 7 join problem, I setup samba to do this automatically, as this is what most people do, as I understand, and I thought that it might be related. I haven't attempted to add an XP since I made this change, so I don't know if it works, but whether it works or not, the problem remains.) Update 1: Here are the relevant parts of smb.conf: [global] panic action = /usr/share/samba/panic-action %d workgroup = ITIA server string = Itia file server announce as = NT interfaces = 147.102.160.1 volume = %h passdb backend = ldapsam:ldap://ldap.itia.ntua.gr:389 ldap admin dn = uid=samba,ou=daemons,dc=itia,dc=ntua,dc=gr ldap ssl = off ldap suffix = dc=itia,dc=ntua,dc=gr ldap user suffix = ou=people ldap group suffix = ou=groups ldap machine suffix = ou=computers unix password sync = no add machine script = smbldap-useradd -w -i %u log file = /var/log/samba/samba-log.all log level = 3 max log size = 5000 syslog = 2 socket options = SO_KEEPALIVE TCP_NODELAY encrypt passwords = true password level = 1 security = user domain master = yes local master = no wins support = yes domain logons = yes idmap gid = 1000-2000 Update 2: The server has a single network interface eth1 (also an unused eth0 that shows up only in the kernel boot messages) and two ip addresses; the main, 147.102.160.1, and an additional one, 147.102.160.37, that comes up with "ip addr add 147.102.160.37/32 dev eth1" (used only for a web site that has a different certificate than other web sites served from the same machine). One of the problems I recently faced was that samba was using the latter IP address. I fixed that by adding the "interfaces = 147.102.160.1" statement in smb.conf. Now: acheloos:/etc/apache2# tcpdump host 147.102.160.40 and not port 5900 tcpdump: verbose output suppressed, use -v or -vv for full protocol decode listening on eth1, link-type EN10MB (Ethernet), capture size 65535 bytes 13:13:56.549048 IP lithaios.itia.civil.ntua.gr.netbios-dgm > 147.102.160.255.netbios-dgm: NBT UDP PACKET(138) 13:13:56.549056 ARP, Request who-has acheloos2.itia.civil.ntua.gr tell lithaios.itia.civil.ntua.gr, length 46 13:13:56.549091 ARP, Reply acheloos2.itia.civil.ntua.gr is-at 00:10:4b:b4:9e:59 (oui Unknown), length 28 13:13:56.549324 IP acheloos.itia.civil.ntua.gr.netbios-dgm > lithaios.itia.civil.ntua.gr.netbios-dgm: NBT UDP PACKET(138) 13:13:56.549608 IP lithaios.itia.civil.ntua.gr.netbios-dgm > acheloos2.itia.civil.ntua.gr.netbios-dgm: NBT UDP PACKET(138) 13:13:56.549741 IP acheloos.itia.civil.ntua.gr.netbios-dgm > lithaios.itia.civil.ntua.gr.netbios-dgm: NBT UDP PACKET(138) 13:13:56.550364 IP lithaios.itia.civil.ntua.gr.netbios-dgm > acheloos.itia.civil.ntua.gr.netbios-dgm: NBT UDP PACKET(138) 13:13:56.550468 IP acheloos.itia.civil.ntua.gr.netbios-dgm > lithaios.itia.civil.ntua.gr.netbios-dgm: NBT UDP PACKET(138) (acheloos2 is the second IP address, 147.102.160.37). The above dump occurs when I click "OK" (to join the domain), until it asks me for the username and password of a user that can join the domain. I don't know why the client is contacting the second IP address. I tried temporarily deactivating it, but I still had some related ARP traffic (though I think not IP traffic).

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• switch duplicates packets and forward in two route

  - by sami

  there is a network including a router, two hosts and a switch which connects hosts to router. i have a virtual machine on my system. the network adapter is set to act as bridge. so the virtual machine and real OS are my 2 hosts on different LAN. they use one network card and are connected to a switch. when each of host send a packet to the other one, the switch duplicate the packet and forward it to both router and the other host. how can I solve the duplicate packet problem? Thanks.

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• Is there a suitable chain for iptables when eth is in Promisc mode?

  - by user1495181

  I have a fron-end machine. Machine have2 eth cards. I want to use netfilter queue to do some checks on the packets. I set eth like this: ifconfig eth0 0.0.0.0 promisc up ifconfig eth1 0.0.0.0 promisc up I want to have an iptable rule like this(only example): iptables -A INPUT -i eth0 -j LOG --log-prefix " eth0 packet " but the packet is no passed through the iptables ,because it dosnt target to this MAC. Promisc mode didnt help. I saw that there is a way to add iptables chain for PROMISC, but need compilation... Is there any simplier way to have iptables rule when packet is not target to this eth. Currently i bypass this by creating a bridge between 2 eth and put rule on the FORWARD, but i done want to create bridge.

