In our File Transfer application the network performance was fair
but we want to get the maximum network performance so one way of achieving through
adaptive bandwidth allocation .So the application will be forced to attain the
available bandwidth.friends!!! if u have any white papers or code for reference
it would be much helpful :)
thanks
krishna
If you just throw it at the TCP session with no control, it will transfer at full speed.
You could also compact the file as you transfer. It will not accelerate the transfer, but will optmize the use of the network, at CPU coast.
If it is not enough, the only [software] way to improve that even more is by using multiple TCP sessions so you will reduce the speed delimitating effects of the latency over the TCP flow control. I beleave 5 concurrent transfers from different offsets of the same file will do the job, faster impossible.
I don't think "adaptive bandwidth allocation" really means anything tangible (considering it's the #2 google hit for that expression!) but I'll try to give an answer that might help you ask a better question.
If an application's network activity can be parallelised (bittorrent is a good example of this) then this is one way of achieving faster network transfers.
In general though, for user space applications the networking conditions are going to be outside the application's control for good reasons. If a userspace application considers it part of its mandate to adjust or affect external operating system-level networking conditions I would consider it malware. QoS for example could be used to prioritise the traffic associated with your application but that is something you might want to suggest and explain in a deployment guide and not try to manage from within your application.
Related
I have a way to control message size when I stream the data through grpc. Unfortunately I am not able to find info on what would be optimal message size. I found this but it is not resolved.
Is keeping it under 4MB threshold good enough or there are some guidelines?
It depends a lot on your application needs, network configuration, and language. Messages around 16-64K are perhaps best suited for the most wide variety of configurations including mobile etc. For pure throughput-oriented workloads in data centers we regularly see GB sized messages, but 1 MB messages are perhaps pretty close to ideal tradeoff of minimal computational overhead and immediate memory capacity needs for the amount of network pipelining that they provide.
I'm in a situation where, logically, UDP would be the perfect choice (i need to be able to broadcast to hundreds of clients). This is in a very small and controlled environment (the whole network is over a few square metters, all devices are local, the network is way oversized with gigabit ethernet and switches everywhere).
Can i simply "ignore" all of the added reliability that needs to be tossed on udp (checking messages arrived, resending them etc) as those mostly apply where the is expected packet loss (the internet) or is it really suggested to handle udp as "may not arrive" even in such conditions?
I'm not asking for theorycrafting, really wondering if anyone could tell me from experience if i'm actually likely to have udp packets missing in such an environment or is it's going to be a really rare event as obviously sending things and assuming that worked is much simpler than handling all possible errors.
This is a matter of stochastics. Even in small local networks, packet losses will occur. Maybe they have an absolute probability of 1e-10 in a normal usage scenario. Maybe more, maybe less.
So, now comes real-world experience: Network controllers and Operating systems do have a tough live, if used in high-throughput scenarios. Worse applies to switches. So, if you're near the capacity of your network infrastructure, or your computational power, losses become far more likely.
So, in the end it's just a question on how high up in the networking stack you want to deal with errors: If you don't want to risk your application failing in 1 in 1e6 cases, you will need to add some flow/data integrity control; which really isn't that hard. If you can live with the fact that the average program has to be restarted every once in a while, well, that's error correction on user level...
Generally, I'd encourage you to not take risks. CPU power is just too cheap, and bandwidth, too, in most cases. Try ZeroMQ, which has broadcast communication models, and will ensure data integrity (and resend stuff if necessary), is available for practically all relevant languages, and runs on all relevant OSes, and is (at least from my perspective) easier to use than raw UDP sockets.
I am looking for networking designs and tricks specific to games. I know about a few problems and I have some partial solutions to some of them but there can be problems I can't see yet. I think there is no definite answer to this but I will accept an answer I really like. I can think of 4 categories of problems.
Bad network
The messages sent by the clients take some time to reach the server. The server can't just process them FCFS because that is unfair against players with higher latency. A partial solution for this would be timestamps on the messages but you need 2 things for that:
Be able to trust the clients clock. (I think this is impossible.)
Constant latencies you can measure. What can you do about variable latency?
A lot of games use UDP which means messages can be lost. In that case they try to estimate the game state based on the information they already have. How do you know if the estimated state is correct or not after the connection is working again?
In MMO games the server handles a large amount of clients. What is the best way for distributing the load? Based on location in game? Bind a groups of clients to servers? Can you avoid sending everything through the server?
Players leaving
I have seen 2 different behaviours when this happens. In most FPS games if the player who hosted the game (I guess he is the server) leaves the others can't play. In most RTS games if any player leaves the others can continue playing without him. How is it possible without dedicated server? Does everyone know the full state? Are they transfering the role of the server somehow?
Access to information
The next problem can be solved by a dedicated server but I am curious if it can be done without one. In a lot of games the players should not know the full state of the game. Fog-of-war in RTS and walls in FPS are good examples. However, they need to know if an action is valid or not. (Eg. can you shoot me from there or are you on the other side of the map.) In this case clients need to validate changes to an unknown state. This sounds like something that can be solved with clever use of cryptographic primitives. Any ideas?
Cheating
Some of the above problems are easy in a trusted client environment but that can not be assumed. Are there solutions which work for example in a 80% normal user - 20% cheater environment? