I am wondering what is happening between the creating a TCP packet and a [Ethernet[IP[TCP-packet]]] leaving the network adapter.
When i use for example a TCP program and want to send a single packet ( or could be more in fact TCP using byte streaming).
So i set up in any language a function called socket(...);
So my OS, refering to any documenation, creating me an interface with a specified port on which I can receive and send data over.
And if I create a TCP package (for example sendto(...), it will be send to the socket.
But what kind processes are done now [1], until my packet will leave the network adapter with an Ethernet + IP Header?
[1]: Where are the following steps happening (OS/Network adapter) and how does it exactly work?
Hope you understand me.. and please correct me if I missunderstood something wrong.
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I`m writing packet generator right now. Testing it with wireshark and VM. I have an exercise on my checklist to sent 3 packets in a row:
1. TCP on 80 port, with SYN=1 and MF=1 flags.
2. TCP on 135 port, with SYN=1 and MF=1 flags.
3. TCP on 80 port, with MF = 0 and offset = 24.
I`m sending all the packets with the same ID field on IP layer.So as I understand Wireshark should try to reassemble these packets.
But will it reassemble packets from different ports?And what should we get as final result?
All I get is 3 IPv4 packets.
http://cs625124.vk.me/v625124860/10bf5/BQFUbKT7zVs.jpg
Addition: I mentioned, that if I change offset of last TCP-packet to 16, than we got a bit different kind of traffic.:
We got one HTTP or continuos packet. And here is wrong checksum. I tried to copy correct checksum to the first TCP packet and then I got RST, so i think that WireShark interpreted SYN from 1-st packet:
http://s28.postimg.org/z3w7ibhjx/image.png
So could you please explain me, was the last result correct? I would appreciate any help. Sorry if it is something basic. It`s my first expirience of writing WinForm application and using Pcap.Net library too. Thanks in advance!Sorry for links, have no reputation(
First, a TCP session is defined by the tuple:
Side A's IP address.
Side A's Port.
Side B's IP address.
Side B's Port.
If you have packets with different tuples, they will not be part of the same TCP session.
You get a RST when the server closes the session.
It is likely the server doesn't like getting SYN packets from port 21 (FTP) to its port 80 (HTTP).
I wanted to implement the three-way handshake using python.And here is what I did:
1-I created syn packet .
2-I send the packet to the destination .
3-I created a function that will listen to all the traffic that pass through my NIC .it's kinda like sniffer.If this function were to find a packet that is destined to my IP address and the same port that I send the syn packet through, it will parse it .
4-If the flags in the captured packet are set to syn+ack,the function will generate a TCP packet with the ack flag set.
The problem is, before I send the ack packet the system send RST packet .
So , what is the meaning of the behaviour?? Is there anyway to stop it??
Note:
I am not implementing the three-way handshake for production purposes.I just want to understand how the protocol TCP work.
Disable the system's TCP/IP stack so that you have complete control over what it sends over the network. Usually the easiest way to do this is simply not to configure IP on the interface you are using. Note that you will need to handle ARP requests.
I'm writing a TCP client to connect, send messages to, and read responses from nodes on a busy network. I do not have access to code running on those nodes, so I have to sculpt the TCP messages I send out over the wire very carefully.
I've decided that I'm going to gather a bunch of live TCP data off of the network and use that as the basis for testing my client - given these input parameters, make sure that the transmitted binary renders into the hex I expect to see.
At this point I'm about to either fire up Wireshark and figure out how to filter for the packets that I'm sending or implement a really simple TCP server that waits for connections, spits the transmitted binary to disk (maybe processed to hex) and then test that I'm sending what I expect to send; neither of which feel like robust or professional solutions.
So: how would you recommend setting up a test harness for a TCP client to verify the right hex messages are coming out?
You should check out aleph for tcp and gloss for protocols. here is a tutorial and an example.
a simple tcp echo-server could be:
(defn echo-handler [ch client-info]
(siphon ch ch)) ;; a simple echo handler
(defn start-server [] (start-tcp-server
echo-handler
{:port 9997}))
(start-server)
gloss will be very useful at spitting the received messages either as hex dump or as values if you encode your protocol in it and use "decode".
I wrote a TCP socket program,and define a text protocol format like: "length|content",
to make it simple, the "length" is always 1-byte-long and it define the number of bytes of "content"
My problem is:
when attackers send packets like "1|a51",it will stay in tcp's receive buffer
the program will parse it wrong and the next packet would start like "5|1XXXX"，
then the rest of the packets remain in the buffer would all parsed wrong，
how to solve this problem?
If you get garbage, just close the connection. It's not your problem to figure out what they meant, if anything.
instead of length|content only, you also need to provide a checksum, if the checksum is not correct, you should drop the connection to avoid partial receive.
this is a typical problem in tcp protocol, since the tcp is stream based. but just as http, which is an application of tcp protocol, it has a structure of request / response to make sure each end of the connection knows when the data has been fully transferred.
but your scenario is a little bit tricky, since the hacker can only affect the connection of his own. while it cannot change the data from other connections, only if he can control the route / switcher between your application and the users.
I need an app that sends an UDP packet to some network server and receives the response. The server replies to the same port number where request came from, so I first need to bind() my socket to any UDP port number.
Hardcoding the UDP port number is a bad idea, as it might be used by any other application running on the same PC.
Is there a way to bind an UDP socket to any port available? IMO it should be an effective way to quickly obtain a free port #, which is used by e.g. accept() function.
If no, then what's the best strategy to try binding and check for WSAEADDRINUSE/EADDRINUSE status: try the ports sequentially starting from from 1025, or 1025+rand(), or some other?
Another option is to specify port 0 to bind(). That will allow you to bind to a specific IP address (in case you have multiple installed) while still binding to a random port. If you need to know which port was picked, you can use getsockname() after the binding has been performed.
Call sendto without calling bind first, the socket will be bound automatically (to a free port).
I must be missing something, why don't you use the udp socket to send back data?
Start with sendto and then use recvfrom function to read incoming data also you get as a bonus the address from which the data was sent, right there for you to send a response back.