If we want to broadcast information from a socket, we need to enable SocketOptions.SO_BROADCAST. However, I don't understand why that is necessary.
My understanding is we set the packet with a broadcast address, just the same way as set a unicast address. Then we just need to send it through a regular socket. If its a UDP socket, then a UDP header will be added to that packet, and then an IP header containing the receiver's IP address (in this case is the broadcast address in the form of 192.168.255.255), and then a MAC address (FF:FF:FF:FF) is added.
I think the router will get the packet and perform the broadcast. I don't understand why we need to set the socket attribute to SO_BROADCAST.
"Socket semantics require that an application set the SO_BROADCAST option on before attempting to send a datagram to a base or broadcast address. This protects the application from accidentally sending a datagram to many systems."
Source
In most cases these calls just call the same thing in the OS. This is likely to be a restriction of your OS, not Java.
To complement Jeremy Friesner's answer, here is a good wording I found about this: https://notes.shichao.io/unp/ch7/
Since an application must set this socket option before sending a broadcast datagram, it prevents a process from sending a broadcast when the application was never designed to broadcast. For example, a UDP application might take the destination IP address as a command-line argument, but the application never intended for a user to type in a broadcast address. Rather than forcing the application to try to determine if a given address is a broadcast address or not, the test is in the kernel: If the destination address is a broadcast address and this socket option is not set, EACCES is returned.
In other words, by setting this option, the application is saying that it is designed to support the broadcast use-case and is ready to handle broadcast IP addresses provided by user.
Since not all apps were designed for broadcast, the option is disable by default.
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'm on the client side. There're multiple network interfaces. How can I let different processes use different network interfaces to communicate? Since I want to connect to the same server, routing seems not working here. Also, connect() doesn't have arguments to specify local address or interface as bind() does.
If your goal is to increase bandwidth to the server by using multiple network interfaces in parallel, then that's probably not something you can (or should) do at the application level. Instead, you should study up on Link Aggregation and then configure your computer and networking stack to use that. Once that is working properly, you will get the parallelization-speedup you want automatically, without the client application having to do anything special to enable it.
"The bind() system call is frequently misunderstood. It is used to
bind to a particular IP address. Only packets destined to that IP
address will be received, and any transmitted packets will carry that
IP address as their source. bind() does not control anything about the
routing of transmitted packets. So for example, if you bound to the IP
address of eth0 but you send a packet to a destination where the
kernel's best route goes out eth1, it will happily send the packet out
eth1 with the source IP address of eth0. This is perfectly valid for
TCP/IP, where packets can traverse unrelated networks on their way to
the destination."
More info e.g. here.
That's why you probably misunderstand bind() call.
The appropriate way to bind to physical topology (to some specific interface) is to use SO_BINDTODEVICE socket option. This is done by setsockopt() call.
Source Policy Routing might be helpful.
Try the following steps:
Use iptables to give packets from different process with different marks.
Use iproute2 to route packets with different marks to different table.
In different table, set the default route to different uplink.
The whole process require certain amount of understanding about linux networking.
Here is an example shows how to route all traffic for a user through one specific uplink: http://www.niftiestsoftware.com/2011/08/28/making-all-network-traffic-for-a-linux-user-use-a-specific-network-interface/
You could try follow similar approach by running different process with different user and route traffic from one user to one uplink.
Also you could let processes communicate with the server with different port and mark the traffic by port.
I have been working on a local LAN service which uses a multicast port to coordinate several machines. Each machine listens on the multicast port for instructions, and when a certain instruction is received, will send messages directly to other machines.
In other words the multicast port is used to coordinate peer-to-peer UDP messaging.
In practice this works quite well but there is a lingering issue related to correctly setting up these peer-to-peer transmissions. Basically, each machine needs to announce on the multicast port its own IP address, so that other machines know where to send messages when they wish to start a P2P transmission.
I realize that in general the idea of identifying the local IP is not necessarily sensible, but I don't see any other way-- the local receiving IP must be announced one way or another. At least I am not working on the internet, so in general I won't need to worry about NATs, just need to identify the local LAN IP. (No more than 1 hop for the multicast packets is allowed.)
I wanted to, if possible, determine the IP passively, i.e., without sending any messages.
I have been using code that calls getifaddrs(), which returns a linked list of NICs on the machine, and I scan this list for non-zero IP addresses and choose the first one.
In general this has worked okay, but we have had issues where for example a machine with both a wired and wifi connection are active, it will identify the wrong one, and the only work-around we found was to turn off the wifi.
Now, I imagine that a more reliable solution would be to send a message to the multicast telling other machines to report back with the source address of the message; that might allow to identify which IP is actually visible to the other machines on the net. Alternatively maybe even just looking at the multicast loopback message would work.
What do you think, are there any passive solutions to identify which address to use? If not, what's the best active solution?
I'm using POSIX socket API from C. Must work on Linux, OS X, Windows. (For Windows I have been using GetAdapterAddresses().)
Your question about how to get the address so you can advertise it right is looking at it from the wrong side. It's a losing proposition to try to guess what your address is. Better for the other side to detect it itself.
When a listening machine receives a message, it is probably doing do using recvfrom(2). The fifth argument is a buffer into which the kernel will store the address of the peer, if the underlying protocol offers it. Since you are using IP/UDP, the buffer should get filled in with a sockaddr_in showing the IP address of the sender.
I'd use the address on the interface I use to send the announcement multicast message -- on the wired interface announce the wired address and on the wireless interface announce the wireless address.
When all the receivers live on the wired side, they will never see the message on the wireless network.
When there is a bridge between the wired and the wireless network, add a second step in discovery for round-trip time estimation, and include a unique host ID in the announcement packet, so multiple routes to the same host can be detected and the best one chosen.
Also, it may be a good idea to add a configuration option to limit the service to certain interfaces.
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)
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/