I was reading the nmap source code because I'd like to find out how does it discover that certain ports are filtered or firewalled. I have some experience with sockets in c and i've built simple port scanners, that's easy - if the connection succeeds, the port is open, otherwise it's closed (because of the RST returned). But in case with the firewalled ports, they don't return RST packet back, and my port scanner just "waits" forever.
If someone's got experience with this topic, please point me to the parts of the nmap code where the actual scanning and port-state determination occurs, or at least tell me if there are any other codes available which deal with this problem.
Use asynchronous socket API calls (i.e. don't wait for the connection to be established, and instead try the next port/address in parallel) and define a reasonable timeout (e.g. if the connection isn't established after a minute you can consider it filtered).
Related
There are many threads regarding how to detect if a socket is connected or not using various methods like getpeername / getsockopt w/ SO_ERROR. https://man7.org/linux/man-pages/man2/getpeername.2.html would be a good way for me to detect if a socket is connected or not. The problem is, it does not say anything about if the connection is in progress... So if i call connect, it is in progress, then i call getpeername, will it say it is an error (-1) even though the connection is still in progress?
If it does, I can implement a counter-like system that will eventually kill the socket if it is still in progress after x seconds.
Short Answer
I think that, if getpeername() returns ENOTCONN, that simply means that the tcp connection request has not yet succeeded. For it to not return ENOTCONN, I think the client end needs to have received the syn+ack from the server and sent its own ack, and the server end needs to have received the client's ack.
Thereafter all bets are off. The connection might subsequently be interrupted, but getpeername() has no way of knowing this has happened.
Long Answer
A lot of it depends on how fussy and short-term one wants to be about knowing if the connection is up.
Strictly Speaking...
Strictly speaking with maximum fussiness, one cannot know. In a packet switched network there is nothing in the network that knows (at any single point in time) for sure that there is a possible connection between peers. It's a "try it and see" thing.
This contrasts to a circuit switched network (e.g. a plain old telephone call), where there is a live circuit for exclusive use between peers (telephones); provided current is flowing, you know the circuit is complete even if the person at the other end of the phone call is silent.
Note that if the two computers were connected by a single Ethernet cable (no router, no switches, just a cable between NICs), that is effectively a fixed circuit (not even a circuit-switched network).
Relaxing a Little...
Focusing on what one can know about a connection in a packet switched network. As others have already said, the answer is that, really, one has to send and receive packets constantly to know if the network can still connect the two peers.
Such an exchange of packets occurs with a tcp socket connect() - the connecting peer sends a special packet to say "please can I connect to you", and the serving peer replies "yes", the client then says "thank you!" (syn->, <-syn+ack, ack->). But thereafter the packets flow between peers only if the applications send and receive data, or elects to close the connection (fin).
Calling something like getpeername() I think is somewhat misleading, depending on your requirements. It's fine, if you trust the network infrastructure and remote computer and its application to not break, and not crash.
It's possible for the connect() to succeed, then something breaks somewhere in the network (e.g. the peer's network connection is unplugged, or the peer crashes), and there is no knowledge at your end of the network that that has happened.
The first thing you can know about it is if you send some traffic and fail to get a response. The response is, initially, the tcp acks (which allows your network stack to clear out some of its buffers), and then possibly an actual message back from the peer application. If you keep sending data out into the void, the network will quite happily route packets as far as it can, but your tcp stack's buffers will fill up due to the lack of acks coming back from the peer. Eventually, your network socket blocks on a call to write(), because the local buffers are full.
Various Options...
If you're writing both applications (server and client), you can write the application to "ping pong" the connection periodically; just send a message that means nothing other than "tell me you heard this". Successful ping-ponging means that, at least within the last few seconds, the connection was OK.
