Server emulator for existing software - networking

For example we have chat client application executable (and we can change server Endpoint).
How I can analyze sockets, packets send to server and back? So I can write server emulator for that client?
This is just example. I know this is very general question. But I need general answer. What techniques you can suggest? What tools? Any tutorials or books?
This is for educational purpose and I have no intention to violate any law.
Edited: Basically I want to get protocol that client/server communicate.

If you consider about writing a server emulator, you must know what to emulate, and therefore you must know the details of the protocol. So I doubt that approach can help to discover the unkown features.
I think that the Wireshark protocol analyser can help you to see all the network dialog between the server and the client. You do not have to write a custom server, just spy the actual exchanges :o)

Related

a webRtc Clearing View

i'm a newbie to webRTC and their is some stuff that i did not get if it was possible i would like an answer to those question and i quiet think that it will be a good reference to all the other guys over the web .
webRTC server code witch left to be handle by the developer what is it job ? i mean their is a lot of signaling method using websocket and socket.io but what did they send to the server ? .
i see some github sources in may learning path that provide these "id" i'm wondering does the server code provide these id and what is it job ?.
i did not get how i can share video conf in real base scenario .. any concret example explanation ?.
i'm wondering if i can use a combination of signalR and webRTC . is is possible thus signlaR provide real time communication and data delivering and the webRTC provide many many services like video conf .. audio .. data exchange .. etc . and is it a valid server code ? .
1) The server-side differs depending on the method used for signalling. For WebRTC specifically, because any browser that supports WebRTC will also support WebSocket, WebSocket is the likely candidate to be used for the signalling method.
Now, the server-side for WebSocket can be somewhat complex, as you have to first handle the handshake to elevate the protocol to either ws or wss, and after that, you have to handle the encrypting and decrypting of all messages sent over the line via WebSocket. This is not trivial at all, but if you do some searching around SO and the web in general for information about how to code the server-side for WebSocket, you should be able to find what you're looking for.
2) I can't understand what you're asking in this question. Could you please provide an example/link? Thanks.
3) You use WebRTC to establish a peer-to-peer connection between two clients to quickly transfer data back and forth. One benefit of this peer-to-peer connection (and the speed at which you can transfer data) is the ability to establish video connections. Also, you can establish video links between more than two clients at a time, although with too many connections, there can be bandwidth issues.
What specifically do you want to know about how to use this technology for video conferencing?
4) I'm not too familiar with SignalR, but looking at the home page, SignalR is used to push data from the server. WebRTC doesn't use a server at all (once the peer-to-peer connection has been established). By that rationale, WebRTC will likely always provide a better, faster connection than SignalR.
Please clarify some of your questions as noted above, and I will help in any way I can. Thanks.
I can answer number 4...
You can of course use SignalR to do the signaling between clients to get WebRTC running, but SignalR has no built-in functionality for the WebRTC signaling so you are in for a pretty nasty job if you are planning on doing it your self.
Since you are asking about SignalR I am jumping to conclusions here and guess that you are a .NET developer? If so there are .NET libraries out there that already have taken care of the signaling for you. One of them is XSockets.NET.
Just install the sample package from XSockets and you will have a multi video chat up and running in a minute.
Sorry for not answering 1,2 and 3... But I hope that the package from XSockets will solve these quesitons :)

