Is it possible to write a gatling script which connects to WebSocket and then performs actions (e.g. new HTTP requests) when certain messages are received (preferably with out of the box support for STOMP messages, but I could probably workaround this).
In other words, "real clients" should be simulated as best as possible. The real clients (angular applications) would load data based on certain WebSocket messages.
I'm looking for something similar to (pseude code, does not work):
val scn = scenario("WebSocket")
.exec(http("Home").get("/"))
.pause(1)
.exec(session => session.set("id", "Steph" + session.userId))
.exec(http("Login").get("/room?username=${id}"))
.pause(1)
.exec(
ws("Connect WS")
.open("/room/chat?username=${id}")
// ---------------------------------------------------------------------
// Is it possible to trigger/exec something (http call, full scenario)
// as reaction to a STOMP/WebSocket message?
.onMessage(check(perform some check, maybe regex?).as("idFromPayload"))
.exec(http("STOMP reaction").get("/entity/${idFromPayload}"))
// ---------------------------------------------------------------------
)
.exec(ws("Close WS").close)
// ideally, closing the websocket should only be done once the full scenario is over
// (or never, until the script terminates in "forever" scenarios)
Is this currently possible? If not, is this planned for future versions of gatling?
To the best of my knowledge, this is not possible with Gatling.
I have since switched to k6 which supports writing and executing test scripts with this kind of logic/behavior.
Related
I have a simple WebSocket application, which is based on Akka HTTP/Reactive streams, like this https://github.com/calvinlfer/akka-http-streaming-response-examples/blob/master/src/main/scala/com/experiments/calvin/ws/WebSocketRoutes.scala#L82.
In other words, I have Sink, Source (which is produced from Publisher), and the Flow:
Flow.fromSinkAndSource(incomingMessages, outgoingMessages)
When I produce more, than 30 messages per second to the client, Akka closes a connection.
I cannot understand, where is a setting, which configure this behaviour. I know about OverflowStrategy, but I don't explicitly configure it.
It seems, that I have OverflowStrategy.fail(), or my problem looks like it.
You can tune Internal buffers.
There are two ways, how to do it:
1) application.conf:
akka.stream.materializer.max-input-buffer-size = 1024
2) You can configure it explicitly for your Flow:
Flow.fromSinkAndSource(incomingMessages, outgoingMessages)
.addAttributes(Attributes.inputBuffer(initial = 1, max = 1024))
We are using Akka HTTP to handle our web socket connections using the akka streams API. We are using a Flow that pipes the incoming messages to a "connection actor". A snippet of the code is below:
val connection = system.actorOf(ConnectionActor.props())
val in = Flow[Message]
.to(Sink.actorRef[Message](connection, WebSocketClosed))
val out = Source
.actorRef[Message](500, OverflowStrategy.fail)
.mapMaterializedValue(ws => connection ! WebSocketOpened(ws))
Flow.fromSinkAndSource(in, out)
When the web socket is closed, the connection actor is sent the "WebSocketClose" message and we clean up internal resources. We now have the requirement to know what the reason for closing the connection was according to the standard WebSocket CloseEvent codes.
Is there a way to get the close code from Akka HTTP and send it on to the connection actor so it can take the appropriate action?
I was able to handle client (browser) error code in an akka-http 10.2.6 server.
My use case was to pipe incoming messages to a Sink created by ActorSink.actorRef[T](). When creating the sink, 2 callbacks onCompleteMessage onFailureMessage can be set to converts normal WebSocket close (code=1000) or error to our custom message types.
I suppose that client close/error maps to Flow complete/failure, that means other sinks should be able to handle close/error in a similar way.
my code
`
As it turns out, this is not presently possible in Akka HTTP. See the following GitHub issue:
https://github.com/akka/akka-http/issues/2458
It looks as though this will need to be addressed before this is possible.
I need tips to build an infrastructe to send 1000 simultaneous voice calls (automated IVR calls with voicexml). Up to now i used asterisk with voiceglue but now i have performance issues.
The infrasturcture was like this:
the asterisk pulls request from queue
the queue consumer create a call file
when the call ends, call file is read and status is sent to the application server
To be honest, i am asking for tips to implement an infrastructure like callfire[1] or voxeo[2]?