Use a library like ZeroMQ. This library solves many issues with using network connections, and also includes (in modern version) socket heartbeats (i.e. a ping pong). It's neat, because ZeroMQ looks after the messy business of making, restoring and monitoring connections with a heartbeat, and can notify the application whenever the connection state changes. Again, you need to be writing both client and server applications, because ZeroMQ has it's own protocol on top of tcp that is not compatible with just a plain old socket. If you're interested in this approach, the words to look for in the API documentation is socket monitor and ZMQ_HEARTBEAT_IVL;
If, really, only one end needs to know the connection is still available, that can be accomplished by having the other end just sending out "pings". That might fit a situation where you're not writing the software at both ends. For example, a server application might be configured (rather than re-written) to stream out data regardless of whether the client wants it or not, and the client ignores most of it. However, the client knows that if it is receiving data it then also knows there is a connection. The server does not know (it's just blindly sending out data, up until its writes() eventually block), but may not need to know.
Ping ponging is also good in that it gives some indication of the performance of the network. If one end is expecting a pong within 5 seconds of sending a ping but doesn't get it, that indicates that all is not as expected (even if packets are eventually turning up).
This allows discrimination between networks that are usefully working, and networks that are delivering packets but too slowly to be useful. The latter is still technically "connected" and is probably represented as connected by other tests (e.g. calling getpeername()), but it may as well not be.
Limited Local Knowledge...
There is limited things one can do locally to a peer. A peer can know whether its connection to the network exists (e.g. the NIC reports a live connection), but that's about it.
My Opinion
Personally speaking, I default to ZeroMQ these days if at all possible. Even if it means a software re-write, that's not so bad as it seems. This is because one is generally replacing code such as connect() with zmq_connect(), and recv() with zmq_revc(), etc. There's often a lot of code removal too. ZeroMQ is message orientated, a tcp socket is stream orientated. Quite a lot of applications have to adapt tcp into a message orientation, and ZeroMQ replaces all the code that does that.
ZeroMQ is also well supported across numerous languages, either in bindings and / or re-implementations.
man connect
If the initiating socket is connection-mode, .... If the connection cannot be established immediately and O_NONBLOCK is not set for the file descriptor for the socket, connect() shall block for up to an unspecified timeout interval until the connection is established. If the timeout interval expires before the connection is established, connect() shall fail and the connection attempt shall be aborted.
If connect() is interrupted by a signal that is caught while blocked waiting to establish a connection, connect() shall fail and set errno to [EINTR], but the connection request shall not be aborted, and the connection shall be established asynchronously.
If the connection cannot be established immediately and O_NONBLOCK is set for the file descriptor for the socket, connect() shall fail and set errno to [EINPROGRESS], but the connection request shall not be aborted, and the connection shall be established asynchronously.
When the connection has been established asynchronously, select() and poll() shall indicate that the file descriptor for the socket is ready for writing.
If the socket is in blocking mode, connect will block while the connection is in progress. After connect returns, you'll know if a connection has been established (or not).
A signal could interrupt the (blocking/waiting) process, the connection routine will then switch to asynchronous mode.
If the socket is in non blocking mode (O_NONBLOCK) and the connection cannot be established immediately, connect will fail with the error EINPROGRESS and like above switching to asynchronous mode, that means, you'll have to use select or poll to figure out if the socket is ready for writing (indicates established connection).
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.
Looking into nginx: ignore some requests without proper Host header got me thinking that it's not actually possible to close(2) a TCP connection without the OS properly terminating the underlying TCP connection by sending an RST (and/or FIN) to the other end.
One workaround would be to use something like tcpdrop(8), however, as can be seen from usr.sbin/tcpdrop/tcpdrop.c on OpenBSD and FreeBSD, it's implemented through a sysctl-based interface, and may have portability issues outside of BSDs. (In fact, it looks like even the sysctl-based implementation may be different enough between OpenBSD and FreeBSD to require a porting layer -- OpenBSD uses the tcp_ident_mapping structure (which, subsequently, contains two sockaddr_storage elements, plus some other info), whereas FreeBSD, DragonFly and NetBSD use an array of two sockaddr_storage elements directly.) It turns out, that OpenBSD's tcpdrop does appear to send the R packet as per tcpdump(8), and can be confirmed by looking at /sys/netinet/tcp_subr.c :: tcp_drop(), which calls tcp_close() in the end (and tcp_close() is confirmed to send RST elsewhere on SO), so, it appears that it wouldn't even work, either.
