Broadcasting ip:port by socket server - sockets

I'm trying to find a way for client to know socket server ip:port, without explicitly defining it. Generally I have a socket server running on portable device that's connect to network over DHCP (via WiFi), and ideally clients should be able to find it automaticaly.
So I guess a question is whether socket server can somehow broadcast it's address over local network? I think UPnP can do this, but I'd rather not get into it.
I'm quite sure that this question was asked on Stack lot's of times, but I could find proper keywords to search for it.

One method of doing this is via UDP broadcast packets. See beej's guide if you're using BSD sockets. And here is Microsoft's version of the same.
Assuming all the clients of the application are on the same side of a router then a broadcast address of 255.255.255.255 (or ff02::1 for IPv6) should be more than adequate.
Multicast is another option, but if this is a LAN-only thing I don't think that's necessary.
Suggestion
Pick a UDP port number (say for the sake of an example we pick 1667). The client should listen to UDP messages on 255.255.255.255:1667 (or whatever the equivalent is. e.g.: IPEndPoint(IPAddress.Any, 1667)). The server should broadcast messages on the same address.
Format Suggestion
UDP Packet: First four bytes as a magic number, next four bytes an IPv4 address (and you might want to add other things like a server name).
The magic number is just in case there is a collision with another application using the same port. Check both the length of the packet and the magic number.
Server would broadcast the packet at something like 30 second time intervals. (Alternatively you could have the server send a response only when a client sends a request via broadcast.)

Some options are:
DNS-SD (which seems to translate to "Apple Bonjour"): it has libraries on macOS, but it needs to install the Bonjour service on Windows. I don't know the Linux situation for this. So, it's multi-platform but you need external libraries.
UDP broadcast or multicast
Some other fancy things like Ethernet broadcast, raw sockets, ...
For your case (clients on a WiFi network), a UDP broadcast packet would suffice, it's multi-platform, and not too difficult to implement from the ground up.
Choosing this option, the two main algorithms are:
The server(s) send an "announce" broadcast packet, with clients listening to the broadcast address. Once clients receive the "announce" packet, they know about the server address. Now they can send UDP packets to the server (which will discover their addresses for sending a reply), or connect using TCP.
The client(s) send a "discover" broadcast packet, with the server(s) listening to the broadcast address. Once the server(s) receive the "discover" packet, it can reply directly to it with an "announce" UDP packet.
One or the other could be better for your application, it depends.
Please consider these arguments:
Servers usually listen to requests and send replies
A server that sends regular "announce" broadcast packets over a WiFi network, for a client that may arrive or not, wastes the network bandwidth, while a client knows exactly when it needs to poll for available servers, and stop once it's done.
As a mix of the two options, a server could send a "gratuitous announce" broadcast packet once it comes up, and then it can listen for "discover" broadcast requests from clients, replying directly to one of them using a regular UDP packet.
From here, the client can proceed as needed: send direct requests with UDP to the server, connect to a TCP address:port provided in the "announce" packet, ...
(this is the scheme I used in an application I am working on)

Related

Can a UDP multicast server send packets outside LAN?

I'm in the process of making a multiplayer game, where the players' movements are sent over the network and their positions are stored in the server. I've been told that UDP would be best since it doesn't rely on constant connection and it won't matter if the client misses a packet. The clients could be on any router, not necessarily within the server's LAN.
Is it possible to set up a server that the clients can connect to that will send all of them periodic updates of the positions of nearby objects/players?
I don't want to have to send packets to each individual client, and I heard multicasting can solve this problem, but every example I've seen only sends packets over a local network. Can I multicast past routers, and if so, how can I do that in Java? (And explain it to me like I have no idea what I'm doing, which is mostly true)
Ex.
Server has IP address 71.10.200.133
Client A has IP address 38.49.339.293
Client B has IP address 37.28.487.388
...
Client Z has IP address 43.38.382.949
Client A sends an update about the player's position to Server
Server sends update to B-Z without iterating a packet to each individual client. How do I accomplish this (if it's possible)?
Multicasts will traverse a router if and only if the router allows it. Unless you're in control of all the routers between you and your clients, the answer to your question is 'no'.
Multicast packets are broadcasts, thus they reach each node on that subnet. For you to send a multicast packet out on the web is not an effecient nor smart way of sending data.
For LAN based traffic:
Multicast is fine
But, for internet traffic I would suggest making a:
UDPClient
or
TCPClient
for internet based traffic and possibly multicast for LAN based (to mix things up a bit).
For internet traffic: Keep in mind, clients will need to initiate the connection first since most routers (household) have a firewall blocking all NEW outside-to-in traffic. So create a socket to listen over a designated port/ports for any incoming connections and from there on use which ever method of packet broadcasting/sending you like
You do also have the option of using Multicast proxies or Layer 2 VPNs if you have the capabilities. L2TP, https://en.wikipedia.org/wiki/Layer_2_Tunneling_Protocol
A layer 2 VPN would relay unicast and multicast packets.
That would basically allow you to control the routers as EJP suggested above.
This questions also 3 year old so you've probably already figured a way to do it by now.

