I am new to python programming and for my project I need to automate some of the misconfiguration cases(by looking into the flow of tcp packets and tcp flags) of DNS over TCP to write into a database. For that I am trying to parse the captured tcp dumps. Can you help of how can I proceed
dig +tcp will tell dig to use TCP:
+[no]tcp
Use [do not use] TCP when querying name servers. The default
behavior is to use UDP unless an AXFR or IXFR query is requested,
in which case a TCP connection is used.
that should make grabbing the pcap much simpler
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I need to write a proxy server in C language on Linux (Ubuntu 20.04). The purpose of this proxy server is as follows. There're illogical governmental barriers in accessing the free internet. Some are:
Name resolution: I ping telegram.org and many other sites which the government doesn't want me to access. I ask 8.8.8.8 to resolve the name, but they response of behalf of the server that the IP may be resolved to 10.10.34.35!
Let's concentrate on this one, because when this is solved many other problems will be solved too. For this, I need to setup such a configuration:
A server outside of my country is required. I prepared it. It's a VPS. Let's call it RS (Remote Server).
A local proxy server is required. Let's call it PS. PS runs on the local machine (client) and knows RS's IP. I need it to gather all requests going to be sent through the only NIC available on client, process them, scramble them, and send them to RS in a way to be hidden from the government.
The server-side program should be running on RS on a specific port to get the packet, unscramble it, and send it to the internet on behalf of the client. After receiving the response from the internet, it should send it back to the client via the PS.
PS will deliver the response to the client application which originates the request. Of course this happens after it will unscramble and will find the original response from the internet.
This is the design and some parts is remained gloomy for me. Since I'm not an expert in network programming context, I'm going to ask my questions in the parts I'm getting into trouble or are not clear for me.
Now, I'm in part 2. See whether I'm right. There're two types of sockets, a RAW socket and a stream socket. A RAW socket is opened this way:
socket(AF_PACKET, SOCK_RAW, htons(ETH_P_ALL));
And a stream socket is opened this way:
socket(AF_INET, SOCK_STREAM, 0);
For RAW sockets, we use sockaddr_ll and for stream sockets we use sockaddr_in. May I use stream sockets between client applications and PS? I think not, because I need the whole RAW packet. I should know the protocol and maybe some other info of the packet, because the whole packet should be retrieved transparently in RS. For example, I should know whether it has been a ping packet (ICMP) or a web request (TCP). For this, I need to have packet header in PS. So I can't use a stream socket, because it doesn't contain the packet header. But until now, I've used RAW sockets for interfaces and have not written a proxy server to receive RAW packets. Is it possible? In another words, I've the following questions to go to next step:
Can a RAW socket be bound to localhost:port instead of an interface so that it may receive all low-level packets containing packet headers (RAW packets)?
I may define a proxy server for browser. But can I put the whole system behind the proxy server so that packets of other apps like PING may route automatically via it?
Do I really need RAW sockets in PS? Can't I change the design to suffice the data I got from the packets payload?
Maybe I'm wrong in some of the concepts and will appreciate your guidance.
Thank you
Can a RAW socket be bound to localhost:port instead of an interface so that it may receive all low-level packets containing packet headers (RAW packets)?
No, it doesn't make sense. Raw packets don't have port numbers so how would it know which socket to go to?
It looks like you are trying to write a VPN. You can do this on Linux by creating a fake network interface called a "tun interface". You create a tun interface, and whenever Linux tries to send a packet through the interface, instead of going to a network cable, it goes to your program! Then you can do whatever you like with the packet. Of course, it works both ways - you can send packets from your program back to Linux through the tun interface, and Linux will act like they just arrived on a network cable.
Then, you can set up your routing table so that all traffic goes to the tun interface, except for traffic to the VPN server ("RS"), which goes to your real ethernet/wifi interface. Otherwise you'd have an endless loop where your VPN program PS tried to send packets to RS but they just went back to PS.
I'm trying code TCP server in C language. I just noticed accept() function returns when connection is already established.
Some clients are flooding with random data some clients are just sending random data for one time, after that I want to close their's current connection and future connections for few minutes (or more, depends about how much load program have).
