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).
Related
I have a trouble to tune TCP client-server communication.
My current project has a client, running on PC (C#) and a server,
running on embedded Linux 4.1.22-ltsi.
Them use UDP communication to exchanging data.
The client and server work in blocking mode and
send short messages one to 2nd
(16, 60, 200 bytes etc.) that include either command or set of parameters.
The messages do note include any header with message length because
UDP is message oriented protocol. Its recvfrom() API returns number of received bytes.
For my server's program structure is important to get and process entire alone message.
The problem is raised when I try to implement TCP communication type instead of UDP.
The server's receive buffer (recv() TCP API) is 2048 bytes:
#define UDP_RX_BUF_SIZE 2048
numbytes = recv(fd_connect, rx_buffer, UDP_RX_BUF_SIZE, MSG_WAITALL/*BLOCKING_MODE*/);
So, the recv() API returns from waiting when rx_buffer is full, i.e after it receives
2048 bytes. It breaks all program approach. In other words, when client send 16 bytes command
to server and waits an answer from it, server's recv() keeps the message
"in stomach", until it will receive 2048 bytes.
I tried to fix it as below, without success:
On client side (C#) I set the socket parameter theSocket.NoDelay.
When I checked this on the sniffer I saw that client sends messages "as I want",
with requested length.
On server side I set TCP_NODELAY socket option to 1
int optval= 1;
setsockopt(fd,IPPROTO_TCP, TCP_NODELAY, &optval, sizeof(optval);
On server side (Linux) I checked socket options SO_SNDLOWAT/SO_RCVLOWAT and they are 1 byte each one.
Please see the attached sniffer's log picture. 10.0.0.10 is a client. 10.0.0.106 is a server. It is seen, that client activates PSH flag (push), informing the server side to move the incoming data to application immediately and do not fill a buffer.
Additional question: what is SSH encrypted packets that runs between the sides. I suppose that it is my Eclipse debugger on PC (running server application through the same Ethernet connection) sends them. Am I right?
So, my problem is how to cause `recv() API to return each short message (16, 60, 200 bytes etc.) instead of accumulating them until receiving buffer fills.
TCP is connection oriented and it also maintains the order in which packets are sent and received.
Having said that, in TCP client, you will receive the stream of bytes and not the individual udp message as in UDP. So you will need to send the packet length and marker as the initial bytes.
So client can first find the packet length and then read data till packet length is reached and then expect new packet length.
You can also check for library like netty, zmq to do this extra work
I modified the ovs2.6.0 and want to handle some packets.
I printk the prev_port(prev_port = nla_get_u32(a)) in do_execute_actions function. However, the prev_port sometimes is not the correct port number which the packet is forwarded to. But the TCP transmission is normal. I guess that the port number in ovs is numbered in a random way?
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.
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 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.