While building an application by python for implementing post-processing in Quantum Key Distribution, I require a server (say Alice) and a client(say Bob) at distant locations to interact and exchange some information while doing the required calculations simultaneously (for which threading is used).
At first, the open-source NGROK service was used as a hosting server but it makes the time required for post-processing very huge mainly due to network congestion.
So, is there a way to establish a direct peer-to-peer connection between Alice and Bob via the internet wherein Alice's system itself acts as a server thereby bypassing a third-party host server? If there is a way out please suggest one otherwise any leads or suggestions for open source services better than NGROK would be highly helpful.
PS: If you need any additional information for providing help, i would be eager to respond.
Related
As per the docs, the NATS server design is a "server first" approach, regarding protecting against "lazy clients". Lazy clients are just booted under poor performance scenarios.
Due to this, I have internalized an assumption that anything connecting from past the edge should not connect directly to the NATS server, but should instead access some middle layer service point that manages and maps the more poorly performing external connections through internally initiated connections to the NATS server.
For example, consider a client service node on a remote customer facility that accesses back end services that are serviced over NATS in some form.
Is my stated assumption true in a VERY STRICT sense, such that it is NEVER advisable for that remote node to connect to a NATS server directly, even given a low number of possible client connections AND a stable network service to those facilities?
Or, is it ok-ish to connect directly from that remote node IF, and ONLY IF, I have a known solid infrastructure (low latency, high bandwidth, dependable, etc).
Finally, how about if the path to that remote node is not a very solid network service? Specifically, is it better to A) as I described, use an intermediary service on the back-end for managing the end point connection requests and passing them along over internal connections, or B) is it preferred to just let it connect directly to the server and let NATS boot it as needed, then use a re-connect on drops approach to keep the connection up as best as it can?
A good example here is a mobile endpoint, which could be up and down regularly for various reasons and due to no fault of the device or infrastructure at all.
Currently, I am designing every NATS solution as a "backend connections only" design. If this is overcomplicating my designs needlessly, of course I want to stop forcing that design constraint. :)
I'm in a high availability project which includes deployment of 2-node high availability cluster for hot replacement of services (applications) running on the cluster nodes. The applications have inbound and outbound tcp connections as well as process udp traffic (mainly for communicating with ntp server).
The problem is pretty standard until one needs to provide a hot migration of services to backup node with all the data stored in RAM. Applications are agnostic of backup mechanisms and it is highly undesirable to modify them.
As only approach to this problem, I've come off with a duplication approach assuming that both cluster nodes will run the same applications repeating calculations of each other. In case of failure the primary server the backup server will become a primary.
However, I have not found any ready solution for proxy which will have synchronous port mirroring. No existing proxy servers (haproxy, dante, 3proxy etc.) support such feature as far as I know. Have I missed something, or I should write a new one from scratch?
A rough sketch of the functionality can be found here:
p.s. I assume that it is possible to compare traffic from the two clones of the same application...
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.
I'm making a network game (1v1) where in-game its p2p - no need for a game server.
However, for players to be able to "find each other", without the need to coordinate in another medium and enter IP addresses (similar to the modem days of network games), I need to have a coordination/matching server.
I can't use regular web hosting because:
The clients will communicate in UDP.
Therefore I'll need to do UDP Hole Punching to be able to go through the NAT
That would require the server to talk in UDP and know the client's IP and port
afaik with regular web hosting (php/etc) I can only get the client's IP address and can only communicate in TCP (HTTP).
Options I am currently considering:
Use a hosting solution where my program can accept UDP connection. (any recommendations?)
UDPonNAT seems to do this but uses GTalk and requires each client to have a GTalk account for this (which probably makes it an unsuitable solution)
Any ideas? Thanks :)
First, let me say that this is well out of my realm of expertise, but I found myself very interested, so I've been doing some searching and reading.
It seems that the most commonly prescribed solution for UDP NAT traversal is to use a STUN server. I did some quick searches to see if there are any companies that will just straight-up provide you with a STUN hosting solution, but if there even were any, they were buried in piles of ads for simple web hosting.
Fortunately, it seems there are several STUN servers that are already up and running and free for public use. There is a list of public STUN servers at voip-info.org.
In addition, there is plenty more information to be had if you explore SO questions tagged "nat".
I don't see any other choice than to have a dedicated server running your code. The other solutions you propose are, shall we say, less than optimal.
If you start small, virtual hosting will be fine. Costs are pretty minimal.
Rather than a full-blown dedicated server, you could just get a cheap shared hosting service and have the application interface with a PHP page, which in turn interfaces with a MySQL database backend.
For example, Lunarpages has a $3/month starter package that includes 5gb of space and 50gb of bandwidth. For something this simple, that's all you should need.
Then you just have your application poll the web page for the list of games, and submit a POST request in order to add their own game to the list.
Of course, this method requires learning PHP and MySQL if you don't already know them. And if you do it right, you can have the PHP page enter a sort of infinite loop to keep the connection open and just feed updates to the client, rather than polling the page every few seconds and wasting a lot of bandwidth. That's way outside the scope of this answer though.
Oh, and if you're looking for something absolutely free, search for a free PHP host. Those exist too! Even with an ad-supported host, your app could just grab the page and ignore the ads when you parse the list of games. I know that T35 used to be one of my favorites because their free plan doesn't track space or bandwidth (it limits the per-file size, to eliminate their service being used as a media share, but it shouldn't be a problem for PHP files). But of course, I think in the long run you'll be better off going with a paid host.
Edit: T35 also says "Free hosting allows 1 domain to be hosted, while paid offers unlimited domain hosting." So you can even just pay for a domain name and link it to them! I think in the short term, that's your best (cheapest) bet. Of course, this is all assuming you either know or are willing to learn PHP in order to make this happen. :)
There's nothing that every net connection will support. STUN is probably good, UPnP can work for this.
However, it's rumored that most firewalls can be enticed to pass almost anything through UDP port 53 (DNS). You might have to argue with the OS about your access to that port though.
Also, check out SIP, it's another protocol designed for this sort of thing. With the popularity of VOIP, there may be decent built-in support for this in more firewalls.
If you're really committed to UDP, you might also consider tunneling it over HTTP.
how about you break the problem into two parts - make a game matcher client (that is distinct from the game), which can communicate via http to your cheap/shared webhost. All gamers who wants to use the game matching function use this. THe game matcher client then launches the actual game with the correct parameters (IP, etc etc) after obtaining the info from your server.
The game will then use the standard way to UDP punch thru NAT, etc etc, as per your network code. The game dont actually need to know anything about the matcher client or matcher server - in the true sense of p2p (like torrents, once you can obtain your peer's IPs, you can even disconnect from the tracker).
That way, your problems become smaller.
An intermediate solution between hosting your own dedicated server and a strictly P2P networking environment is the gnutella model. In that model, there are superpeers that act like local servers, having known IP addresses and being connected to (and thus having knowledge of) more clients than a typical peer. This still requires you to run at least one superpeer yourself, but it gives you the option to let other people run their own superpeers.
Closed. This question is opinion-based. It is not currently accepting answers.
Want to improve this question? Update the question so it can be answered with facts and citations by editing this post.
Closed 3 years ago.
Improve this question
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)