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• openfire cannot subscribe gmail user

  - by cometta

  i trying to add gmail user with my local openfire, but get error below. I think something wrong with dns srv. can anyone suggest how to troubleshoot? </error> </presence> at org.jivesoftware.openfire.spi.RoutingTableImpl.routePacket(RoutingTableImpl.java:217) at org.jivesoftware.openfire.server.OutgoingSessionPromise$PacketsProcessor.returnErrorToSender(OutgoingSessionPromise.java:285) at org.jivesoftware.openfire.server.OutgoingSessionPromise$PacketsProcessor.run(OutgoingSessionPromise.java:204) at java.util.concurrent.ThreadPoolExecutor$Worker.runTask(ThreadPoolExecutor.java:651) at java.util.concurrent.ThreadPoolExecutor$Worker.run(ThreadPoolExecutor.java:676) at java.lang.Thread.run(Thread.java:613) 2010.04.25 23:30:57 Error returning error to sender. Original packet: <presence id="lBI4K-24" to="[email protected]" type="subscribe" from="[email protected]"/> org.jivesoftware.openfire.PacketException: Cannot route packet of type IQ or Presence to bare JID: <presence id="lBI4K-24" to="[email protected]" from="[email protected]" type="error"> <error code="404" type="cancel"> <remote-server-not-found xmlns="urn:ietf:params:xml:ns:xmpp-stanzas"/> </error> </presence> at org.jivesoftware.openfire.spi.RoutingTableImpl.routePacket(RoutingTableImpl.java:217) at org.jivesoftware.openfire.server.OutgoingSessionPromise$PacketsProcessor.returnErrorToSender(OutgoingSessionPromise.java:285) at org.jivesoftware.openfire.server.OutgoingSessionPromise$PacketsProcessor.run(OutgoingSessionPromise.java:219) at java.util.concurrent.ThreadPoolExecutor$Worker.runTask(ThreadPoolExecutor.java:651) at java.util.concurrent.ThreadPoolExecutor$Worker.run(ThreadPoolExecutor.java:676) at java.lang.Thread.run(Thread.java:613) 2010.04.25 23:31:56 Error returning error to sender. Original packet: <presence id="gmEsS-26" to="[email protected]" type="subscribe" from="[email protected]"/> org.jivesoftware.openfire.PacketException: Cannot route packet of type IQ or Presence to bare JID: <presence id="gmEsS-26" to="[email protected]" from="[email protected]" type="error"> <error code="404" type="cancel"> <remote-server-not-found xmlns="urn:ietf:params:xml:ns:xmpp-stanzas"/> </error> </presence> at org.jivesoftware.openfire.spi.RoutingTableImpl.routePacket(RoutingTableImpl.java:217) at org.jivesoftware.openfire.server.OutgoingSessionPromise$PacketsProcessor.returnErrorToSender(OutgoingSessionPromise.java:285) at org.jivesoftware.openfire.server.OutgoingSessionPromise$PacketsProcessor.run(OutgoingSessionPromise.java:219) at java.util.concurrent.ThreadPoolExecutor$Worker.runTask(ThreadPoolExecutor.java:651) at java.util.concurrent.ThreadPoolExecutor$Worker.run(ThreadPoolExecutor.java:676) at java.lang.Thread.run(Thread.java:613) 2010.04.25 23:31:56 Error returning error to sender. Original packet: <presence id="gmEsS-27" to="[email protected]" type="subscribe" from="[email protected]"/> org.jivesoftware.openfire.PacketException: Cannot route packet of type IQ or Presence to bare JID: <presence id="gmEsS-27" to="[email protected]" from="[email protected]" type="error"> <error code="404" type="cancel"> <remote-server-not-found xmlns="urn:ietf:params:xml:ns:xmpp-stanzas"/> </error> </presence> at org.jivesoftware.openfire.spi.RoutingTableImpl.routePacket(RoutingTableImpl.java:217) at org.jivesoftware.openfire.server.OutgoingSessionPromise$PacketsProcessor.returnErrorToSender(OutgoingSessionPromise.java:285) at org.jivesoftware.openfire.server.OutgoingSessionPromise$PacketsProcessor.run(OutgoingSessionPromise.java:204) at java.util.concurrent.ThreadPoolExecutor$Worker.runTask(ThreadPoolExecutor.java:651) at java.util.concurrent.ThreadPoolExecutor$Worker.run(ThreadPoolExecutor.java:676) at java.lang.Thread.run(Thread.java:613)

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• Traffic estimation for a multiplayer flash game

  - by Steve Addington

  hey, i want to know if my rough traffic estimations are right, it would be for a pretty simple realtime flashgame in the style of haxball (but not as a soccer game) heres a video of it http://www.youtube.com/watch?v=z_xBdFg1RcI So here comes my estimation, i dont know if they are realistic! i hope someone can help me. consider the packet attached as a typical one sent every 200ms, its 148bytes + 64 bytes of header will make around a 200bytes packet. The server will receive 200bytes x 6 players x 5 times a sec=6000bytes/s=5.85Kbytes/s=46.9kbit/s plus he has to send all back to the players, so at this point are 94Kbit/s.The server received all the information, perform the definitive calculation and send the new position to all players, in a bigger packet of around 900bytes that have to be delivered to the others 6, which makes 900bytes x 6 players x 5 times a sec=27000bytes/s=26Kbytes/s=210kbit/s. overall that would be 26kbyte per second. thats like 130mb traffic per hour for a 6player room. but somehow i think the numbers are too high? that would be really much traffic for such a simple game. did i calculate something wrong?

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• Client side latency when using prediction

  - by Tips48

  I've implemented Client-Side prediction into my game, where when input is received by the client, it first sends it to the server and then acts upon it just as the server will, to reduce the appearance of lag. The problem is, the server is authoritative, so when the server sends back the position of the Entity to the client, it undo's the effect of the interpolation and creates a rubber-banding effect. For example: Client sends input to server - Client reacts on input - Server receives and reacts on input - Server sends back response - Client reaction is undone due to latency between server and client To solve this, I've decided to store the game state and input every tick in the client, and then when I receive a packet from the server, get the game state from when the packet was sent and simulate the game up to the current point. My questions: Won't this cause lag? If I'm receiving 20/30 EntityPositionPackets a second, that means I have to run 20-30 simulations of the game state. How do I sync the client and server tick? Currently, I'm sending the milli-second the packet was sent by the server, but I think it's adding too much complexity instead of just sending the tick. The problem with converting it to sending the tick is that I have no guarantee that the client and server are ticking at the same rate, for example if the client is an old-end PC.

