I want to know the technical reasons why the lift webframework has high performance and scalability? I know it uses scala, which has an actor library, but according to the install instructions it default configuration is with jetty. So does it use the actor library to scale?
Now is the scalability built right out of the box. Just add additional servers and nodes and it will automatically scale, is that how it works? Can it handle 500000+ concurrent connections with supporting servers.
I am trying to create a web services framework for the enterprise level, that can beat what is out there and is easy to scale, configurable, and maintainable. My definition of scaling is just adding more servers and you should be able to accommodate the extra load.
Thanks
Lift's approach to scalability is within a single machine. Scaling across machines is a larger, tougher topic. The short answer there is: Scala and Lift don't do anything to either help or hinder horizontal scaling.
As far as actors within a single machine, Lift achieves better scalability because a single instance can handle more concurrent requests than most other servers. To explain, I first have to point out the flaws in the classic thread-per-request handling model. Bear with me, this is going to require some explanation.
A typical framework uses a thread to service a page request. When the client connects, the framework assigns a thread out of a pool. That thread then does three things: it reads the request from a socket; it does some computation (potentially involving I/O to the database); and it sends a response out on the socket. At pretty much every step, the thread will end up blocking for some time. When reading the request, it can block while waiting for the network. When doing the computation, it can block on disk or network I/O. It can also block while waiting for the database. Finally, while sending the response, it can block if the client receives data slowly and TCP windows get filled up. Overall, the thread might spend 30 - 90% of it's time blocked. It spends 100% of its time, however, on that one request.
A JVM can only support so many threads before it really slows down. Thread scheduling, contention for shared-memory entities (like connection pools and monitors), and native OS limits all impose restrictions on how many threads a JVM can create.
Well, if the JVM is limited in its maximum number of threads, and the number of threads determines how many concurrent requests a server can handle, then the number of concurrent requests will be determined by the number of threads.
(There are other issues that can impose lower limits---GC thrashing, for example. Threads are a fundamental limiting factor, but not the only one!)
Lift decouples thread from requests. In Lift, a request does not tie up a thread. Rather, a thread does an action (like reading the request), then sends a message to an actor. Actors are an important part of the story, because they are scheduled via "lightweight" threads. A pool of threads gets used to process messages within actors. It's important to avoid blocking operations inside of actors, so these threads get returned to the pool rapidly. (Note that this pool isn't visible to the application, it's part of Scala's support for actors.) A request that's currently blocked on database or disk I/O, for example, doesn't keep a request-handling thread occupied. The request handling thread is available, almost immediately, to receive more connections.
This method for decoupling requests from threads allows a Lift server to have many more concurrent requests than a thread-per-request server. (I'd also like to point out that the Grizzly library supports a similar approach without actors.) More concurrent requests means that a single Lift server can support more users than a regular Java EE server.
at mtnyguard
"Scala and Lift don't do anything to either help or hinder horizontal scaling"
Ain't quite right. Lift is highly statefull framework. For example if a user requests a form, then he can only post the request to the same machine where the form came from, because the form processeing action is saved in the server state.
And this is actualy a thing which hinders scalability in a way, because this behaviour is inconistent to the shared nothing architecture.
No doubt that lift is highly performant but perfomance and scalability are two different things. So if you want to scale horizontaly with lift you have to define sticky sessions on the loadbalancer which will redirect a user during a session to the same machine.
Jetty maybe the point of entry, but the actor ends up servicing the request, I suggest having a look at the twitter-esque example, 'skitter' to see how you would be able to create a very scalable service. IIRC, this is one of the things that made the twitter people take notice.
I really like #dre's reply as he correctly states the statefulness of lift being a potential problem for horizontal scalability.