Looking for a good method to transfer critical real time data over internet

I am searching for a good method to transfer data over internet, and I work in C++/windows environment. The data is binary, a compressed blob of an extracted image. Input and requirements are as follows:
6kB/packet # 10 packets/sec (60kBytes per second)
Reliable data transfer
I am new to network programming and so far I could figure out that one of the following methods will be suitable.
Sockets
MSMQ (MS Message Queuing)
The Client runs on a browser (Shows realtime images on browser). While server runs native C++ code. Please let me know if there are any other methods for achieving the same? Which one should I go for and why?
If the server determines the pace at which images are sent, which is what it looks like, a server push style solution would make sense. What most browsers (and even non-browsers) are settling for these days are WebSockets.
The main advantage WebSockets have over most proprietary protocols, apart from becoming a widely adopted standard, is that they run on top of HTTP and can thus permeate (most) proxies and firewalls etc.
On the server side, you could potentially integrate node.js, which allows you to easily implement WebSockets, and comes with a lot of other libraries. It's written in C++, and extensible via C++ and JavaScript, which node.js hosts a VM for. node.js's main feature is being asynchronous at every level, making that style of programming the default.
But of course there are other ways to implement WebSockets on the server side, maybe node.js is more than you need. I have implemented a C++ extension for node.js on Windows and use socket.io to do WebSockets and non-WebSocket transports for older browsers, and that has worked out fine for me.
But that was textual data. In your binary data case, socket.io wouldn't do it, so you could check out other libraries that do binary over WebSockets.
Is there any specific reason why you cannot run a server on your windows machine? 60kb/seconds, looks like some kind of an embedded device?
Based on our description, you ned to show image information, in realtime on a browser. You can possibly use HTTP. but its stateless, meaning once the information is transferred, you lose the connection. You client needs to poll the C++/Windows machine. If you are prety confident the information generated is periodic, you can use this approach. This requires a server, so only if a yes to my first question
A chat protocol. Something like a Jabber client running on your client, and a Jabber server on your C++/Windows machine. Chat protocols allow almost realtime
While it may seem to make sense, I wouldn't use MSMQ in this scenario. You may not run into a problem now, but MSMQ messages are limited in size and you may eventually hit a wall because of this.
I would use TCP for this application, TCP is built with reliability in mind and you can simply feed data through a socket. You may have to figure out a very simple protocol yourself but it should be the best choice.
Unless you are using an embedded device that understands MSMQ out of the box, your best bet to use MSMQ would be to use a proxy and you are then still forced to play with TCP and possibly HTTP.
I do home automation that includes security cameras on my personal time and I use the .net micro framework and even if it did have MSMQ capabilities I still wouldn't use it.
I recommend that you look into MJPEG (Motion JPEG) which sounds exactly like what you would like to do.
http://www.codeproject.com/Articles/371955/Motion-JPEG-Streaming-Server

How do client-side web-based agents work?

I'm not sure if I'm asking the question properly. I'm referring to locally installed software, often called an "Agent" that keeps in regular communication with some host via HTTP. e.g. When you install LogMeIn, the Agent keeps in communication with the logmein.com server so that when you visit logmein.com with your web browser and connect to the agent, the server is able to initiate communication. The Agent, however, isn't a webserver, nor are any ports forwarded to the Agent. So, is the Agent constantly polling the server asking like a broken record, "Can I help you? Can I help you? Can I help you?" Or is the http connection from Agent to server somehow kept open? I know you can keep an http connection open, but A) how, and B) for how long? Does the Agent need to act like a less annoying broken record asking, "Can I help you? Yet? Yet? Yet?" with much more time in between each question? Or can the Agent ask once and wait indefinitely, asking again only once it learns that the connection has been dropped?
Bottom line is, I'd like to create a small little sample program for trying my hand at writing a client/server application that communicates via the Internet using HTTP. Either side needs to be able to initiate commands / requests. The Agent would likely communicate with the Server using some sort of API, perhaps RESTful. When I start the experiment, I'll be using Perl. It'd be fun to create a Hello World project that would have samples in many languages for many platforms how to write the agent and how to communicate with the server. The agent code would do client side things (e.g. determine public IP address) and send the data to the server. The server would act on the data (e.g. store IP address in a database). The server might also initiate a command to the Agent (e.g. Hey, Agent! What's your CPU type?) Proper authentication / authorization between Agent and Server is of course a necessity.
Are there any existing projects to model off of? Any existing documents? Perhaps I'm just missing terminology and if I just knew that everything I was asking can be summarized by the term foo, then the doors would be opened wide for what I could find in searches!
I looked into the code of Ubuntu's Landscape. It uses Python's Twister -- a web server for HTML5 Websockets. So I'd say what I was looking for in an answer is Websockets (bi-directional communication). That now has opened up a wealth of options, node.js, twister, mojolicious, and many many more as web servers. Turns out using Ajax to poll every few seconds is a very bad idea -- an overwhelming slam on web servers. Keep the connection open.

