I am working on designing an API for consuming messages from an application that will generate a very large amount of data; 10+ of GB/s is likely, even for smaller clients. I am looking for a protocol that allows me to deliver this data in a way that is easy for clients to consume.
The obvious answer for me is: split up the messages so they are consumable over multiple connections. Each connection would consume a fraction of the overall load.
But if I do this, there are a few things I need to account for:
How does the user know they are falling behind and need to launch more connections?
Twitter says consumers should check timestamps, which could work for us
When they launch a new connection to consume more of the data, how do they specify that this is part of the same consumption session?
We could give the session a name, correlate that with a "direct" amqp queue, and let our queue do the hard work
Is there something very important I am missing.
Probably.
For this reason, I'd much rather a protocol that already exists.
The protocol would be considered extra awesome if it:
is websocket or streaming HTTP friendly
supports data compression
The problems you are describing are pretty much the same issues that video streaming has to deal with, which you probably already know. The key HTTP friendly streaming protocols are HLS (Apple), SmoothStreaming (Microsoft), HDS (Adobe) and MPEG-DASH (open protocol, but new).
When considering video streaming, it is also worth understanding whether your streams are more like 'live' streams or 'static' content - the former is generated on the fly and any given part of the live stream may only be available for a set tine, while the latter stored on the server in full and generally any part is available at any time (until the content is removed). How you stream and playback these is subtly different.
It may be that you can simply reuse one of the above video streaming protocols by wrapping your data as if it were video (or maybe it even is video), and implementing your own custom client on the receiving side.
Alternatively, these protocols could provide a good reference point if you wanted to create your own simpler protocol - there are several open source streaming servers you could look to for ideas or even adapt to your needs if that looks like a sensible route:
http://gstreamer.freedesktop.org
http://icecast.org
Video streaming is quite complex as you may already be aware, but if your use cases are simpler you may be able to ignore or remove much of the complexity - for example you may not need seek, multiple format and bit rate streams, accompanying streams (for subtitles etc). Being able to simplify like this might justify the effort to modify one of the above for your needs, if you are not able to use them out of the box.
One final point - video and audio streaming protocols usually have a built in way of dealing with delayed or lost packets. Depending on your application these may not be applicable to you so you should look carefully at this aspect if reusing a video or audio streaming protocol or server. For example, audio clients are typically tolerant of a small amount of packet loss, and will generally discard delayed packets rather than pause the audio (packets received outside the 'jitter buffer' window). If your application cannot tolerate any packet loss, then you will need to look carefully at the underlying solution and protocol to make sure it really meets your needs over all network conditions.
I am about to develop a service that involves an interactive audio live streaming. Interactive in the sense that a moderator can have his stream paused and upon request, stream audio coming from one of his listeners (during the streaming session).
Its more like a Large Pipe where what flows through but the water can come in from only one of many small pipes connected to it at a time with a moderator assigned to each stream controlling which pipe is opened. I know nothing about media streaming, I dont know if a cloud service provides an interactive programmable solution such as this.
I am a programmer and I will be able to program the logic involved in such interaction. The issue is I am a novice to media streaming, don't have any knowledge if its technologies and various software used on the server for such purpose, are there any books that can introduce on to the technologies employed in media streaming, and I am trying to avoid using Flash,?
Clients could be web or mobile. I dont think I will have any problem with integrating with client system. My issue is implementing the server side
You are effectively programming a switcher. Basically, you need to be able to switch from one audio stream to the other. With uncompressed PCM, this is very simple. As long as the sample rates and bit depth are equal, cut the audio on any frame (which is sample-accurate) and switch to the other. You can resample audio and apply dithering to convert between different sample rates and bit depths.
The complicated part is when lossy codecs get involved. On a simlar project, I have gone down the road of trying to stitch streams together, and I can tell you that it is nearly impossible, even with something as simple as MP3. (The bit reservoir makes things difficult.) Plus, it sounds as if you will be supporting a wide variety of devices, meaning you likely won't be able standardize on a codec anyway. The best thing to do is take multiple streams and decode them at the mix point of your system. Then, you can switch from stream to stream easily with PCM.
At the output of your system, you'll want to re-encode to some lossy codec.
Due to latency, you don't typically want the server doing this switching. The switching should be done at the desk of the person encoding the stream so that way they can cue it accurately. Just write something that does all of the switching and encoding, and use SHOUTcast/Icecast for hosting your streams.
Can anyone provide any direction or links on how to use the adaptive bitrate feature that DSS says it supports? According to the release notes for v6.0.3:
3GPP Release 6 bit rate adaptation support
I assume that this lets you include multiple video streams in the 3gp file with varying bitrates, and DSS will automatically serve the best stream based on the current bandwidth. At least that's what I hope it does.
I guess I'm not sure what format DSS is expecting to receive the file. I tried just adding several streams to a 3gp file which resulted in Quicktime unable to play it, and VLC opening up a different window for each stream in the file.
Any direction would be much appreciated.
Adaptive streaming used in DSS 6.x uses a dropped frame approach to reduce overall bandwidth rather than dynamic on the fly bitrate adjustments. The result of this can be unpredictable. The DSS drops the frames, and does not need the video encoded in any special way for it to work.
I need to test the application I developed, that uses HTTP Live Streaming ( audio only ) and I would like to see how it works in comparison with other similar apps. How can I conclude that an app uses HTTP Live Streaming by using a packet sniffer?
Look for a request for a playlist file with an .m3u8 extension and in the response look for #EXTM3U on the first line. The IETF draft should give you all the information you need to interpret the rest of the traffic.
Here are some values you might find useful if you want to look at the stream closer.
I have never tried this tool but Apple released a stream validator, all you need is the playlist URL.
Bear in mind that it's possible to use a number of different streaming technologies and it's a popular practice with alot of services to use the one which makes most sense at the time. If you don't simulate the correct environment you might never get an HLS stream at all.
I plan on writing an automated bot for a game.
The tricky part is figuring out how they encoded their protocol... To make the bot run around is easy, simply make the character run and record what it does in wireshark. However, interpreting the environment is more difficult... It recieves about 5 packets each second if you are idle, hence lots of garbarge.
My plan: Because the game runs under TCP, I will use freecap (http://www.freecap.ru/eng) to force the game to connect to a proxy running on my machine. I will need this proxy to be capable of packet injection, or perhaps a server that is capable of resending captured packets. This way I can recreate and tinker around with what the server sends, and understand their protocol encoding.
Does anyone know where I can get a proxy that allows packet injection or where I can perform packet injection (not via hardware, as is the case with wireless or anything!)
Where of if I can find a server/proxy that resends captured packets (ie: replays a connection).
Any better tools or methodologies for pattern matching? Something which can highlight patterns from mutliple messages would be GREAT.
OR, is there a better way to decipher this here? Possibly a dissasembly strategy (via hooking a winsock function and starting the dissassembly from there) ? I have not done this before so I am not sure. OR , any other ideas?
Network traffic interception and protocol analysis is generally a less favored method to accomplish your goal here. For most modern games, encryption is a serious factor, and there are serious headaches associated with the protocol analysis for any but trivial factors of the most common gameplay scenarios.
Most modern implementations* of what you are trying to do rely on reading and manipulating the memory space and process of a running client. The client will have already done all the hard parts for you, including decrypting the traffic and sorting it into far more easy to read data structures. For interacting with the server you can call functions built into the client instead of crafting entire series of packets from scratch. The plus to this approach is that you have to do far less work to interpret the data and produce activity. The minus is that there is often some data in the network traffic that would be useful to a bot but is discarded by the client, or that you may want to send traffic to the server that the client cannot produce (which, in my own well-developed hierarchy for such, is a few steps farther down the "cheating" slope).
*...I say this having seen the evolution of the majority of MMORPG botting/hacking communities from network protocol analyzers like ShowEQ and Odin's Eye / Excalibur to memory-based applications like MacroQuest and InnerSpace. On that note, InnerSpace provides an excellent extensible framework for the memory/process-based variant of what you are attempting, and you should look into it as a basis for your project if you abandon the network analysis approach.
