I'm trying to re-write one of our old Servers using ZeroMQ, for now I have the following Server setup, (which works for Zmq requests):
using (var context = ZmqContext.Create())
using (var server = context.CreateSocket(SocketType.REP)) {
server.Bind("tcp://x.x.x.x:5705");
while (true) { ... }
This kind of setup works fine if I use the Zmq client library to connect context.CreateSocket(SocketType.REQ)
But unfortunately we've got a lot of legacy code that needs to connect to this server and the sockets are created using .net socket libs:
Socket = new Socket(ipAddress.AddressFamily, SocketType.Stream, ProtocolType.Tcp);
Socket.Connect(ipAddress, port);
Is there a way to write a ZeroMQ Server to accept these traditional .net socket connections?
You can achieve this using ZMQ_STREAM sockets.
Please note that since zeroMQ 4.x, the RAW router option has been deprecated for a new ZMQ_STREAM socket type, that works the same way as ROUTER + RAW.
It seems it is bound to evolve, though.
I recently tried ZMQ_STREAM sockets in version 4.0.1.
You can open one, use zmq_rcv until you receive the whole message (you have to check it is whole yourself), or zmq_msg_rcv to let ZeroMQ handle it. You will receive an identifier message part, just like the identifier you would find in ROUTER sockets, directly followed by one ONLY body part. There is no empty delimiter between them like there would be using a REQ Socket talking to a ROUTER Socket. So if you route them, be sure to add it yourself.
Beware though: if there is latency on the other end or if your message exceeds ZeroMQ ZMQ_STREAM buffers (mine are 8192 bytes long), your message can be interpreted by zeroMQ as a series of messages.
In that case, you will receive as many different ZeroMQ messages including both the identifier part and the body part, and it is your job to aggregate them, knowing that if several clients are talking to the STREAM socket, they might get mixed up. I personnally use a hash table using the binary identifier as a key, and delete the entry from the table when I know the message is complete and sent to the next node.
Sending through a ZMQ_STREAM with zmq_msg_send or zmq_send works fine as is.
You probably have to use zmq's RAW socket type (instead of REP) to connect with and read client data without zmq-specific framing.
HTTP Server in C (from Pieter's blog)
http://hintjens.com/blog:42
RAW Socket type info
https://github.com/hintjens/libzmq/commit/777c38ae32a5d1799b3275d38ff8d587c885dd55
Related
I'm trying to create a server that sets up a Unix socket and listens for clients which send/receive data. I've made a small repository to recreate the problem.
The server runs and it can receive data from the clients that connect, but I can't get the server response to be read from the client without an error on the server.
I have commented out the offending code on the client and server. Uncomment both to recreate the problem.
When the code to respond to the client is uncommented, I get this error on the server:
thread '' panicked at 'called Result::unwrap() on an Err value: Os { code: 11, kind: WouldBlock, message: "Resource temporarily unavailable" }', src/main.rs:77:42
MRE Link
Your code calls set_read_timeout to set the timeout on the socket. Its documentation states that on Unix it results in a WouldBlock error in case of timeout, which is precisely what happens to you.
As to why your client times out, the likely reason is that the server calls stream.read_to_string(&mut response), which reads the stream until end-of-file. On the other hand, your client calls write_all() followed by flush(), and (after uncommenting the offending code) attempts to read the response. But the attempt to read the response means that the stream is not closed, so the server will wait for EOF, and you have a deadlock on your hands. Note that none of this is specific to Rust; you would have the exact same issue in C++ or Python.
To fix the issue, you need to use a protocol in your communication. A very simple protocol could consist of first sending the message size (in a fixed format, perhaps 4 bytes in length) and only then the actual message. The code that reads from the stream would do the same: first read the message size and then the message itself. Even better than inventing your own protocol would be to use an existing one, e.g. to exchange messages using serde.
I am interested in learning Vapor, so I decided to work on a website that displays government issued weather alerts. Alert distribution is done via a TCP/IP data stream (streaming1.naad-adna.pelmorex.com port 8080).
What I have in mind is to use IBM's BlueSocket (https://github.com/IBM-Swift/BlueSocket) to create a socket, though after this point, I gave it a bit of thought but was unable to come to a conclusion on what the next steps would be.
Alerts are streamed over the data stream, so I am aware the socket would need to be opened and listened on but wasn't able to get to much past that.
A few things with the data stream are that the start and end of an alert is detected using the start and end tags of the XML document (alert and /alert). There are no special or proprietary headers added to the data, it's only raw XML. I know some alerts also include an XML declaration so I assume the encoding should be taken into account if the declaration is available.