If I'm establishing the connection myself through C, is there a way to subsequently drop it without an acknowledgement to the other side, e.g., without initiating RST?
If I'm establishing the connection myself through C, is there a way to subsequently drop it without an acknowledgement to the other side, e.g., without initiating RST?
No. Even if there was, if the peer subseqently sent anything it would be answered by an RST.
NB Normal TCP termination uses a FIN, not an RST.
Cheating an attacker in this way could be a good idea. Of course, in this case you are already reserving server resources for the established connection. In the most basic mode you can use netfilter to drop any TCP outgoing segment with RST or FIN flags set. This rule iptables rule could be an example:
sudo iptables -A OUTPUT -p tcp --tcp-flags FIN,RST SYN -j DROP
Of course, this rule will affect all your TCP connections. I wrote it just to provide a lead of how you can do it. Go to https://www.netfilter.org/ for getting more ideas working on your solution. Basically you should be able to do the same only for the selected connections.
Because of how TCP works, if you're able to implement it, the client (or attacker) will keep the connection open for a long time. To understand the effect it would have in a client read here: TCP, recv function hanging despite KEEPALIVE where I provide results of a test in which the other side doesn't return any TCP segment (not even an ACK). In my configuration, it takes 13 minutes for the socket to enter an error state (that depends on Linux parameters like tcp_retries1 and tcp_retries2).
Just consider a DoS attack will usually imply connections from thousands of different devices and not necessarily many connections from the same device. This is very easy to detect and block in the firewall. So, it's very improbable that you are going to generate resources exhaustion in the client. Also this solution will not work for cases of half-open connection attack.
I was trying to learn the usage of option SO_KEEPALIVE in socket programming in C language under Linux environment.
I created a server socket and used my browser to connect to it. It was successful and I was able to read the GET request, but I got stuck on the usage of SO_KEEPALIVE.
I checked this link keepalive_description#tldg.org but I could not find any example which shows how to use it.
As soon as I detect the client's request on accept() function I set the SO_KEEPALIVE option value 1 on the client socket. Now I don't know, how to check if the client is down, how to change the time interval between the probes sent etc.
I mean, how will I get the signal that the client is down? (Without reading or writing at the client - I thought I will get some signal when probes are not replied back from client), how should I program it after setting the option SO_KEEPALIVE on).
Also if suppose the probes are sent every 3 secs and the client goes down in between I will not get to know that client is down and I may get SIGPIPE.
Anyways importantly I wanna know how to use SO_KEEPALIVE in the code.
To modify the number of probes or the probe intervals, you write values to the /proc filesystem like
echo 600 > /proc/sys/net/ipv4/tcp_keepalive_time
echo 60 > /proc/sys/net/ipv4/tcp_keepalive_intvl
echo 20 > /proc/sys/net/ipv4/tcp_keepalive_probes
Note that these values are global for all keepalive enabled sockets on the system, You can also override these settings on a per socket basis when you set the setsockopt, see section 4.2 of the document you linked.
You can't "check" the status of the socket from userspace with keepalive. Instead, the kernel is simply more aggressive about forcing the remote end to acknowledge packets, and determining if the socket has gone bad. When you attempt to write to the socket, you will get a SIGPIPE if keepalive has determined remote end is down.
You'll get the same result if you enable SO_KEEPALIVE, as if you don't enable SO_KEEPALIVE - typically you'll find the socket ready and get an error when you read from it.
You can set the keepalive timeout on a per-socket basis under Linux (this may be a Linux-specific feature). I'd recommend this rather than changing the system-wide setting. See the man page for tcp for more info.