same application listening for packets from two different ip's

Can one application handle incoming UDP packets coming from two different ip's? If so, can those two connections use the same port number?
Yes, that's the point of a server in general: multiple clients can connect to the server on the given UDP port and they can all broadcast data on the same channel. The server doesn't need to have a separate socket connection for each client, instead it just broadcasts data via its socket connection to the same channel that the clients broadcast.
UDP is a little like sitting in a room where everybody yells at each other, while TCP is like talking on the phone with multiple people at the same time.

UDP for multiplayer game

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.

Emulating accept() for UDP (timing-issue in setting up demultiplexed UDP sockets)

For an UDP server architecture that will have long-lived connections, one architecture is to have one socket that listens to all incoming UDP traffic, and then create separate sockets for each connection using connect() to set the remote address. My question is whether it is possible to do this atomically similar to what accept() does for TCP.
The reason for creating a separate socket and using connect() is that this makes it easy to spread the packet-processing across multiple threads, and also make it easier to have the socket directly associated with the data structures that are needed for processing.
The demultiplexing logic in the networking stack will route the incoming packets to the most specific socket.
Now my question is basically what happens when one wants to emulate accept() for UDP like this:
Use select() with a fd-set that includes the UDP server-socket.
Then read a packet from the UDP server-socket.
Then create a new UDP socket which is then connect()ed to the remote address
I call select() with a fd-set that includes both sockets.
What is returned?
given that a packet arrives to the OS somewhere between 1 and 3.
Will the packet be demultiplexed to the UDP server-socket, or will it be demultiplexed to the more specific socket created in 3. That is, at what point does demultiplexing take place? When the packet arrives, or must it happen "as if" it arrived at point 4?
Follow-up question in case the above does not work: What's the best way to do this?
I see that this discussion is from 2009, but since it keeps popping up when I search, I thought I should share my approach. Both to get some feedback and because I am curios about how the author of the question solved the problem.
The way I chose emulate UDP-accept was a combination of number one and two in nik's answer. I have a root thread which listens on a given socket. I have chosen to use TCP for simplicity, but changing this socket to UDP is not very hard. When a client wants to "connect" to my server using UDP, it first connects to the TCP socket and requests a new connection.
The root thread then proceeds by creating a UDP socket, binds it to a local interface, does connect and sets up data structures. This file descriptor is then passed to the thread that will be responsible for the connection. The IP/port information of the new UDP socket is passed back to the client, which creates a new UDP socket and sends data to the provided IP/port.
This approach works well for my use, but the additional steps for setting up a flow introduces an overhead. In some cases, this overhead might not be acceptable.
I found this question after asking it myself here...
UDP server and connected sockets
Since connect() is available for UDP to specify the peer address, I wonder why accept() wasn't made available to effectively complete the connected UDP session from the server side. It could even move the datagram (and any others from the same client) that triggered the accept() over to the new descriptor.
This would enable better server scalability (see the rationale behind SO_REUSEPORT for more background), as well as reliable DTLS authentication.
This will not work.
You have two simple options.
Create a multi-threaded program that has a 'root' thread listening on the UDP socket and 'dispatching' received packets to the correct thread based on the source. This is because you want to segregate processing by source.
Extend your protocol so the the sources accept an incoming connection on some fixed port and then continue with the protocol communication. In this case you would let the source request on the standard UDP port (of your choice), then your end will respond from a new UDP socket to the sources' UDP port. This way you have initiated a new UDP path from your end backwards to the known UDP port of each source. That way you have different UDP sockets at your end.

UDP Response

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/