I was then thinking of using XMLParser to parse the XML and use the data I am interested in from the alert.
So really, the main thing I am struggling with is, when the socket is open, what would be the method to listen to it, determine the start and end of the alert and then pass that XML alert for processing.
I would appreciate any input, I am also not restricted to BlueSocket so if there is a better option for what I am trying to achieve, I would be more than open to it.
So really, the main thing I am struggling with is, when the socket is
open, what would be the method to listen to it, determine the start
and end of the alert and then pass that XML alert for processing.
The method that you should use is read(into data: inout Data). It stores any available data that the server has sent into data. There are a few reasons for this method to fail, such as the connection disconnecting.
Here's an example of how to use it:
import Foundation
import Socket
let s = try Socket.create()
try s.connect(to: "streaming1.naad-adna.pelmorex.com", port: 8080)
while true {
if try Socket.wait(for: [s], timeout: 0, waitForever: true) != nil {
var alert = Data()
try s.read(into: &alert)
if let message = String(data: alert, encoding: .ascii) {
print(message)
}
}
}
s.close()
First create the socket. The default is what we want, a IPv4 TCP Stream.
Second connect() to the server using the hostname and port. Without this step, the socket isn't connected and cannot receive or send any data.
wait() until hostname has sent us some data. It returns a list of sockets that have data available to read.
read() the data, decode it and print it. By default this call will block if there is no data available on the socket.
close() the socket. This is good practice.
You might also like to consider thinking about:
non blocking sockets
error handling
streaming (a single call to read() might not give a complete alert).
I hope this answers your question.
I'm using Wireshark packet analyzer & when I filter for all "Websocket" packets I see what I am sending /receiving to the host. When I check individual packets mine always show as [MASKED], but you can 'Umask Payload' which shows the data in clear text that looks like this:
<IC sid="52ccc752-6080-4668-8f55-662020d83979" msqid="120l93l9l114l30l104"/>
However, if I 'Follow TCP stream & look at that same packet, the data shows up as encoded in some way like this:
....K#....../...y#..|...}...f...s...~...}...{G..r...kN.."G..z...r...'...'...z...d.
The problem is all Websocket packets I receive from the host come as encoded, it is NOT SSL & I can't figure out how to decode them, I have no idea what they are even encoded as (but yet my browser can decode it).
I assume that whatever method they are coming back to me as encoded data is the same method that my data is encoded when I use 'Follow TCP stream'.
Can someone please help me figure out how to decode the data the host is sending me? See host data below
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Client to server data is XORed with a mask (included in the dataframe). Some people suggest this is in order to throw off bad caching mechanisms responding to new websocket requests with server messages from older sessions. The masking makes sure that even messages containing identical data will appear differently to applications that do not understand websockets.
Also note that there are many different size options for the headers themselves.
Refer to RFC 6455 Section 5 which defines the masking/unmasking process for payloads sent from the client to the server.
https://www.rfc-editor.org/rfc/rfc6455
If you find any freeware VBA code to do the job of forming packets let me know! :-)
Is it possible to use a webserver with websockets as a wrapper to another server to pass messages from the "real server" to a web client and back?
Im curious of this as I have a game server written in Ada that has an OS-tied client. I would like to swap this client to a webclient based on Javascript, so that the game can be played in a normal browser. What can be done?
That is the purpose of websockify. It is designed to bridge between WebSocket clients and regular TCP servers. It was created as part of the noVNC which is an HTML5 VNC app that can connect to normal VNC servers. However, websockify is generic and there are now many other projects using it.
Disclaimer: I created websockify and noVNC.
You can easily accomplish this by using AWS:
http://libre.adacore.com/tools/aws/
There's support for websockets in AWS, and you can make use of it's excellent socket (AWS.Net) packages for normal socket support.
Websockets are, contrary to what some people believe, not pure sockets. The raw data is encapsulated and masked by the websocket protocol which isn't widely supported yet. That means an application which wasn't designed for it, you can't communicate with it directly via web sockets.
When you have an application which uses a protocol based on normal sockets, and you want to communicate with it with websockets, there are two options.
Either you use a websocket gateway which does the unpacking / packing of the websocket traffic and forwards it as pure socket traffic to the application. This has the advantage that you needn't modify the application, but it has the disadvantage that it also masks the real IP address of the client which might or might not be a problem for certain applications.
Or you implement websocket in your application directly. This can be done by having two different ports the server listens to, one for normal connections and one for websocket connections. Any data which is received or sent through the websocket-port is sent through your websocket implementation before sending / after receiving it, and otherwise processed by the same routines.