[1]https://www.callfire.com/
[2]http://voxeo.com/
you can go with voxeo prophecy (http://voxeo.com/prophecy/) one of the good server which have the capability of making simultaneous voice calls
Note: The requirement which your are expecting to do will not only possible with voxeo prophecy it should also depend the web server like Tomcat, IIS e.t.c in case if you dealing with databases like Sql , Oracle e.t.c
Please do refer to know the architecture
http://www.alpensoftware.com/define_VoiceOverview.html
CallFire's API has a CreateBroadcast method where you can throw up an IVR using their XML in seconds. You can read up on the documentation here:
https://www.callfire.com/api-documentation/rest/version/1.1#!/broadcast
CallFire also offers a PHP-SDK, hosted on Github, with examples of how to do this. The SDK is minimal setup and allows you to easily tap into the APIs robust functionality. Version 1.1 can be found here, with instructions on how to get started: https://github.com/CallFire/CallFire-PHP-SDK
The method call might look something like this. Note the required dependencies.
<?php
use CallFire\Api\Rest\Request;
use CallFire\Api\Rest\Response;
require 'vendor/autoload.php';
$dialplan = <<<DIALPLAN
<dialplan><play type="tts">Congratulations! You have successfully configured a CallFire I V R.</play></dialplan>
DIALPLAN;
$client = CallFire\Api\Client::Rest("<api-login>", "<api-password>", "Broadcast");
$request = new Request\CreateBroadcast;
$request->setName('My CallFire Broadcast');
$request->setType('IVR');
$request->setFrom('15551231234'); // A valid Caller ID number
$request->setDialplanXml($dialplan);
$response = $client->CreateBroadcast($request);
$result = $client::response($response);
if($result instanceof Response\ResourceReference) {
// Success
}
You can read this:
http://www.voip-info.org/wiki/view/Asterisk+auto-dial+out
Main tip: you WILL have ALOT of issues. If you are not expert with at least 5 years development experience with asterisk, you have use already developed dialling cores or hire guru. There are no opensource core that can do more then 300 calls on single server.
You can't do 1000 calls on single asterisk in app developed by "just nice developer". It will just not work.
Task of create dialling core for 1000 calls is "rocket science" type task. It require very special dialling core, very special server/server tunning and very specialized dialler with pre-planning.
1000 calls will result 23Mbit to 80Mbit bandwidth usage with SMALL packets, even that single fact can result you be banned on your hosting and require linux network stack be tunned.
You can use ICTBroadcast REST API to integerate your application with reknown autodialer , please visit following link for more detail
http://www.ictbroadcast.com/news/using-rest-api-integerate-ictbroadcast--third-party-application-autodialer
ICTBroadcast is based on asterisk communication engine
I've already done this for phone validation and for phone message broadcasting using Asterisk and Freeswitch. I would go with Freeswitch and xmlrpc:
https://wiki.freeswitch.org/wiki/Freeswitch_XML-RPC
streaming data out of play, is quite easy.
here's a quick example of how I intend to do it (please let me know if i'm doing it wrong):
def getRandomStream = Action { implicit req =>
import scala.util.Random
import scala.concurrent.{blocking, ExecutionContext}
import ExecutionContext.Implicits.global
def getSomeRandomFutures: List[Future[String]] = {
for {
i <- (1 to 10).toList
r = Random.nextInt(30000)
} yield Future {
blocking {
Thread.sleep(r)
}
s"after $r ms. index: $i.\n"
}
}
val enumerator = Concurrent.unicast[Array[Byte]] {
(channel: Concurrent.Channel[Array[Byte]]) => {
getSomeRandomFutures.foreach {
_.onComplete {
case Success(x: String) => channel.push(x.getBytes("utf-8"))
case Failure(t) => channel.push(t.getMessage)
}
}
//following future will close the connection
Future {
blocking {
Thread.sleep(30000)
}
}.onComplete {
case Success(_) => channel.eofAndEnd()
case Failure(t) => channel.end(t)
}
}
}
new Status(200).chunked(enumerator).as("text/plain;charset=UTF-8")
}
now, if you get served by this action, you'll get something like:
after 1757 ms. index: 10.
after 3772 ms. index: 3.
after 4282 ms. index: 6.
after 4788 ms. index: 8.
after 10842 ms. index: 7.
after 12225 ms. index: 4.
after 14085 ms. index: 9.
after 17110 ms. index: 1.
after 21213 ms. index: 2.
after 21516 ms. index: 5.
where every line is received after the random time has passed.
now, imagine I want to preserve this simple example when streaming data from the server to the client, but I also want to support full streaming of data from the client to the server.