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• Ubuntu 12.04 connected to wireless network but internet not working

  - by A.J.

  I can connect to my house's wireless network just fine, but when I'm connected I can't browse the web. Firefox starts connecting to a site and then just poops out. This doesn't happen on my roommates' computers (running Windows) or on our 3DSes, so I know it's just my laptop. I already tried sudo dhclient -r sudo dhclient sudo ifconfig eth0 down sudo ifconfig eth0 up Results of a few commands I was asked to run in comments: ping -c 2 4.2.2.2 PING 4.2.2.2 (4.2.2.2) 56(84) bytes of data. ^C --- 4.2.2.2 ping statistics --- 2 packets transmitted, 0 received, 100% packet loss, time 1007ms ping -c 2 google.com PING google.com (173.194.33.38) 56(84) bytes of data. --- google.com ping statistics --- 2 packets transmitted, 0 received, 100% packet loss, time 1006ms nm-tool NetworkManager Tool State: connected (global) - Device: eth0 ----------------------------------------------------------------- Type: Wired Driver: atl1c State: unavailable Default: no HW Address: 88:AE:1D:6B:4E:E7 Capabilities: Carrier Detect: yes Speed: 100 Mb/s Wired Properties Carrier: off - Device: wlan0 [JUSTICE] ----------------------------------------------------- Type: 802.11 WiFi Driver: ath9k State: connected Default: yes HW Address: 1C:65:9D:65:C6:31 Capabilities: Speed: 1 Mb/s Wireless Properties WEP Encryption: yes WPA Encryption: yes WPA2 Encryption: yes Wireless Access Points (* = current AP) HOME-9B18: Infra, 00:26:F3:53:9B:18, Freq 2412 MHz, Rate 54 Mb/s, Strength 34 WPA WPA2 cougdad48 Network: Infra, 60:33:4B:E4:C4:5D, Freq 2437 MHz, Rate 54 Mb/s, Strength 22 WPA2 cougdad48 Guest Network: Infra, 66:33:4B:E4:C4:5D, Freq 2437 MHz, Rate 54 Mb/s, Strength 20 WPA2 belkin.ade: Infra, 94:44:52:FF:8A:DE, Freq 2457 MHz, Rate 54 Mb/s, Strength 20 WPA WPA2 *JUSTICE: Infra, 00:24:01:7B:9F:7E, Freq 2462 MHz, Rate 54 Mb/s, Strength 88 WEP CenturyLink: Infra, B2:B2:DC:8E:E2:58, Freq 2462 MHz, Rate 54 Mb/s, Strength 17 WPA WPA2 IPv4 Settings: Address: 192.168.0.11 Prefix: 24 (255.255.255.0) Gateway: 192.168.0.1 DNS: 192.168.0.1 (JUSTICE is my home's network.) ping -c 2 198.168.0.1 PING 198.168.0.1 (198.168.0.1) 56(84) bytes of data. --- 198.168.0.1 ping statistics --- 2 packets transmitted, 0 received, 100% packet loss, time 1007ms

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• recent unreliable wireless connection on 10.04 and 10.10

  - by gabkdlly

  Recently, my internet connection over wireless has become unreliable, on both a Dell laptop running Ubuntu 10.04 as well as my Desktop running Ubuntu 10.10 . The problem does not seem to occur on a laptop running Windows Vista. The problem does not seem to occur on my Openmoko Freerunner ( running Android 1.5 ), though I hardly ever use this device to connect over WLAN, so the problem may have just slipped by. This problem does not seem to appear when I boot into Ubuntu 9.10 from a live CD ( more precisely, I was able to ping fu-berlin.de for an hour without any packet loss ). Under Ubuntu 10.10, I am experiencing about 33% packet loss. On my main Ubuntu Desktop, I have tried the following wireless devices: a Longshine PCI card ( an old device with an RTL8180L chip ) a D-Link DWL-510 PCI card ( this device threw warnings in dmesg ) a USB device from MSI ( US54EX ). Usually my wireless network shows up in the network manager with a normal signal strength, even when the connection speed is slow ( which happens often ) or the connection gets reset ( asking me to click connect to re-authenticate my wireless connection ). I have observed this problem with a Netgear KWGR614 Router ( with the manufacturers firmware ), as well as with a TP-LINK TL-WR741ND router running OpenWrt. Taking a look at my routers logs, I find many instances of the following line: Tuesday,04 Jan 2011 03:53:01 [TCP SYN Flood][Deny access policy matched, dropping packet] I know that the Netgear router is susceptible to denial of service attacks, as I have previously been able to disrupt its operation by putting an nmap scan into a while loop. I use WEP on the Netgear router and WPA on the TP-LINK to encrypt the wireless connections. Is it possible that someone is jamming my signal ?

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• Should I allow sending complete structures when using PUT for updates in a REST API or not?