THe Kaazing HTML5 Gateway is a great way of bringing your TCP-based protocol to a web client. The Kaazing gateway basically takes your protocol running on top of TCP and converts it to WebSocket so you can access the protocol in the client. You would still need to write a JavaScript protocol library for the protocol that your back end uses. But if you can work with the protocol on top of TCP, then it's not hard to do it with JavaScript.
I used the following code in ruby to wrapp my sockets. The code was adapted from em-websocket-proxy. There might be some specifics for my project in it but generally switching remote_host and remote_port and connecting to localhost:3000 should set you up with a new connection to your server through a websocket.
require 'rubygems'
require 'em-websocket'
require 'sinatra/base'
require 'thin'
require 'haml'
require 'socket'
class App < Sinatra::Base
get '/' do
haml :index
end
end
class ServerConnection < EventMachine::Connection
def initialize(input, output)
super
#input = input
#output = output
#input_sid = #input.subscribe { |msg| send_data msg+ "\n" }
end
def receive_data(msg)
#output.push(msg)
end
def unbind
#input.unsubscribe(#input_sid)
end
end
# Configuration of server
options = {:remote_host => 'your-server', :remote_port => 4000}
EventMachine.run do
EventMachine::WebSocket.start(:host => '0.0.0.0', :port => 8080) do |ws|
ws.onopen {
output = EM::Channel.new
input = EM::Channel.new
output_sid = output.subscribe { |msg| ws.send msg; }
EventMachine::connect options[:remote_host], options[:remote_port], ServerConnection, input, output
ws.onmessage { |msg| input.push(msg)}
ws.onclose {
output.unsubscribe(output_sid)
}
}
end
App.run!({:port => 3000})
end
Enjoy! And ask if you have questions.
I've been having some problems with the below code that I've pieced together. All the events work as advertised however, when a client drops off-line without first disconnecting the close event doesn't get call right away. If you give it a minute or so it will eventually get called. Also, I find if I continue to send data to the client it picks up a close event faster but never right away. Lastly, if the client gracefully disconnects, the end event is called just fine.
I understand this is related to the other listen events like upgrade and ondata.
I should also state that the client is an embedded device.
client http request:
GET /demo HTTP/1.1\r\n
Host: example.com\r\n
Upgrade: Websocket\r\n
Connection: Upgrade\r\n\r\n
//nodejs server (I'm using version 6.6)
var http = require('http');
var net = require('net');
var sys = require("util");
var srv = http.createServer(function (req, res){
});
srv.on('upgrade', function(req, socket, upgradeHead) {
socket.write('HTTP/1.1 101 Web Socket Protocol Handshake\r\n' +
'Upgrade: WebSocket\r\n' +
'Connection: Upgrade\r\n' +
'\r\n\r\n');
sys.puts('upgraded');
socket.ondata = function(data, start, end) {
socket.write(data.toString('utf8', start, end), 'utf8'); // echo back
};
socket.addListener('end', function () {
sys.puts('end'); //works fine
});
socket.addListener('close', function () {
sys.puts('close'); //eventually gets here
});
});
srv.listen(3400);
Can anyone suggest a solution to pickup an immediate close event? I am trying to keep this simple without use of modules. Thanks in advance.
close event will be called once TCP socket connection is closed by one or another end with few complications of rare cases when system "not realising" that socket been already closed, but this are rare cases. As WebSockets start from HTTP request server might just keep-alive till it timeouts the socket. That involves the delay.
In your case you are trying to perform handshake and then send data back and forth, but WebSockets are a bit more complex process than that.
The handshake process requires some security procedure to validate both ends (server and client) and it is HTTP compatible headers. But different draft versions supported by different platforms and browsers do implement it in a different manner so your implementation should take this in account as well and follow official documentation on WebSockets specification based on versions you need to support.
Then sending and receiving data via WebSockets is not pure string. Actual data sent over WebSockets protocol has data-framing layer, which involves adding header to each message you send. This header has details over message you sending, masking (from client to server), length and many other things. data-framing depends on version of WebSockets again, so implementations will vary slightly.
I would encourage to use existing libraries as they already implement everything you need in nice and clean manner, and have been used extensively across commercial projects.
As your client is embedded platform, and server I assume is node.js as well, it is easy to use same library on both ends.
Best suit here would be ws - actual pure WebSockets.
Socket.IO is not good for your case, as it is much more complex and heavy library that has multiple list of protocols support with fallbacks and have some abstraction that might be not what you are looking for.