So, lets say i'm implementing a new BodyParser that parses the input into a List[Future[String]]. this means, that now, my Action could look like something like this:
def getParsedStream = Action(myBodyParser) { implicit req =>
val xs: List[Future[String]] = req.body
val enumerator = Concurrent.unicast[Array[Byte]] {
(channel: Concurrent.Channel[Array[Byte]]) => {
xs.foreach {
_.onComplete {
case Success(x: String) => channel.push(x.getBytes("utf-8"))
case Failure(t) => channel.push(t.getMessage)
}
}
//again, following future will close the connection
Future.sequence(xs).onComplete {
case Success(_) => channel.eofAndEnd()
case Failure(t) => channel.end(t)
}
}
}
new Status(200).chunked(enumerator).as("text/plain;charset=UTF-8")
}
but this is still not what I wanted to achieve. in this case, I’ll get the body from the request only after the request was finished, and all the data was uploaded to the server. but I want to start serving request as I go. a simple demonstration, would be to echo any received line back to the user, while keeping the connection alive.
so here's my current thoughts:
what if my BodyParser would return an Enumerator[String] instead of List[Future[String]]?
in this case, I could simply do the following:
def getParsedStream = Action(myBodyParser) { implicit req =>
new Status(200).chunked(req.body).as("text/plain;charset=UTF-8")
}
so now, i'm facing the problem of how to implement such a BodyParser.
being more precise as to what exactly I need, well:
I need to receive chunks of data to parse as a string, where every string ends in a newline \n (may contain multiple lines though...). every "chunk of lines" would be processed by some (irrelevant to this question) computation, which would yield a String, or better, a Future[String], since this computation may take some time. the resulted strings of this computation, should be sent to the user as they are ready, much like the random example above. and this should happen simultaneously while more data is being sent.
I have looked into several resources trying to achieve it, but was unsuccessful so far.
e.g. scalaQuery play iteratees -> it seems like this guy is doing something similar to what I want to do, but I couldn't translate it into a usable example. (and the differences from play2.0 to play2.2 API doesn't help...)
So, to sum it up: Is this the right approach (considering I don't want to use WebSockets)? and if so, how do I implement such a BodyParser?
EDIT:
I have just stumble upon a note on the play documentation regarding this issue, saying:
Note: It is also possible to achieve the same kind of live
communication the other way around by using an infinite HTTP request
handled by a custom BodyParser that receives chunks of input data, but
that is far more complicated.
so, i'm not giving up, now that I know for sure this is achievable.
What you want to do isn't quite possible in Play.
The problem is that Play can't start sending a response until it has completely received the request. So you can either receive the request in its entirety and then send a response, as you have been doing, or you can process requests as you receive them (in a custom BodyParser), but you still can't reply until you've received the request in its entirety (which is what the note in the documentation was alluding to - although you can send a response in a different connection).
To see why, note that an Action is fundamentally a (RequestHeader) => Iteratee[Array[Byte], SimpleResult]. At any time, an Iteratee is in one of three states - Done, Cont, or Error. It can only accept more data if it's in the Cont state, but it can only return a value when it's in the Done state. Since that return value is a SimpleResult (i.e, our response), this means there's a hard cut off from receiving data to sending data.
According to this answer, the HTTP standard does allow a response before the request is complete, but most browsers don't honor the spec, and in any case Play doesn't support it, as explained above.
The simplest way to implement full-duplex communication in Play is with WebSockets, but we've ruled that out. If server resource usage is the main reason for the change, you could try parsing your data with play.api.mvc.BodyParsers.parse.temporaryFile, which will save the data to a temporary file, or play.api.mvc.BodyParsers.parse.rawBuffer, which will overflow to a temporary file if the request is too large.
Otherwise, I can't see a sane way to do this using Play, so you may want to look at using another web server.
"Streaming data in and out simultaneously on a single HTTP connection in play"
I haven't finished reading all of your question, nor the code, but what you're asking to do isn't available in HTTP. That has nothing to do with Play.
When you make a web request, you open a socket to a web server and send "GET /file.html HTTP/1.1\n[optional headers]\n[more headers]\n\n"
You get a response after (and only after) you have completed your request (optionally including a request body as part of the request). When and only when the request and response are finished, in HTTP 1.1 (but not 1.0) you can make a new request on the same socket (in http 1.0 you open a new socket).
It's possible for the response to "hang" ... this is how web chats work. The server just sits there, hanging onto the open socket, not sending a response until someone sends you a message. The persistent connection to the web server eventually provides a response when/if you receive a chat message.
Similarly, the request can "hang." You can start to send your request data to the server, wait a bit, and then complete the request when you receive additional user input. This mechanism provides better performance than continually creating new http requests on each user input. A server can interpret this stream of data as a stream of distinct inputs, even though that wasn't necessarily the initial intention of the HTTP spec.
HTTP does not support a mechanism to receive part of a request, then send part of a response, then receive more of a request. It's just not in the spec. Once you've begun to receive a response, the only way to send additional information to the server is to use another HTTP request. You can use one that's already open in parallel, or you can open a new one, or you can complete the first request/response and issue an additional request on the same socket (in 1.1).
If you must have asynchronous io on a single socket connection, you might want to consider a different protocol other than HTTP.
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.