How do I connect a pair of clients together via a server for an online game?

I'm developing a multi-player game and I know nothing about how to connect from one client to another via a server. Where do I start? Are there any whizzy open source projects which provide the communication framework into which I can drop my message data or do I have to write a load of complicated multi-threaded sockety code? Does the picture change at all if teh clients are running on phones?
I am language agnostic, although ideally I would have a Flash or Qt front end and a Java server, but that may be being a bit greedy.
I have spent a few hours googling, but the whole topic is new to me and I'm a bit lost. I'd appreciate help of any kind - including how to tag this question.
If latency isn't a huge issue, you could just implement a few web services to do message passing. This would not be a slow as you might think, and is easy to implement across languages. The downside is the client has to poll the server to get updates. so you could be looking at a few hundred ms to get from one client to another.
You can also use the built in flex messaging interface. There are provisions there to allow client to client interactions.
Typically game engines send UDP packets because of latency. The fact is that TCP is just not fast enough and reliability is less of a concern than speed is.
Web services would compound the latency issues inherent in TCP due to additional overhead. Further, they would eat up memory depending on number of expected players. Finally, they have a large amount of payload overhead that you just don't need (xml anyone?).
There are several ways to go about this. One way is centralized messaging (client/server). This means that you would have a java server listening for udp packets from the clients. It would then rebroadcast them to any of the relevant users.
A second way is decentralized (peer to peer). A client registers with the server to state what game / world it's in. From that it gets a list of other clients in that world. The server maintains that list and notifies the other clients of people who join / drop out.
From that point forward clients broadcast udp packets directly to the other users.
If you look for communication framework with high performance try look at ACE C++ framework (it has Java bindings).
Official web-site is: http://www.cs.wustl.edu/~schmidt/ACE-overview.html
You could also look into Flash Media Interactive Server, or if you want a Java implementation, Wowsa or Red5. Those use AMF and provide native functionality for ShareObjects including synching of the ShareObjects among connected clients.
Those aren't peer to peer though (yet, it's coming soon I hear). They use centralized messaging managed by the server.
Good luck

What is the best architecture to bridge to XMPP? [closed]

