We develop local multiplayer games in AIR+AS3+Flex environment. We use the p2plocal library, which based on the RTMFP protocol. We found the following error at the testing:
it seems there is a limit of the nodes numbers. Around about 16 nodes, errors appear in the RTMFP network. For example: if there are 15 nodes in the network already, and a new one joined to it, some older node dont see the new one and "vica versa". Over 16 nodes, the RTMFP newtwork is inaccurate: some nodes dont see some others without any logic.
Is this a problem of the RTMFP? Or maybe a Windows socket limit? Or..?
Anyone has any idea?
RTMFP is UDP based for P2P communication and is highly dependant on network (may not work reliably).
Test your RTMFP capabilities with these tools:
http://cc.rtmfp.net/
http://blog.yoz.sk/2010/07/rtmfp-connection-tester/
To develop a solution that works reliably for most users, a RTMP relay server can be used.
Related
I am trying to use WebRTC to implement browser to browser communication.
I want to allow communication between two browsers running on two different computers in the same LAN.
Can somebody confirm is this possible and if so, how.
I tried looking up the demos and seems they all have examples for application running in the same page. But I would like to connect to a peer computer using some IP address.
Regards
I have just solved your problem using https://github.com/feross/simple-peer
Yes, it is possible to create a Peer to pear connection using PEERJS. PeerJS simplifies WebRTC peer-to-peer data, video. PeerJS wraps the browser's WebRTC implementation to provide a complete, configurable, and easy-to-use peer-to-peer connection API within a LAN Network. PEERJS
Is there a way to stream a video and audio on a website just to the clients, using a camera installed on the server - for instance, like youtube does ?
I've started reading webrtc, but if I use webrtc I should create a stun/turn server and other things, which for one way stream I think is not necessary (this is just my understanding of the things..) because I don't need anything from the clients, literally, neither their video, or audio..
So is there a way to achieve this using html5, streaming just in one direction:
server (camera) -> clients
Is there something about this out there, or should I stick with webrtc ?
I'm going to explain a possible solution for this scenario, there might be others, but I hope mine gives you a rough idea of how you could do it and a start point to explore more about the amazing possibilities of WebRTC. Please let me know if something is not understood.
So, WebRTC is a free, open project that provides browsers and mobile applications with Real-Time Communications (RTC) capabilities via simple APIs. Sweet, that is: WebRTC has a quite good browser support (not in every browser though, Safari just started supporting it a month ago with Safari 11). But in this case we want to use WebRTC in the server side. At the end of the day we can still think about peer-to-peer real time communication, where one of our peers is the server.
I don't know if you are familiar with Node.js, but I recommend you to write your Server app with it (<3 Javascript!):
There are a few libraries that wrap WebRTC functionality to be used in the server side, like node-webrtc and node-rtc-peer-connection.
But I recommend you to to take a look at electron-werbrtc, since
the others might be using deprecated methods or be incomplete.
electron-webrtc runs a headless Electron client in the background to
use Chromium's built-in WebRTC implementation. So with it you should
be able to access the Camera in your server and create a stream to be
served to the other peer (the browser).
All above would be the WebRTC related tasks, in this case: streaming video peer(server)-to-peer(browser).
Now, let's talk about signaling process, stun and turn.
Signaling: imagine now a scenario peer-to-peer with 2 browsers, they want to establish a direct connection and stream video and audio between each other. But they don't know each other, like if I don't know your home address, I can't send you a letter. So they need a service that helps them know each other, so they can have the other's IP. This should be done by what is called "a signaling server". If somehow you know the other peer IP, you wouldn't need a signaling server.
STUN/TURN: the scheme above works perfectly in a local area network where each peer has its own IP address and there are no firewalls and routers between them. But otherwise, you can have peers behind a NAT or firewalls, and then your signaling server won't be able to make both peers to discover themselves. If you have peers behind a NAT, you'll need a STUN server, and if you have peers behind firewalls you'll need a TURN server. This is a bit simplified, but I just want you to have the general picture of when you might need STUN/TURN servers.
To better understand Signaling, STUN and TURN, there is a very graphic article that explains them perfectly.
Now, for your scenario:
I think you prob don't need STUN/TURN servers and also you prob don't need to implement the signaling process, because the browsers that are supposed to receive the stream from the server will know that server address, right? So they can establish a WebRTC connection with it.
EDIT: it is likely that you will need to implement some sort of handshake between the server and the clients (browsers), so this will be the signaling process. This is not part of WebRTC and this is why you need to implement it yourself. As I said, it is the way 2 peers can discover each other, but they also exchange information as their local media conditions, like codecs, resolutions they can handle, etc. For your case, your signaling server could be hosted in the same server you use to strea: you can build a small node.js app that runs there and that manages all the signaling process easily, it is not a big deal. I recommend you to read this article, and specially the section "How can I build a signaling service?". In general all WebRTC articles from that site are very helpful.
Does this make sense to you? I think with it you can start digging a little bit more and see if with this is enough or you need to implement more stuff. Hope it helps!
