I'm just reading about TCP and UDP and from what I've read when the packet gets encapsulated in the IP protocol and the header gets added is it not possible to create a false IP address at this stage?
Granted, the response will never be received on your connection, but is this possible?
I do not want to do this btw. I'm in no way associated with immoral ethics. It's just something that I had to ask whilst reading about TCP and UDP. I'm actually learning how to use sockets in C++ for a game I'm working on.
Yes, this is possible.
Use raw sockets and craft your own packets
Use scapy, hping, etc
Search for "packet crafting"
EDIT
A nice scapy tutorial that teaches you to send lots of valid and invalid stuff is here.
Related
I'm writing a server that, among other things, needs to be constantly sending data in different multicast addresses. The packages being sent might be received by a client side (an app) which will be switching between the mentioned addresses.
I'm using Perfect (https://github.com/PerfectlySoft/Perfect) for writing the server side, however had no luck using the Perfect-Net module nor using CocoaAsyncSocket. How could i implement both the sender and the receiver using swift? Any could snippet would be really useful.
I've been reading about multicasting and when it comes to the receiver, i've notice that in most languages (i.e. java or c#) the receiver often indicates a port number and a multicast ip-address, but when is the connection with the server being made? When does the socket bind to the real server ip-address?
Thanks in advance
If we talk about the TCP/IP stack, only IP and UDP support broadcasts and multicasts. They're both connectionless, and this is why you see only sending and receiving to special multicast addresses, but no binds and connects. You see it in different languages because (a) protocols are language-agnostic and (b) most implementations put reasonable efforts in trying to be compatible with BSD sockets interface.
If you want that true multicast, you'll need to find a swift implementation of sockets that allow setting options. Usual names for this operation is setsockopt. Multicast sender side doesn't need anything beyond a basic UDP socket (I suggest using UDP, not IP), while sender needs to be added to a multicast group. This Python example pretty much describes it.
However, it's worth noting that routers don't route broadcasts and multicasts. Hence you cannot use it over internet. If you need to use internet in your project, I'd advise you to use TCP - or websockets if your clients are browsers - and send messages to "groups" of them manually.
I guess you actually want Perfect-Kafka or Perfect-Mosquitto - Message Queue which allows a server to publish live streams to the client side subscribers. Low-level sockets will not easily fulfill your requirement.
I am trying to implement a program that is able to send TCP packets alone by itself. Ideally I want to send a packet to a port in a computer and have it processed. This means that I am trying to do it without having a client/server files pair.
However, I am finding it really hard to do this as anywhere I look there is mention of both the client and the server files. And if I try to run the client file or the server files by themselves, it doesn't even work as they depend on each other.
Is it possible to do what I want?
Not sure why you want to do that. But in order to send such packets, you need to use API for IP raw socket (SOCK_RAW), instead of TCP.
So you can send IP packets with payload of your "crafted TCP".
I'm trying to spec out the foundations for a server application who's purpose will be to..
1 'receive' tcp and/or udp packets
2 interpret the contents (i.e. header values)
To add more detail, this server will receive 'sip invites' and respond with a '302 redirect'.
I have experience with Net::Pcap and perl, and know I could achieve this by looping for filtered packets, decoding and then using something like Net::SIP to respond.
However, there's a lot of bloat in both of these modules/applications I don't need. The server will be under heavy load, and if I run TCPDUMP on it's own, it loses packets in the kernel due to server load, so worry it wont be appropriate :(
Should I be able to achieve the same thing by 'listening' on a socket (using IO::Socket for example) and decoding a packet?
Unfortunatly by debugging, it's hard to tell if IO::Socket will give me the opportunity to see a raw packet? And instead it automatically decodes the message to a readable format!
tl;dr: I want to capture lots of SIP Invites, analyse the head values, and respond with a SIP 302 redirect. Is there a better way than using tcpdump (via Net::Pcap) to achieve this?
Thanks,
Moose
Is there a better way than using tcpdump (via Net::Pcap) to achieve this?