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If I have a separate system with its own concept of users and presence, what is the most appropriate architecture for creating a bridge to an XMPP server network? As far as I can tell there are three primary ways:
Act as a server. This creates one touchpoint, but I fear it has implications for compatibility, and potentially creates complexity in my system for emulating a server.
Act as a clients. This seems to imply that I need one connection per user in my system, which just isn't going to scale well.
I've heard of an XMPP gateway protocol, but it's unclear if this is any better than the client solution. I also can't tell if this is standard or not.
Any suggestions or tradeoffs would be appreciated. For example, would any of these solutions require running code inside the target XMPP server (not likely something I can do).
The XMPP gateway protocol you've heard of is most likely to do with transports. A transport is a server that connects to both a XMPP server and a non-XMPP server. By running a transport, I can use my Jabber client to talk to someone using, say, MSN Messenger.
A transport typically connects once to the remote network for each JID that it sees as online. That is, it's your option 2 in reverse. This is because there is no special relationship between the transport and the non-XMPP network; the transport is simply acting as a bunch of regular clients. For this to work, XMPP clients must first register with the transport, giving login credentials for the remote network, and allowing the transport to view their presence.
The only reason this has a chance of scaling better is that there can be many transports for the same remote network. For example, my Jabber server could run a transport to MSN, another Jabber server could run another one, and so on, each one providing connections for a different subset of XMPP users. While this spreads out the load on the Jabber side, and load balancing on your system may spread out the load as well, it still requires many connections between the two systems.
In your case, because (I assume) the non-XMPP side of things is cooperating, putting a XMPP server interface on the non-XMPP server is likely your best bet. That server interface is best suited for managing the mapping between XMPP JIDs and how that JID will appear on its own network, rather than forcing XMPP users to register and so on.
In case you haven't seen these, you might find them useful:
http://www.jabber.org/jabber-for-geeks/technology-overview
http://www.xmpp.org/protocols/
http://www.xmpp.org/extensions/
Hope that helps.
I too am working on a similar system.
I am going with the gateway/component route. I have looked at several options and settled with this one.
The gateway is basically a component with the specific purpose of bridging Jabber/XMPP with another network. You will have to build most of the things you take for granted when using XMPP as a client. Stuff like roster control.
There is very little help online on the actual design and building of a component. Like the above answer I found that the xmpp protocols/extensions to be of help. The main ones being:
Basic Client 2008
Basic Server 2008
Intermediate Client 2008
Intermediate Server 2008
Reading through these will show you what XEPs you will be expected to be able to handle. Ignore the stuff that will be handled by the server that your component will be attched to.
It's a shame that Djabberd has such poor documentation as their system of "everything is a module" gave the possibility of backend of the server could interface directly to the other network. I made no headway on this.
There are basically two types of server to server (s2s) connections. The first is either called a gateway or a transport, but they're the same thing. This is probably the kind you're looking for. I couldn't find specific documentation for the non-XMPP side, but how XMPP thinks about doing translations to legacy servers is at http://xmpp.org/extensions/xep-0100.html. The second kind really isn't explained in any additional XEPs -- it's regular XMPP s2s connections. Look for "Server-to-Server Communication" in RFC 3920 or RFC 3920bis for the latest draft update.
Since you have your own users and presence on your server, and it's not XMPP, the concepts aren't going to map completely to the XMPP model. This is where the work of the transport comes in. You have to do the translation from your model to the XMPP model. While this is some work, you do get to make all the decisions.
Which brings us right to one of the key design choices -- you need to really decide which things you are going to map to XMPP from your service and what you aren't. These feature and use case descriptions will drive the overall structure. For example, is this like a transport to talk to AOL or MSN chat services? Then you'll need a way to map their equivalent of rosters, presence, and keep session information along with logins and passwords from your local users to the remote server. This is because your transport will need to pretend to be those users and will need to login for them.
Or, maybe you're just an s2s bridge to someone else's XMPP based chess game, so you don't need a login on the remote server, and can just act similarly to an email server and pass the information back and forth. (With normal s2s connections the only session that would be stored would be SASL authentication used with the remote server, but at the user level s2s just maintains the connection, and not the login session.)
Other factors are scalability and modularity on your end. You nailed some of the scalability concerns. Take a look at putting in multiple transports to balance the load. For modularity, see where you want to make decisions about what to do with each packet or action. For example, how do you handle and keep track of subscription data? You can put it on your transport, but then that makes using multiple transports harder. Or if you make that decision closer to your core server you can have simpler transports and use some common code if you need to talk to services other than XMPP. The trade off is a more complex core server with more vulnerability potential.
What architecture you should use depends on the non-XMPP system.
Do you operate the non-XMPP system? If yes, you should find a way to add an XMPP-S2S interface to that system, in other words, make it act as an XMPP server. AOL is using this approach for AIM. Unfortunately, they have restricted their gateway to GoogleTalk.
You don't operate the non-XMPP system but it has a federation interface that you can use - i. e. your gateway can talk to the other system as a server and has a namespace of its own. In this case, you can build a gateway that acts as a federated server on both sides. For I don't know of any example of a gateway that uses this approach but you could use it if you want to build a public XMPP-to-SIP bridge.
If the non-XMPP system doesn't give you a federation interface, then you have no other option but acting as a bunch of clients. In the XMPP world, this is called a "transport". The differences between a transport and a normal server are basically:
the JIDs of the transport are mapped from another system (e.g. john.doe\40example.net#msngateway.example.org - really ugly!)
XMPP users who want to use the transport need to create an account on the non-XMPP system and give the login credentials of that account to the transport service. The XMPP protocol even has a protocol extension that allows XMPP users to do transport registrations in-band.
One other approach is to work with your XMPP server vendor. Most have internal APIs that make injecting presence possible from third party applications. For example, Jabber XCP provides an API for this that's really easy to use.
(Disclosure: I work for Jabber, Inc, the company behind Jabber XCP)

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