I am building a small cards game for Windows Store using HTML/JS as my programming languages. One of the features that I would like to add is multiplayer capability. My game it's based on a 1 versus 1 player (unlike Hearts where you need 4 players), so an ad-hoc peer-to-peer connection is enough. Also, keep in mind that I am only considering local network multiplayer, without internet support (meaning that "privateNetworkClientServer" capability is required on that app manifest).
So I am imagining, when a player want to start a multiplayer game, the app will periodically broadcast a message to find any candidates. Meanwhile he will also have to listen for those same messages (in case of another player is broadcasting them also). When they find which other we transmit the game state back and forward to perform the required games changes.
My question is, does WinRT provide any functionality out of the box to do something like this? If no, do you have any suggestion for my problem?
Thanks
Look at the documentation for the PeerFinder class. Proximity can use either NFC or by browsing on the same subnet. Note, in the case of WiFi, not all WiFI cards support the browsing model, so some older PCs may not be able to use this solution.
The proximity sample application on msdn should help you with this.
trying to create a program that works as following .video is captured by web cam of laptop and it is streamed to android phone using wifi .According to the video displayed, user types some messages and sents back to laptop simultaneously.
googled and found that making an ad-hoc wireless network will work,but forum
discussions says its not supported by android.want my app to work from 2.2 onwards
Which Socket communication protocol (UDP or TCP) protocol should be used to stream video?
Since want to implement two way communication, which one must be the server (laptop or mobile device)
Please guide me how to implement this
use connectify or any virtual router to get connected to your android phone.so app can be connected with your laptop or desktop.
go for UDP its fast and recover data even if it's lost and also use a proper data streaming strategy or a protocol.
According to your application design , i will recommend you to make your laptop work as a server. Also it would be best if you choose to use threading in both application and recommend your app to act as server or client as it's required shows more robustness..!
What is a good framework to build a multiplayer game in Actionscript?
I want to create a multiplayer 2D shooter like Asteroids on the Blackberry Playbook; my main concern is latency - a shooter wouldn't be fun if the bullets are super-jerky and unexpectedly hit people.
I'm guessing that a UDP-based framework would be the best. Can anyone point me to the right direction?
There are many things you can use off the shelf but the basic setup is very simple but you have a few options.
The most common is server push, things like Flash Media Server, LiveCycle Data Services from Adobe or other tools like SmartFoxServer can do this. With this setup the server saves the connections to everyone that connects to the server and passes or "pushes" applications state to the people connected every time the data changes in the application.
Another option is called long pulling, this can be done with any web server really. How this works is the data stores the state of the application, when the application starts it calls the server, when it responds the client calls the server again.
There are a few other ways to do it but these are the most common. But this has nothing to do with protocol like HTTP, UDP, AMF, XMPP, or whatever else. The protocol is the format that the data is sent. With these out of the box servers they normally output a few of these but the fastest formats are binary like AMF but not always the best, there are advantages to each, because each gives you different features for keeping track of things.
If you are talking about have a game that takes over the world that has millions of users then you need to think about scaling and what happens when you need two or 100 servers and how do they talk to each other. But for now keep in mind that the more the server does the slower it will get, if you are sending small amounts of data it will be able to handle more users. Stick with making one efficient server and worry about that later if you get there.
You also need to thing about what server side programming language you want to mess with if any. Some services don't let you do anything, these normally cost money and don't do as much. Adobe likes Java but there are servers that output all of these protocols in most every language. My favorit lately has been Node.js a super fast way to run JavaScript on the server. Node.js has a built in HTTP server but it is just as easy to create a simple server that sends basic text through a Socket or XMLSocket. A server like this will easily handle many thousands of users. There are many games that use Socket.IO and if you want to see a simple example of what I'm talking about you can check out this.
Assuming you want to use Flash/Flex and not Java (Blackberry/Android) or native SDKs for Playbook -
There is a book as an inspiration: http://www.packtpub.com/flash-10-multiplayer-game-essentials/book it uses Pulse SDK at the server side. But you could use an own sockets-program on the server side. I use Perl as TCP-sockets server (sends gzipped XML around) in a small card game but this wouldn't work for your shooter.
Flash does not support UDP out of the box
But there is peer-to-peer networking protocol RTMFP in the upcoming Flash Media Server Enterprise 4 (price is out of reach for mere mortals)
So your best bet is to buy an Amazon-service for RTMFP then you can pay-per-use and stay scalable...
You can either do a constant post/get request with the server to get data for the game, but for a multiplayer shooter i'd surgest SmartFoxServer: http://www.smartfoxserver.com/
Out of the box, Adobe AIR supports UDP through datagram packets.
http://help.adobe.com/en_US/air/reference/html/flash/net/DatagramSocket.html
I couldn't find a particular networking API for flash, but perhaps you can build one. Libgren is open source and you can use that for reference.
You can also look into RTMFP though it's focus is on transmitting audio/video and some messages (through TCP I think).