  - by dafmetal

  I am designing a REST API and I wonder what the recommended way to handle updates to resources would be. More specifically, I would allow updates through a PUT on the resource, but what should I allow in the body of the PUT request? Always the complete structure of the resource? Always the subpart (that changed) of the structure of the resource? A combination of both? For example, take the resource http://example.org/api/v1/dogs/packs/p1. A GET on this resource would give the following: Request: GET http://example.org/api/v1/dogs/packs/p1 Accept: application/xml Response: <pack> <owner>David</owner> <dogs> <dog> <name>Woofer</name> <breed>Basset Hound</breed> </dog> <dog> <name>Mr. Bones</name> <breed>Basset Hound</breed> </dog> </dogs> </pack> Suppose I want to add a dog (Sniffers the Basset Hound) to the pack, would I support either: Request: PUT http://example.org/api/v1/dogs/packs/p1 <dog> <name>Sniffers</name> <breed>Basset Hound</breed> </dog> Response: HTTP/1.1 200 OK or Request: PUT http://example.org/api/v1/dogs/packs/p1 <pack> <owner>David</owner> <dogs> <dog> <name>Woofer</name> <breed>Basset Hound</breed> </dog> <dog> <name>Mr. Bones</name> <breed>Basset Hound</breed> </dog> <dog> <name>Sniffers</name> <breed>Basset Hound</breed> </dog> </dogs> </pack> Response: HTTP/1.1 200 OK or both? If supporting updates through subsections of the structure is recommended, how would I handle deletes (such as when a dog dies)? Through query parameters?

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• How do I make my multicast program work between computers on different networks?

  - by George

  I made a little chat applet using multicast. It works fine between computers on the same network, but fails if the computers are on different networks. Why is this? import java.io.*; import java.net.*; import java.awt.*; import java.awt.event.*; import javax.swing.*; public class ClientA extends JApplet implements ActionListener, Runnable { JTextField tf; JTextArea ta; MulticastSocket socket; InetAddress group; String name=""; public void start() { try { socket = new MulticastSocket(7777); group = InetAddress.getByName("233.0.0.1"); socket.joinGroup(group); socket.setTimeToLive(255); Thread th = new Thread(this); th.start(); name =JOptionPane.showInputDialog(null,"Please enter your name.","What is your name?",JOptionPane.PLAIN_MESSAGE); tf.grabFocus(); }catch(Exception e) {e.printStackTrace();} } public void init() { JPanel p = new JPanel(new BorderLayout()); ta = new JTextArea(); ta.setEditable(false); ta.setLineWrap(true); JScrollPane sp = new JScrollPane(ta); p.add(sp,BorderLayout.CENTER); JPanel p2 = new JPanel(); tf = new JTextField(30); tf.addActionListener(this); p2.add(tf); JButton b = new JButton("Send"); b.addActionListener(this); p2.add(b); p.add(p2,BorderLayout.SOUTH); add(p); } public void actionPerformed(ActionEvent ae) { String message = name+":"+tf.getText(); tf.setText(""); tf.grabFocus(); byte[] buf = message.getBytes(); DatagramPacket packet = new DatagramPacket(buf,buf.length, group,7777); try { socket.send(packet); } catch(Exception e) {} } public void run() { while(true) { byte[] buf = new byte[256]; String received = ""; DatagramPacket packet = new DatagramPacket(buf, buf.length); try { socket.receive(packet); received = new String(packet.getData()).trim(); } catch(Exception e) {} ta.append(received +"\n"); ta.setCaretPosition(ta.getDocument().getLength()); } } }

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• Feedback on Optimizing C# NET Code Block

  - by Brett Powell

  I just spent quite a few hours reading up on TCP servers and my desired protocol I was trying to implement, and finally got everything working great. I noticed the code looks like absolute bollocks (is the the correct usage? Im not a brit) and would like some feedback on optimizing it, mostly for reuse and readability. The packet formats are always int, int, int, string, string. try { BinaryReader reader = new BinaryReader(clientStream); int packetsize = reader.ReadInt32(); int requestid = reader.ReadInt32(); int serverdata = reader.ReadInt32(); Console.WriteLine("Packet Size: {0} RequestID: {1} ServerData: {2}", packetsize, requestid, serverdata); List<byte> str = new List<byte>(); byte nextByte = reader.ReadByte(); while (nextByte != 0) { str.Add(nextByte); nextByte = reader.ReadByte(); } // Password Sent to be Authenticated string string1 = Encoding.UTF8.GetString(str.ToArray()); str.Clear(); nextByte = reader.ReadByte(); while (nextByte != 0) { str.Add(nextByte); nextByte = reader.ReadByte(); } // NULL string string string2 = Encoding.UTF8.GetString(str.ToArray()); Console.WriteLine("String1: {0} String2: {1}", string1, string2); // Reply to Authentication Request MemoryStream stream = new MemoryStream(); BinaryWriter writer = new BinaryWriter(stream); writer.Write((int)(1)); // Packet Size writer.Write((int)(requestid)); // Mirror RequestID if Authenticated, -1 if Failed byte[] buffer = stream.ToArray(); clientStream.Write(buffer, 0, buffer.Length); clientStream.Flush(); } I am going to be dealing with other packet types as well that are formatted the same (int/int/int/str/str), but different values. I could probably create a packet class, but this is a bit outside my scope of knowledge for how to apply it to this scenario. If it makes any difference, this is the Protocol I am implementing. http://developer.valvesoftware.com/wiki/Source_RCON_Protocol

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• XNA Multiplayer Games and Networking

  - by JoshReuben

  ·        XNA communication must by default be lightweight – if you are syncing game state between players from the Game.Update method, you must minimize traffic. That game loop may be firing 60 times a second and player 5 needs to know if his tank has collided with any player 3 and the angle of that gun turret. There are no WCF ServiceContract / DataContract niceties here, but at the same time the XNA networking stack simplifies the details. The payload must be simplistic - just an ordered set of numbers that you would map to meaningful enum values upon deserialization.   Overview ·        XNA allows you to create and join multiplayer game sessions, to manage game state across clients, and to interact with the friends list ·        Dependency on Gamer Services - to receive notifications such as sign-in status changes and game invitations ·        two types of online multiplayer games: system link game sessions (LAN) and LIVE sessions (WAN). ·        Minimum dev requirements: 1 Xbox 360 console + Creators Club membership to test network code - run 1 instance of game on Xbox 360, and 1 on a Windows-based computer   Network Sessions ·        A network session is made up of players in a game + up to 8 arbitrary integer properties describing the session ·        create custom enums – (e.g. GameMode, SkillLevel) as keys in NetworkSessionProperties collection ·        Player state: lobby, in-play   Session Types ·        local session - for split-screen gaming - requires no network traffic. ·        system link session - connects multiple gaming machines over a local subnet. ·        Xbox LIVE multiplayer session - occurs on the Internet. Ranked or unranked   Session Updates ·        NetworkSession class Update method - must be called once per frame. ·        performs the following actions: o   Sends the network packets. o   Changes the session state. o   Raises the managed events for any significant state changes. o   Returns the incoming packet data. ·        synchronize the session à packet-received and state-change events à no threading issues   Session Config ·        Session host - gaming machine that creates the session. XNA handles host migration ·        NetworkSession properties: AllowJoinInProgress , AllowHostMigration ·        NetworkSession groups: AllGamers, LocalGamers, RemoteGamers   Subscribe to NetworkSession events ·        GamerJoined ·        GamerLeft ·        GameStarted ·        GameEnded – use to return to lobby ·        SessionEnded – use to return to title screen   Create a Session session = NetworkSession.Create(         NetworkSessionType.SystemLink,         maximumLocalPlayers,         maximumGamers,         privateGamerSlots,         sessionProperties );   Start a Session if (session.IsHost) {     if (session.IsEveryoneReady)     {        session.StartGame();        foreach (var gamer in SignedInGamer.SignedInGamers)        {             gamer.Presence.PresenceMode =                 GamerPresenceMode.InCombat;   Find a Network Session AvailableNetworkSessionCollection availableSessions = NetworkSession.Find(     NetworkSessionType.SystemLink,       maximumLocalPlayers,     networkSessionProperties); availableSessions.AllowJoinInProgress = true;   Join a Network Session NetworkSession session = NetworkSession.Join(     availableSessions[selectedSessionIndex]);   Sending Network Data var packetWriter = new PacketWriter(); foreach (LocalNetworkGamer gamer in session.LocalGamers) {     // Get the tank associated with this player.     Tank myTank = gamer.Tag as Tank;     // Write the data.     packetWriter.Write(myTank.Position);     packetWriter.Write(myTank.TankRotation);     packetWriter.Write(myTank.TurretRotation);     packetWriter.Write(myTank.IsFiring);     packetWriter.Write(myTank.Health);       // Send it to everyone.     gamer.SendData(packetWriter, SendDataOptions.None);     }   Receiving Network Data foreach (LocalNetworkGamer gamer in session.LocalGamers) {     // Keep reading while packets are available.     while (gamer.IsDataAvailable)     {         NetworkGamer sender;          // Read a single packet.         gamer.ReceiveData(packetReader, out sender);          if (!sender.IsLocal)         {             // Get the tank associated with this packet.             Tank remoteTank = sender.Tag as Tank;              // Read the data and apply it to the tank.             remoteTank.Position = packetReader.ReadVector2();             …   End a Session if (session.AllGamers.Count == 1)         {             session.EndGame();             session.Update();         }   Performance •        Aim to minimize payload, reliable in order messages •        Send Data Options: o   Unreliable, out of order -(SendDataOptions.None) o   Unreliable, in order (SendDataOptions.InOrder) o   Reliable, out of order (SendDataOptions.Reliable) o   Reliable, in order (SendDataOptions.ReliableInOrder) o   Chat data (SendDataOptions.Chat) •        Simulate: NetworkSession.SimulatedLatency , NetworkSession.SimulatedPacketLoss •        Voice support – NetworkGamer properties: HasVoice ,IsTalking , IsMutedByLocalUser

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• Using WKA in Large Coherence Clusters (Disabling Multicast)

  - by jpurdy

  Disabling hardware multicast (by configuring well-known addresses aka WKA) will place significant stress on the network. For messages that must be sent to multiple servers, rather than having a server send a single packet to the switch and having the switch broadcast that packet to the rest of the cluster, the server must send a packet to each of the other servers. While hardware varies significantly, consider that a server with a single gigabit connection can send at most ~70,000 packets per second. To continue with some concrete numbers, in a cluster with 500 members, that means that each server can send at most 140 cluster-wide messages per second. And if there are 10 cluster members on each physical machine, that number shrinks to 14 cluster-wide messages per second (or with only mild hyperbole, roughly zero). It is also important to keep in mind that network I/O is not only expensive in terms of the network itself, but also the consumption of CPU required to send (or receive) a message (due to things like copying the packet bytes, processing a interrupt, etc). Fortunately, Coherence is designed to rely primarily on point-to-point messages, but there are some features that are inherently one-to-many: Announcing the arrival or departure of a member Updating partition assignment maps across the cluster Creating or destroying a NamedCache Invalidating a cache entry from a large number of client-side near caches Distributing a filter-based request across the full set of cache servers (e.g. queries, aggregators and entry processors) Invoking clear() on a NamedCache The first few of these are operations that are primarily routed through a single senior member, and also occur infrequently, so they usually are not a primary consideration. There are cases, however, where the load from introducing new members can be substantial (to the point of destabilizing the cluster). Consider the case where cluster in the first paragraph grows from 500 members to 1000 members (holding the number of physical machines constant). During this period, there will be 500 new member introductions, each of which may consist of several cluster-wide operations (for the cluster membership itself as well as the partitioned cache services, replicated cache services, invocation services, management services, etc). Note that all of these introductions will route through that one senior member, which is sharing its network bandwidth with several other members (which will be communicating to a lesser degree with other members throughout this process). While each service may have a distinct senior member, there's a good chance during initial startup that a single member will be the senior for all services (if those services start on the senior before the second member joins the cluster). It's obvious that this could cause CPU and/or network starvation. In the current release of Coherence (3.7.1.3 as of this writing), the pure unicast code path also has less sophisticated flow-control for cluster-wide messages (compared to the multicast-enabled code path), which may also result in significant heap consumption on the senior member's JVM (from the message backlog). This is almost never a problem in practice, but with sufficient CPU or network starvation, it could become critical. For the non-operational concerns (near caches, queries, etc), the application itself will determine how much load is placed on the cluster. Applications intended for deployment in a pure unicast environment should be careful to avoid excessive dependence on these features. Even in an environment with multicast support, these operations may scale poorly since even with a constant request rate, the underlying workload will increase at roughly the same rate as the underlying resources are added. Unless there is an infrastructural requirement to the contrary, multicast should be enabled. If it can't be enabled, care should be taken to ensure the added overhead doesn't lead to performance or stability issues. This is particularly crucial in large clusters.

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• linux routing bug?

  - by Balázs Pozsár

  I have been struggling with this not easily reproducible issue since a while. I am using linux kernel v3.1.0, and sometimes routing to a few IP addresses does not work. What seems to happen is that instead of sending the packet to the gateway, the kernel treats the destination address as local, and tries to gets its MAC address via ARP. For example, now my current IP address is 172.16.1.104/24, the gateway is 172.16.1.254: # ifconfig eth0 eth0 Link encap:Ethernet HWaddr 00:1B:63:97:FC:DC inet addr:172.16.1.104 Bcast:172.16.1.255 Mask:255.255.255.0 UP BROADCAST RUNNING MULTICAST MTU:1500 Metric:1 RX packets:230772 errors:0 dropped:0 overruns:0 frame:0 TX packets:171013 errors:0 dropped:0 overruns:0 carrier:0 collisions:0 txqueuelen:1000 RX bytes:191879370 (182.9 Mb) TX bytes:47173253 (44.9 Mb) Interrupt:17 # route -n Kernel IP routing table Destination Gateway Genmask Flags Metric Ref Use Iface 0.0.0.0 172.16.1.254 0.0.0.0 UG 0 0 0 eth0 172.16.1.0 0.0.0.0 255.255.255.0 U 1 0 0 eth0 I can ping a few addresses, but not 172.16.0.59: # ping -c1 172.16.1.254 PING 172.16.1.254 (172.16.1.254) 56(84) bytes of data. 64 bytes from 172.16.1.254: icmp_seq=1 ttl=64 time=0.383 ms --- 172.16.1.254 ping statistics --- 1 packets transmitted, 1 received, 0% packet loss, time 0ms rtt min/avg/max/mdev = 0.383/0.383/0.383/0.000 ms root@pozsybook:~# ping -c1 172.16.0.1 PING 172.16.0.1 (172.16.0.1) 56(84) bytes of data. 64 bytes from 172.16.0.1: icmp_seq=1 ttl=63 time=5.54 ms --- 172.16.0.1 ping statistics --- 1 packets transmitted, 1 received, 0% packet loss, time 0ms rtt min/avg/max/mdev = 5.545/5.545/5.545/0.000 ms root@pozsybook:~# ping -c1 172.16.0.2 PING 172.16.0.2 (172.16.0.2) 56(84) bytes of data. 64 bytes from 172.16.0.2: icmp_seq=1 ttl=62 time=7.92 ms --- 172.16.0.2 ping statistics --- 1 packets transmitted, 1 received, 0% packet loss, time 0ms rtt min/avg/max/mdev = 7.925/7.925/7.925/0.000 ms root@pozsybook:~# ping -c1 172.16.0.59 PING 172.16.0.59 (172.16.0.59) 56(84) bytes of data. From 172.16.1.104 icmp_seq=1 Destination Host Unreachable --- 172.16.0.59 ping statistics --- 1 packets transmitted, 0 received, +1 errors, 100% packet loss, time 0ms When trying to ping 172.16.0.59, I can see in tcpdump that an ARP req was sent: # tcpdump -n -i eth0|grep ARP tcpdump: verbose output suppressed, use -v or -vv for full protocol decode listening on eth0, link-type EN10MB (Ethernet), capture size 96 bytes 15:25:16.671217 ARP, Request who-has 172.16.0.59 tell 172.16.1.104, length 28 and /proc/net/arp has an incomplete entry for 172.16.0.59: # grep 172.16.0.59 /proc/net/arp 172.16.0.59 0x1 0x0 00:00:00:00:00:00 * eth0 Please note, that 172.16.0.59 is accessible from this LAN from other computers. Does anyone have any idea of what's going on? Thanks. update: replies to the comments below: there are no interfaces besides eth0 and lo the ARP req cannot be seen on the other end, but that's how it should work. the main problem is that an ARP req should not even be sent at the first place the problem persist even if I add an explicit route with the command "route add -host 172.16.0.59 gw 172.16.1.254 dev eth0"

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• Cablemodem (SBG6580) firewall denying some outbound traffic? Why? Not configured [migrated]