As I've done a few game bots in the past (for fun, not profit or griefing of course - writing game bots is a lot of fun), I recommend the following:
If you can code and there isn't cheat protection preventing you from doing it, I highly recommend writing an injected DLL for the following reasons:
Your DLL will be able to access the game's memory space directly, and once you reverse-engineer the data structures (either by poking around memory or by code disassembly), you'll have access to lots of data. This will also allow you to bypass any network encryption the game may have. The downside of accessing process memory directly is that offsets and data structures change between versions - however, data structures don't change very often with a stable game, and you can compensate offset changes by searching for code patterns instead of using fixed offsets.
Either way, you'll still be able to hook WinSock functions using API hooks (check out Microsoft Detours and the excellent but now-commercial madCodeHook).
otherwise, I can only advise that you give live/interactive packet editors like WPE Pro a try.
In most scenarios, the coolest methods (code reverse-engineering and direct memory access) tend to be the least productive. They require a lot of skill (to understand the code) and time, both initially (to go through all the code and develop code to interact with the data structure) and for maintainance (in case the game is being updated). (Of course, they sometimes do allow doing cool stuff which is impossible to do with the official client, but most of the time this is obvious as blatant cheating, and likely to attract the GMs quickly). Most of the time bots are made by replacing game graphics/textures with solid colours, and creating simple "pixel" bots which search for certain colours on the screen and react accordingly (e.g. click them).
Hope this helps, and remember - cheating is only fun when it doesn't make the game less fun for everyone else ;)
There are probably a few reasonable assumptions you can make that should simplify your task enormously. However, to make the best use of them you will probably need greater comfort with sleeves-rolled-up programming than it sounds like you have.
First, it's a safe bet that the encryption they are using falls into one of three categories:
None
Cheesy
Far better than you are likely to crack
With the odds of the middle case being very low.
Next, it's a safe bet that the packets are encrypted / decrypted close to the edge of the program (right as they come in, right before they go out) and that the body of the game deals with them in decrypted form.
Finally, the protocol they are using most likely consists of either
ascii with data blocks
binary goo
So do a little packet sniffing with a card set in promiscuous mode for unencrypted ascii. If you see some, great, you're ahead of the game. But if you don't give up the whole tapping-the-line idea and instead start following the code as it returns from the sending data out by breakpointing and stepping with a debugger. Figure the outermost layer or three will be standard network stuff, then will come the encryption layer, and beyond that the huge mass of stuff that deals with the protocol unencrypted.
You should be able to get this far in an hour if you're hot, a weekend if you're reasonably skilled, motivated, and diligent, and never if you are hopeless. But it is possible in principle (and doubtlessly far easier in practice) to do it this way.
Once you get to where something that looks like unencrypted goo comes in, gets mungled, and the mungled form goes out, then start worrying about what it means.
-- MarkusQ
A) I play a MMO and do not support bots, voting down...
B) Download Backtrack v.3, run an arpspoof on your default gateway and your host. There is an application that will spoof the remote host's SSL cert sslmitm (I believe is the name) which will then allow you to create a full connection through your host. Then fireup tcpdump/ethereal/wireshark (choose your pcap poison) and move around do random stuff to find out what packet is doing what. That will be your biggest challenge; but proxying with a Man in the Middle attack on yourself is the way to go.
C) I do not condone this activity, this information is only being provided as free information.
Sounds like there is not encryption going on, so you could do a network approach.
A great place to start would be to find the packet ID's - most of the time, something near the front of the packet is going to be an ID of the type of the packet. For example move could be 1, shoot fired could be "2", chat could be "4".
You can write your own proxy that listens on one port for your game to connect, and then connects to the server. You can make keypresses to your proxy fire off commands, or you can make your proxy write out debugging info to help you go further.
(I've written a bot for an online in game in PHP - of all things.)