Yes. Using libpcap (that's what you meant instead of tcpdump in that question) is a bad way to implement a TCP-based service, as you will have to reimplement much of TCP yourself (libpcap gives you raw network-layer packets), and the packets your program gets will also get delivered to the Internet protocol stack on your machine, so:
if there's nothing on your machine listening on the TCP port to which the other machines are trying to connect, the connection requests will get a RST from the TCP code and think the connection attempt failed;
if there is something on your machine listening on that port, it'll probably accept the connection, and it and your program will both try to communicate with the other machine, which will probably confuse its TCP stack and cause various bad and random things to happen.
It's not much better for UDP:
if there's nothing on your machine listening on the UDP port to which the other machines are trying to connect, the connection requests will probably get an ICMP Port Unreachable message from the UDP code, which may make it think the connection attempt failed;
if there is something on your machine listening on that port, it'll probably accept the connection, and it and your program will both try to communicate with the other machine, which will probably confuse its SIP stack and cause various bad and random things to happen.
IO:Socket will probably not give you raw packets, and that's a good thing; you won't have to implement your own IP and TCP/UDP stack. If your goal is to implement a redirect server on your machine, you have no need to receive raw packets; you want to receive SIP INVITEs with all the lower-level processing done for you by your machine's IP/TCP/UDP stack.
If you already have a SIP implementation on your machine, and you want to act as a "firewall" for it, so that, for some INVITEs, you send back a 302 redirect and prevent the SIP implementation on your machine from ever seeing the INVITEs in question, you will need to use the same mechanism that your particular OS uses to implement firewalls. There is no libpcap-like wrapper for those mechanisms, as far as I know.
I have no experience with sockets nor multiplayer programming.
I need to code a multiplayer mode for a game I made in c++. It's a puzzle game but the game mode will not be turn-based, it's more like cooperative.
I decided to use UDP, so I've read some tutorials, and all the samples I find decribes how to create a client that sends data and a server that receives it.
My game will be played by two players, and both will send and receive data to/from the other.
Do I need to code a client and a server?
Should I use the same socket to send and receive?
Should I send and receive data in the same port?
Thanks, I'm kind of lost.
Read how the masters did it:
http://www.bluesnews.com/abrash/chap70.shtml
Read the code:
git clone git://quake.git.sourceforge.net/gitroot/quake/quake
Open one UDP socket and use sendto and recvfrom. The following file contains the functions for the network client.
quake/libs/net/nc/net_udp.c
UDP_OpenSocket calls socket (PF_INET, SOCK_DGRAM, IPPROTO_UDP)
NET_SendPacket calls sendto
NET_GetPacket calls recvfrom
Do I need to code a client and a server?
It depends. For a two player game, with both computers on the same LAN, or both on the open Internet, you could simply have the two computers send packets to each other directly.
On the other hand, if you want your game to work across the Internet, when one or both players are behind a NAT and/or firewall, then you have the problem that the NAT and/or firewall will probably filter out the other player's incoming UDP packets, unless the local player goes to the trouble of setting up port-forwarding in their firewall... something that many users are not willing (or able) to do. In that case, you might be better off running a public server that both clients can connect to, which forwards data from one client to another. (You might also consider using TCP instead of UDP in that case, at least as a fallback, since TCP streams are in general likely to have fewer issues with firewalls than UDP packets)
Should I use the same socket to send and receive?
Should I send and receive data in the same port?
You don't have to, but you might as well -- there's no downside to using just a single socket and a single port, and it will simplify your code a bit.
Note that this answer is all about using UDP sockets. If you change your mind to use TCP sockets, it will almost all be irrelevant.
Do I need to code a client and a server?
Since you've chosen to to use UDP (a fair choice if your data isn't really important and benefits more from lower latency than reliable communication), you don't have much of a choice here: a "server" is a piece of code for receiving packets from the network, and your "client" is for sending packets into the network. UDP doesn't provide any mechanism for the server to communicate to the client (unlike TCP which establishes a 2 way socket). In this case, if you want to have two way communication between your two hosts, they'll each need server and client code.