I can save bad client IP addresses in a array, can save timings too but I cant find any function for abort current connection or deny future connections from bad clients.
I found a function for windows OS called WSAAccept that allows you deny connections by user choice, but I don't use windows OS.
I tried code raw TCP server which allows you access TCP packet from begin including all TCP header and it doesn't accept connections automatically. I tried handle connections by program side including SYN ACK and other TCP signals. It worked but then I noticed raw TCP server receiving all packets in my network interface, when other programs using high traffic it makes my program laggy too.
I tried use libnetfilter which allows you filter whole traffic in your network interface. It works too but like raw TCP server it also receiving whole network interface's packets which is making it slow when there is lot of traffic. Also I tried compare libnetfilter with iptables. libnetfilter is slower than iptables.
So in summary how I can abort client's current and future connection without hurt other client connections?
I have linux with debian 10.
Once you do blacklisting on packet level you could get very fast vulnerable to very trivial attacks based on IP spoofing. For a very basic implementation an attacker could use your packet level blacklisting to blacklist anyone he wants by just sending you many packets with a fake source IP address. Usually you don't want to touch these filtering (except you really know what you are doing) and you just trust your firewall etc. .
So I recommend really just to close the file descriptor immediately after getting it from accept.
I'm testing SSH connection under Linux.
With using tcpdump I noticed that TCP FIN flag is set in common SSH packet data.
For testing purposes I'd like to achieve the situation where packet with TCP FIN flag is sent as a separate packet, so it would be a packet with no data, but with FIN flag set.
I've been looking for such a possibility in "man 7 socket" but didn't find.
My question is - how to achieve such a functionality in Linux? Any ideas?
Tuning a TCP connection this way can not be done and it would not make really sense to control it. If you need to produce traffic with this behavior you cannot use normal TCP sockets but need to use raw sockets where you can set the header like you want. You would of course need to re-implement all the parts of the TCP connection you need in your application just to achieve this feature.
There's already a question How exactly does a remote program like team viewer work which gives a basic description, but I'm interested in how the comms works once the client has registered with the server. If the client is behind a NAT then it won't have its own IP address so how can the server (or another client) send a message to it? Or does the client just keep polling the server to see if its got any requests?
Are there any open source equivalents of LogMeIn or TeamViewer?
The simplest and most reliable way (although not always the most efficient) is to have each client make an outgoing TCP connection to a well-known server somewhere and keep that connection open. As long as the TCP connection is open, data can pass over that TCP connection in either direction at any time. It appears that both LogMeIn and TeamViewer use this method, at least as a fall-back. The main drawbacks for this technique are that all data has to pass through a TeamViewer/LogMeIn company server (which can become a bottleneck), and that TCP doesn't handle dropped packets very well -- it will stall and wait for the dropped packets to be resent, rather than giving up on them and sending newer data instead.
The other technique that they can sometimes use (in order to get better performance) is UDP hole-punching. That technique relies on the fact that many firewalls will accept incoming UDP packets from remote hosts that the firewalled-host has recently sent an outgoing UDP packet to. Given that, the TeamViewer/LogMeIn company's server can tell both clients to send an outgoing packet to the IP address of the other client's firewall, and after that (hopefully) each firewall will accept UDP packets from the other client's Internet-facing IP address. This doesn't always work, though, since different firewalls work in different ways and may not include the aforementioned UDP-allowing logic.
I have a very large network trace file which contains both tcp and udp packets.I want to find out the flows in the trace file.For that I have a hash function which takes in source ip address,destination ip address,source port,destination port and protocol.In case of TCP I can understand that the flow means all the packets which have the same 5 parameters same.But what does it mean in case of UDP.how does the concept of flow apply in case of UDP.? I am a novice in packet processing.And once I have idendified a flow (tcp and udp) how do I work out the direction of the flow.?Do I have to look at SYN flag ? If yes what do I do for UDP?
Netflow applies to any protocol including TCP and UDP. So to answer your question, yes, UDP packets should be treated the same as TCP.
If you do Netflow processing you might find this spec useful - very detailed and for many versions. I can confirm that it is accurate and works fine with Cisco and Juniper devices (at least version 7)