The problem -
Instead of me describing the whole thing again, check out the discussion (Not the content) on this post. http://javasmith.blogspot.com/2010/02/automagically-cluster-web-sessions-in.html
Solution would be as #dre said sticky session configuration on load balancer on the front and adding more instances. But since request handling in lift is done in thread + actor combination you can expect one instance handle more requests than normal frameworks. This would give an edge over having sticky sessions in other frameworks. i.e. Individual instance's capacity to process more may help you to scale
you have Akka lift integration which would be another advantage in this.
Related
If I have a server running on my machine, and several clients running on other networks, what are some concepts of testing for synchronicity between them? How would I know when a client goes out-of-sync?
I'm particularly interested in how network programmers in the field of game design do this (or just any continuous network exchange application), where realtime synchronicity would be a commonly vital aspect of success.
I can see how this may be easily achieved on LAN via side-by-side comparisons on separate machines... but once you branch out the scenario to include clients from foreign networks, I'm just not sure how it can be done without clogging up your messaging system with debug information, and therefore effectively changing the way that synchronicity would result without that debug info being passed over the network.
So what are some ways that people get around this issue?
For example, do they simply induce/simulate latency on the local network before launching to foreign networks, and then hope for the best? I'm hoping there are some more concrete solutions, but this is what I'm doing in the meantime...
When you say synchronized, I believe you are talking about network latency. Meaning, that a client on a local network may get its gaming information sooner than a client on the other side of the country. Correct?
If so, then I'm sure you can look for books or papers that cover this kind of topic, but I can give you at least one way to detect this latency and provide a way to manage it.
To detect latency, your server can use a type of trace route program to determine how long it takes for data to reach each client. A common Linux program example can be found here http://linux.about.com/library/cmd/blcmdl8_traceroute.htm. While the server is handling client data, it can also continuously collect the latency statistics and provide the data to the clients. For example, the server can update each client on its own network latency and what the longest latency is for the group of clients that are playing each other in a game.
The clients can then use the latency differences to determine when they should process the data they receive from the server. For example, a client is told by the server that its network latency is 50 milliseconds and the maximum latency for its group it 300 milliseconds. The client then knows to wait 250 milliseconds before processing game data from the server. That way, each client processes game data from the server at approximately the same time.
There are many other (and probably better) ways to handle this situation, but that should get you started in the right direction.
Background: I'm running a webservice in which each request involves a fair amount of computations (up to 10 seconds on a quadcore machine).
Each request can be broken down to about 150 independent (and equally small) subtasks.
What I'm after: l'm looking for a hosting service that allows me to serve these kinds of requests efficiently in a scalable manner.
What I've considered: I've looked into Google App Engine and Rackspace.
It seems to me as if GAE is intended for simple requests, requiering litte resources to process. Problem with something like Rackspace is that I can't tell in advance how many vCPUs I may need (and even if I knew how big future spikes would be, I don't want to sit with, say, 40 servers idling the rest of the time)
Questions:
Would it be possible to use GAE in the following way:
For each request, split it up into 150 subtasks
Process all subtasks independently by doing 150 concurrent HTTP requests to the same webapp (but through a differrnt method)
Collect the results from the "subresults" and return a response to the original request.
Is there any possibility that Map Reduce for GAE could be of any help?
Is there any other service better suited for this task?
Yes, this is possible. The usual way would be to use Task Queue, possibly via DeferredTask helper class.
1.3 Normal web requests (to frontend instances) are limited to 30s, so doing this in synchronous way is not guaranteed to succeed. Also note that instances are artificially limited to do 10 parallel requests (if multithreading is enabled).
Yes, this is a job for map reduce. But note that map reduce is async - you give it tasks to do and it will be done sometime in the future.
Given the processing you need you might want to look at GAE backends (they are long running with multithrading and come in different sizes). If you need even more processing power, then you might want to look at Compute Engine.