Now, you could choose to use UDP broadcasts, where both clients listen and send on the broadcast address (usually 192.168.1.255 for home networks, but it can be anything and is configurable). This is slightly more complex to code for, but it would eliminate the need for client/server configuration and may be seen as more plug 'n play for your users. However, note that this will not work over the Internet.
Alternatively, you can create a hybrid method where hosts are discovered by broadcasting and listening for broadcasts, but then once the hosts are chosen you use host to host unicast sockets. You could provide fallback to manually specify network settings (remote host/port for each) so that it can work over the Internet.
Finally, you could provide a true "server" role that all clients connect to. The server would then know which clients connected to it and would in turn try to connect back to them. This is a server at a higher level, not at the socket level. Both hosts still need to have packet sending (client) and receiving (server) code.
Should I use the same socket to send and receive?
Well, since you're using UDP, you don't really have a choice. UDP doesn't establish any kind of persistent connection that they can communicate back and forth over. See the above point for more details.
Should I send and receive data in the same port?
In light of the above question, your question may be better phrased "should each host listen on the same port?". I think that would certainly make your coding easier, but it doesn't have to. If you don't and you opt for the 3rd option of the first point, you'll need a "connect back to me on this port" datafield in the "client's" first message to the server.
UDP doesnot sends any ack back, but will it send any response?
I have set up client server UDP program. If I give client to send data to non existent server then will client receive any response?
My assumption is as;
Client -->Broadcast server address (ARP)
Server --> Reply to client with its mac address(ARP)
Client sends data to server (UDP)
In any case Client will only receive ARP response. If server exists or not it will not get any UDP response?
Client is using sendto function to send data. We can get error information after sendto call.
So my question is how this info is available when client doesn't get any response.
Error code can be get from WSAGetLastError.
I tried to send data to non existent host and sendto call succeeded . As per documentation it should fail with return value SOCKET_ERROR.
Any thoughts??
You can never receive an error, or notice for a UDP packet that did not reach destination.
The sendto call didn't fail. The datagram was sent to the destination.
The recipient of the datagram or some router on the way to it might return an error response (host unreachable, port unreachable, TTL exceeded). But the sendto call will be history by the time your system receives it. Some operating systems do provide a way to find out this occurred, often with a getsockopt call. But since you can't rely on getting an error reply anyway since it depends on network conditions you have no control over, it's generally best to ignore it.
Sensible protocols layered on top of UDP use replies. If you don't get a reply, then either the other end didn't get your datagram or the reply didn't make it back to you.
"UDP is a simpler message-based connectionless protocol. In connectionless protocols, there is no effort made to set up a dedicated end-to-end connection. Communication is achieved by transmitting information in one direction, from source to destination without checking to see if the destination is still there, or if it is prepared to receive the information."
The machine to which you're sending packets may reply with an ICMP UDP port unreachable message.
The UDP protocol is implemented on top of IP. You send UDP packets to hosts identified by IP addresses, not MAC addresses.
And as pointed out, UDP itself will not send a reply, you will have to add code to do that yourself. Then you will have to add code to expect the reply, and take the proper action if the response is lost (typically resend on a timer, until you decide the other end is "dead"), and so on.
If you need reliable UDP as in ordering or verification such that TCP/IP will give you take a look at RUDP or Reliable UDP. Sometimes you do need verification but a mixture of UDP and TCP can be held up on the TCP reliability causing a bottleneck.
For most large scale MMO's for isntance UDP and Reliablity UDP are the means of communication and reliability. All RUDP does is add a smaller portion of TCP/IP to validate and order certain messages but not all.
A common game development networking library is Raknet which has this built in.
RUDP
http://www.javvin.com/protocolRUDP.html
An example of RUDP using Raknet and Python
http://pyraknet.slowchop.com/