  - by lairdb

  I finally got around to turning the syslog on for my cablemodem (Motorola Surfboard SBG6580) and I'm seeing about the expected amount of inbound attackage being blocked... 2014-05-30 21:59:02 Local0.Alert 192.168.111.1 May 31 04:58:56 2014 SYSLOG[0]: [Host 192.168.111.1] UDP 12.230.209.198,4500 --> 66.27.xx.xx,61459 DENY:Firewall interface [IP Fragmented Packet] attack 2014-05-30 21:59:02 Local0.Alert 192.168.111.1 May 31 04:58:56 2014 SYSLOG[0]: [Host 192.168.111.1] TCP 17.172.232.109,5223 --> 66.27.xx.xx,53814 DENY:Firewall interface access request 2014-05-30 21:59:02 Local0.Alert 192.168.111.1 May 31 04:58:57 2014 SYSLOG[0]: [Host 192.168.111.1] UDP 12.230.209.198,443 --> 66.27.xx.xx,53385 DENY: Firewall interface [IP Fragmented Packet] attack 2014-05-30 21:59:02 Local0.Alert 192.168.111.1 May 31 04:58:57 2014 SYSLOG[0]: [Host 192.168.111.1] UDP 12.230.209.198,4500 --> 66.27.xx.xx,61459 DENY:Firewall interface [IP Fragmented Packet] attack 2014-05-30 21:59:10 Local0.Alert 192.168.111.1 May 31 04:59:04 2014 SYSLOG[0]: [Host 192.168.111.1] UDP 12.230.209.198,443 --> 66.27.xx.xx,59960 DENY: Firewall interface [IP Fragmented Packet] attack 2014-05-30 21:59:10 Local0.Alert 192.168.111.1 May 31 04:59:04 2014 SYSLOG[0]: [Host 192.168.111.1] UDP 12.230.209.198,4500 --> 66.27.xx.xx,61459 DENY:Firewall interface [IP Fragmented Packet] attack ...and that's great. (Sad, but great.) But I'm also seeing a HUGE amount of what appears to be denied outbound connectivity: 2014-05-30 16:30:10 Local0.Alert 192.168.111.1 May 30 23:30:04 2014 SYSLOG[0]: [Host 192.168.111.1] TCP 192.168.111.100,58969 --> 38.81.66.127,443 DENY: Inbound or outbound access request 2014-05-30 16:30:10 Local0.Alert 192.168.111.1 May 30 23:30:04 2014 SYSLOG[0]: [Host 192.168.111.1] TCP 192.168.111.100,58969 --> 38.81.66.127,443 DENY: Inbound or outbound access request 2014-05-30 16:30:10 Local0.Alert 192.168.111.1 May 30 23:30:04 2014 SYSLOG[0]: [Host 192.168.111.1] TCP 192.168.111.100,58965 --> 162.222.41.13,443 DENY: Inbound or outbound access request 2014-05-30 16:30:10 Local0.Alert 192.168.111.1 May 30 23:30:04 2014 SYSLOG[0]: [Host 192.168.111.1] TCP 192.168.111.100,58965 --> 162.222.41.13,443 DENY: Inbound or outbound access request 2014-05-30 16:30:10 Local0.Alert 192.168.111.1 May 30 23:30:04 2014 SYSLOG[0]: [Host 192.168.111.1] TCP 192.168.111.100,58964 --> 38.81.66.179,443 DENY: Inbound or outbound access request 2014-05-30 16:30:10 Local0.Alert 192.168.111.1 May 30 23:30:04 2014 SYSLOG[0]: [Host 192.168.111.1] TCP 192.168.111.100,58964 --> 38.81.66.179,443 DENY: Inbound or outbound access request ...and Spot checking suggests that it's all legitimate traffic (Opening connections to CrashPlan, etc.), I have no restrictions configured in the modem; I don't see why it should be blocking anything. Am I misreading the log entry, and it's not actually being denied? (Seems unlikely.) Is the ISP (TWC) pushing deny tables that are not exposed in the UI? (Tinfoil hat too tight.) I'm confused. (The good news, such as it is, is that AFAIK I'm not experiencing any actual issues... but maybe I am; tough to tell.) Thanks.

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• Need help modifying C++ application to accept continuous piped input in Linux

  - by GreeenGuru

  The goal is to mine packet headers for URLs visited using tcpdump. So far, I can save a packet header to a file using: tcpdump "dst port 80 and tcp[13] & 0x08 = 8" -A -s 300 | tee -a ./Desktop/packets.txt And I've written a program to parse through the header and extract the URL when given the following command: cat ~/Desktop/packets.txt | ./packet-parser.exe But what I want to be able to do is pipe tcpdump directly into my program, which will then log the data: tcpdump "dst port 80 and tcp[13] & 0x08 = 8" -A -s 300 | ./packet-parser.exe Here is the script as it is. The question is: how do I need to change it to support continuous input from tcpdump? #include <boost/regex.hpp> #include <fstream> #include <cstdio> // Needed to define ios::app #include <string> #include <iostream> int main() { // Make sure to open the file in append mode std::ofstream file_out("/var/local/GreeenLogger/url.log", std::ios::app); if (not file_out) std::perror("/var/local/GreeenLogger/url.log"); else { std::string text; // Get multiple lines of input -- raw std::getline(std::cin, text, '\0'); const boost::regex pattern("GET (\\S+) HTTP.*?[\\r\\n]+Host: (\\S+)"); boost::smatch match_object; bool match = boost::regex_search(text, match_object, pattern); if(match) { std::string output; output = match_object[2] + match_object[1]; file_out << output << '\n'; std::cout << output << std::endl; } file_out.close(); } } Thank you ahead of time for the help!