Unless all of these 150 subtasks are read-only activities, trying to run them all in a single thread is just not safe. Web requests are unreliable - people can cancel, hit refresh if it takes too long, close windows in the middle, or just time out due to network issues. The background HTTP requests, likewise, can have a whole mess of problems. The standard solution is to have your front-end code simply build a list of things that need to be done, so it can get back to the user quickly, and have a back-end 'worker' process handle the (potentially unreliable) subtasks. Depending on what your application is doing, you might bounce the user to a "working" screen (like searching for airfare) where they can safely wait for the results to come up, or it might just be stuffed away as a "pending" job (like ordering something from Amazon).
There's countless different ways to handle this basic workflow. If you stick with Google App Engine, they have a "task queue" as part of the platform - providing a simple mechanisms for creating & dispatching background tasks. If you go with Rackspace, their cloud offering is less of a unified platform so you'll have to either roll your own queue or get one to plug into your setup.
When should I use Actors vs. Remote Actors in Akka?
I understand that both can scale a machine up, but only remote actors can scale out, so is there any practical production use of the normal Actor?
If a remote actor only has a minor initial setup overhead and does not have any other major overhead to that of a normal Actor, then I would think that using a Remote Actor would be the standard, since it can scale up and out with ease. Even if there is never a need to scale production code out, it would be nice to have the option (if it doesn't come with baggage).
Any insight on when to use an Actor vs. Remote Actor would be much appreciated.
Remote Actors cannot scale up, they are only remote references to a local actor on another machine.
For Akka 2.0 we will introduce clustered actors, which will allow you to write an Akka application and scale it up only using config.
Regular Actors can be used in sending out messages in local project.
As for the Remote Actors, you can used it in sending out messages to dependent projects that are connected to the project sending out the message.
Please refer here for the Remote Akka Actors
http://doc.akka.io/docs/akka/snapshot/scala/remoting.html
The question asks "If a remote actor only has a minor initial setup overhead and does not have any other major overhead then I would think that using a Remote Actor would be the standard". Yet the Fallacies of distributed computing make the point that it is a design error to assume that remoting with any technology has no overhead. You have the overhead of copying the messages to bytes and transmitting it across the network interface. You also have all the complexity of different processes being up, down, stalled or unreachable and of the network having hiccups leading to lost, duplicated or reordered messages.
This great article has real world examples of weird network errors which make remoting hard to make bullet proof. The Akka project lead Roland Kuln in his free video course about akka says that in his experience for every 1T of network messages being sent he sees a corruption. Notes on Distributed Systems for Young Bloods says "distributed systems tend to need actual, not simulated, distribution to flush out their bugs" so even good unit tests wont make for a perfect system. There is lots of advice that remoting is not "free" but hard work to get perfect.
If you need to use remoting for availability, or to move to huge scale, then note that akka does at-least-once delivery with possible duplication. So you must ensure that duplicated messages don't create bad results.
The moment you start to use remoting you have a distributed system which creates challenges which are discussed in Distributed systems for fun and profit. Unless you are doing very simply things like stateless calculators that are idempotent to duplicated messages things get tricky. One of assignments on that akka video course at the link above is to make a replicated key-value store which can deal with lost messages by writing the logic yourself. Its far from being an easy assignment. State distributed across different processes gets very hard, actors encapsulate state, therefore distributing actors can get very hard, depending on the consistency and availability requirements of the system you are building.
This all implies that if you can avoid remoting and achieve what you need to achieve then you would be wise to avoid it. If you do need remoting then Akka makes it easy due to its location transparency. So whilst its a great toolbox to take with you on the job; you should double check if the job needs all the tools or only the simplest ones in the box.
I like to know the server (TCP based) architecture to support large scale of clients(at least10K) to implement Fix server. My points are
How we design it.
How to listen on the open port? Use select or poll or any other function.
How to process the response of the client? On large scale we cannot create the one thread for each client.
Should the processing of response is in the different executable and share the request and response to the server executable through IPC.
There is much more on it. I would appreciate if anyone explains it or provide any link.
Thanks
An excellent resource for information on this topic is The C10K problem. Although the dimensions there seem a little old, the techniques are still applicable today.