Finally, if your client is a web browser, it's quite likely that it will close the socket fairly quickly anyway - most of them will only hold keepalive (HTTP 1.1) connections open for a relatively short time (30s, 1 min etc). Of course if the client machine has disappeared or network down (which is what SO_KEEPALIVE is really useful for detecting), then it won't be able to actively close the socket.
As already discussed, SO_KEEPALIVE makes the kernel more aggressive about continually verifying the connection even when you're not doing anything, but does not change or enhance the way the information is delivered to you. You'll find out when you try to actually do something (for example "write"), and you'll find out right away since the kernel is now just reporting the status of a previously set flag, rather than having to wait a few seconds (or much longer in some cases) for network activity to fail. The exact same code logic you had for handling the "other side went away unexpectedly" condition will still be used; what changes is the timing (not the method).
Virtually every "practical" sockets program in some way provides non-blocking access to the sockets during the data phase (maybe with select()/poll(), or maybe with fcntl()/O_NONBLOCK/EINPROGRESS&EWOULDBLOCK, or if your kernel supports it maybe with MSG_DONTWAIT). Assuming this is already done for other reasons, it's trivial (sometimes requiring no code at all) to in addition find out right away about a connection dropping. But if the data phase does not already somehow provide non-blocking access to the sockets, you won't find out about the connection dropping until the next time you try to do something.
(A TCP socket connection without some sort of non-blocking behaviour during the data phase is notoriously fragile, as if the wrong packet encounters a network problem it's very easy for the program to then "hang" indefinitely, and there's not a whole lot you can do about it.)
Short answer, add
int flags =1;
if (setsockopt(sfd, SOL_SOCKET, SO_KEEPALIVE, (void *)&flags, sizeof(flags))) { perror("ERROR: setsocketopt(), SO_KEEPALIVE"); exit(0); };
on the server side, and read() will be unblocked when the client is down.
A full explanation can be found here.
im writing a multithreaded winsock application and im having some issues with closing the sockets.
first of all, is there a limit for a number of simultaneously open sockets? lets say like 32 sockets all in once.
i establish a connection on one of the sockets, and passing information and it all goes right.
problem is when i disconnect the socket and then reconnect to the same destination, i get a RST from the server after my SYN.
i dont have the code for the server app so i cant debug it.
when i used SO_LINGER and it sent a RST flag at the end of each session - it worked.
but i dont want to end my connections this way.
when not using SO_LINGER a FIN flag was sent but it seems the connection was not really closed.
any help?
thanks
On Unix there's a file descriptor limit per process - I'm guessing on Windows it's "handles".
You are probably bind()-ing your client socket to a fixed port. That might be the reason the server is rejecting your subsequent connection. Try normal ephemeral ports.
Firstly, I agree with Nikolai, are you binding your client socket?
If so it sounds like the socket on the server side is still in TIME_WAIT and is discarding the new connection attempt. By binding the client socket you're forcing the server to try and reuse the exact same connection that is currently in the 2MSL wait period, it can't be reused at this point in time and so you're seeing what you're seeing. There's usually no need to bind the client port, stop doing it and your problem will likely go away.
Secondly, yes, there are limits to the number of open sockets on Windows platforms but they're resource related rather than some hard coded number.
Each open socket uses some 'non paged pool' memory and each pending read or write request on a socket is also likely to use both 'non paged pool' and have pages of memory locked in memory during I/O (there's a limit to the number of pages that can be locked). That said on Vista and later there's much more 'non paged pool' available than on earlier versions of Windows and even then I've managed to achieve more than 70,000 concurrent active connections on a pretty low spec XP box (see here: http://www.lenholgate.com/blog/2005/11/windows-tcpip-server-performance.html). Note that there are some separate limits on the number of outbound connections that you can establish (which is more likely to be of interest to you) but that's around 4000 by default and can be tuned by setting MAX_USER_PORT see here: Maximum number of concurrent TCP/IP connections - Win XP SP3 for more details.