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• Converting a size_t into an integer (c++)

  - by JeanOTF

  Hello, I've been trying to make a for loop that will iterate based off of the length of a network packet. In the API there exists a variable (size_t) by event.packet-dataLength. I want to iterate from 0 to event.packet-dataLength - 7 increasing i by 10 each time it iterates but I am having a world of trouble. I looked for solutions but have been unable to find anything useful. I tried converting the size_t to an unsigned int and doing the arithmetic with that but unfortunately it didn't work. Basically all I want is this: for (int i = 0; i < event.packet->dataLength - 7; i+=10) { } Though every time I do something like this or attempt at my conversions the i < # part is a huge number. They gave a printf statement in a tutorial for the API which used "%u" to print the actual number however when I convert it to an unsigned int it is still incorrect. I'm not sure where to go from here. Any help would be greatly appreciated :)

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• Understanding byte order and functions like CFSwapInt32HostToBig

  - by Typeoneerror

  I've got an enumeration in my game. A simple string message with an appended PacketType is being sent with the message (so it knows what to do with the message) over GameKit WIFI connection. I used Apple's GKRocket sample code as a starting point. The code itself is working fantastically; I just want to understand what the line with CFSwapInt32HostToBig is doing. What on earth does that do? and why does it need to do it? My guess is that it's making sure the PacketType value can be converted to an unsigned integer so it can send it reliably, but that doesn't sound all that correct to me. The documentation states "Converts a 32-bit integer from big-endian format to the host’s native byte order." but I don't understand what the means really. typedef enum { PacketTypeStart, // packet to notify games to start PacketTypeRequestSetup, // server wants client info PacketTypeSetup, // send client info to server PacketTypeSetupComplete, // round trip made for completion PacketTypeTurn, // packet to notify game that a turn is up PacketTypeRoll, // packet to send roll to players PacketTypeEnd // packet to end game } PacketType; // .... - (void)sendPacket:(NSData *)data ofType:(PacketType)type { NSLog(@"sendPacket:ofType(%d)", type); // create the data with enough space for a uint NSMutableData *newPacket = [NSMutableData dataWithCapacity:([data length]+sizeof(uint32_t))]; // Data is prefixed with the PacketType so the peer knows what to do with it. uint32_t swappedType = CFSwapInt32HostToBig((uint32_t)type); // add uint to data [newPacket appendBytes:&swappedType length:sizeof(uint32_t)]; // add the rest of the data [newPacket appendData:data]; // Send data checking for success or failure NSError *error; BOOL didSend = [_gkSession sendDataToAllPeers:newPacket withDataMode:GKSendDataReliable error:&error]; if (!didSend) { NSLog(@"error in sendDataToPeers: %@", [error localizedDescription]); } }

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• recvfrom returns invalid argument when *from* is passed

  - by Aditya Sehgal

  I am currently writing a small UDP server program in linux. The UDP server will receive packets from two different peers and will perform different operations based on from which peer it received the packet. I am trying to determine the source from where I receive the packet. However, when select returns and recvfrom is called, it returns with an error of Invalid Argument. If I pass NULL as the second last arguments, recvfrom succeeds. I have tried declaring fromAddr as struct sockaddr_storage, struct sockaddr_in, struct sockaddr without any success. Is their something wrong with this code? Is this the correct way to determine the source of the packet? The code snippet follows. ` /*TODO : update for TCP. use recv */ if((pkInfo->rcvLen=recvfrom(psInfo->sockFd, pkInfo->buffer, MAX_PKTSZ, 0, /* (struct sockaddr*)&fromAddr,*/ NULL, &(addrLen) )) < 0) { perror("RecvFrom failed\n"); } else { /*Apply Filter */ #if 0 struct sockaddr_in* tmpAddr; tmpAddr = (struct sockaddr_in* )&fromAddr; printf("Received Msg From %s\n",inet_ntoa(tmpAddr->sin_addr)); #endif printf("Packet Received of len = %d\n",pkInfo->rcvLen); } `

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• TCP/IP Implementation General Questions

  - by user2971023

  I've implemented the concepts shown here; http://wiki.unity3d.com/index.php/Simple_TCP/IP_Client_-_Server outside of unity and it works. (though i had to create the TCPIPServerApp from scratch as i could not find the base project anywhere). I have some general questions on how to use tcp/ip properly however. I've done some research on tcp/ip itself but I'm still a little confused. It seems like using the method above doesn't guarantee that I'll see the message (res). It just checks on every update to see if there is a different message in res. What if multiple messages are sent and the program lags or something, will i miss the earlier packet(s)? Should i instead do an array so it stores the last X messages? How do i know the data was received? Do I need to add a message id and build in my own ack into the data? Should i check to see if the port is in use before setting up a connection? Sorry for all the questions. This is all new to me but I enjoy this very much! ... Below already answered By Anton, Thanks It sounds like tcp uses its own packet numbering to ensure the packets end up in the right order on the other side. What if a packet is missed, are the subsequent packets thrown away? Or is this numbering and packet ordering, only for handling data that is broken out into multiple packets? TCP will automatically break the data into multiple packets if necessary right?

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• recvfrom() return values in Stop-and-Wait UDP?

  - by mavErick

  I am trying to implement a Stop-and-Wait UDP client-server socket program in C. As known, there are basically three possible scenarios for Stop-and-Wait flow control. i.e., After transmitting a packet, the sender receives a correct ACK and thus starts transmitting the next packet; the sender receives an incorrect ACK and thus retransmits this packet; the sender receives no ACK within a TIMEOUT and thus retransmits this packet. My idea is to differentiate these three scenarios with the return value of recvfrom() on the sender side. For scenario 1&2: recvfrom() just returns the length of the received ACK. Since in my implementation the incorrect ACK is of the same length of the correct one, so I will have to go deeper and check the contents of the ACK. It's not a big deal. I know how to do. Problems come when I am trying to recognize scenario 3 where no ACK is received. What confuses me is that my recvfrom() is within a while loop, so the recvfrom() will be called constantly. What will it return when the receiver is not actually sending the sender ACK? Is it 0 or 1?

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