The architecture depends on what you want to do with the clients incoming data. My guess is that for every incoming message you would perform some computations and probably also return a response.
In that case I would create 1 main listener thread that receives all the incoming messages (Actually, if your hardware has more than 1 physical network device, I would use a listener thread per device and make sure each one is listening to a specific device).
Get the number of CPUs that you have on your machine and create worker threads for each CPU and bind them each thread to one cpu (Maybe number of working thread should be num_of_cpu-1, to leave an availalbe cpu for the listener and dispatcher).
Each thread has a queue and semaphore, the main listener thread just push the incoming data into those queues. There are many way to perform load balancing (Will talk about it later).
Each working thread just works on the requests given to it, and put the response on another queue that is read by the dispatcher.
The dispatcher - there are 2 options here, use a thread for dispatcher (or thread per network device as for listeners), or have the dispatcher actually be the same thread as the listener.
There is some advantage to put them both on the same thread, since it makes it easier to detect lost socket connection and use the same fds for both reading and writing without thread synchronization. However, it could be that using 2 different threads would give better performance, it need to be tested.
Note about load balancing:
This is a topic of its own.
The simplest thing is to use 1 queue for all working threads, but the problem is that they have to lock in order to pop items and the locking can damage performance. (But you get the most balanced load).
Another quite simple approach would be to have a private queue for every worker and perform round-robin when inserting. After every X cycles check the size of all the queues. If some queues are much larger than others then leave them out for the next X cycles and then recheck them again. This is not the best approach, but a simple one to implement and gives some load balancing while no locking is needed.
By the way - There is a way to implement queue between 2 threads without blocking - but this is also another topic.
I hope it helps,
Guy
If the client and server are on a secure network then the security aspect is to be minimal - to the extent that the transfers are encrypted. If the clients and the server are not on a secure network - you first want the server and client to authenticate each other and then initiate encrypted data transfer. For data transfer, server-side authentication should suffice. At the end of this authentication use the session key to generate encrypted data stream (symmetric). consider using TFTP it is simple to implement and scales reasonably well.
I already wrote here about the http chat server I want to create: Alternative http port?
This http server should stream text to every user in the same chat room on the website. The browser will stay connected and wait for further html code. (yes that works, the browser won't reject the connection).
I got a new question: Because this chat server doesn't need to receive information from the client, it's not necessary to listen to the client after the server sent its first response. New chat messages will be send to the server on a new connection.
So I can open 2 threads, one waiting for new clients (or new messages) and one for the html streaming.
Is this a good idea or should I use one thread per client? I don't think it's good to have one thread/client when there are many chat users online, since the server should handle multiple different chats with their own rooms.
3 posibilities:
1. One thread for all clients, send text to each client successive - there shouldn't be much lag since it's only text
this will be like: user1.send("text");user2.send("text"),...
2. One thread per chat or chatroom
3. One thread per chat user - ... many...
Thank you, I haven't done much with sockets yet ;).
Right now, you seem to be thinking in terms of a given thread always carrying out a given (type of) task. While that basic design can make sense, to produce a scalable server like this, it generally doesn't work very well.
Often a slightly more abstract viewpoint works out better: you have tasks that need to get done, and threads that do those tasks -- but a thread doesn't really "care" about what task it executes.
With this viewpoint, you simply need to create some sort of data structure that describes each task that needs to be done. When you have a task you want done, you fill in a data structure to describe the task, and hand it off to get done. Somewhere, there are some threads that do the tasks.
In this case, the exact number of threads becomes mostly irrelevant -- it's something you can (and do) adjust to fit the number of CPU cores available, the type of tasks, and so on, not something that affects the basic design of the program.
I think easiest pattern for this simple app is to have pool of threads and then for each client pick available thread or make it wait until one becomes available.
If you want serious understanding of http server architecture concepts google following:
apache architecture
nginx architecture