setting up a UDP server and client using netty - networking

Netty is really well documented when it comes to TCP, but I wanted to try a simple UDP server-client example and didn't find any good code out there. (mostly mailing lists and users with allegedly buggy code)
Anyone care to provide some simple example? Thanks!

Maybe this will help you
https://github.com/normanmaurer/javamagazin-netty-ws/tree/master/src/main/java/me/normanmaurer/javamagazin/netty/examples/ws
It also bootstrap a simple udp server

Related

Is there a kind of opensource framework base on dpdk, also implement userspace tcp/ip protocol?

Besides, this framework support multithread construct, provide POSIX_like interface(including epoll), also support multicast.
Thank you all!
you may try using vpp by FDio.
you can give a try to UDPDK which is the most usable and closest to UDP like implementation (with DPDK under the hood) that I have found so far though you have to add/tweak some things before integrating in prod code

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 :)

Is there a way to intercept all http, https traffic

I have used a lot of parent control software but none of them is perfect. I am thinking to write my own. I want to use either C++ or java or combination of two. My main issue is how to capture all traffic originating from browser.
I want to do it in a way hack proof way.
I appreciate greatly any help on this.
Thanks in advance.
You can't intercept data transfers from your http/https connections
You will have to build a Packet sniffer and find a way to filter out the packages you are looking for, To my suprise im not getting any solid results when i try to google C++ packet sniffer tutorials, but thats defeneteley the way to go.
For windows you need create filter driver for network adapter. Under linux you can use raw sockets for this purpose. Unfortunately, windows not support full row socket functionality.

How do modern implementations of Comet/Reverse AJAX work? Any stable C# WCF or ASP.NET implementations?

What is the correct way (or best) way to implement Comet, HTTP Push, or Reverse AJAX?
What .NET implementations would you recommend?
I have hear about, WebSync and PokeIn, both are paid implementations, I have used PokeIn and its pretty straight forward. If you are looking forward to code your own COMET implementation, I just can say that its a complex task, because you need to modify the natural behaviour if IIS. Its a hacky way to get around the limitations of the HTTP protocol and you need to know really well what you doing so don't end up breaking things around =).
It's also known as long-lived
requests. This is also by far the most
complex method to implement.
Basically, a request is made by the
client, and the server very slowly
responds, which causes the connection
to be maintained. Periodically, when
the server has something to push,
it'll "burst" send the information, so
to speak. This approach gives you
real-time push, which is great. But,
it has a serious down-side: holding
connections open like that isn't how
the underlying protocols are meant to
work, and most servers aren't terribly
happy about it. If your traffic gets
too great, you'll chew up threads on
the server and wind up bringing your
site down.
ref: http://www.coderanch.com/t/121668/HTML-JavaScript/does-Reverse-Ajax-Works
JOBG is correct re: the complexities; it's probably not a task you want to undertake lightly. I'm one of the authors of WebSync, and I can attest that it's a difficult task.
There are a ton of examples in the download, and the community edition is free.
Microsoft is developing HTTP push in SignalR
https://github.com/SignalR/SignalR

How to setup Quality of Service?

I'm talking about http://en.wikipedia.org/wiki/Quality_of_service. With streaming stackoverflow podcasts and downloading the lastest updates to ubuntu, I would like to have QoS working so I can use stackoverflow without my http connections timing out or taking forever.
I'm using an iConnect 624 ADSL modem which has QoS built-in but I can't seem to get it to work. Is it even possible to control the downstream (ie. from ISP to your modem)?
I don't know if this will help you, but I've never been a fan of using the ISP provided box directly. Personally I use a Linksys wrt54gl, with DD-wrt, behind(DMZ) my ISP provided box.
DD-wrt has excellent QoS management.
Sorry I can't be more help with your existing hardware.
You just need the tc command to handle the QoS on Linux boxen. However I wouldn't expect that much from it because of the results I obtained and detailed here.

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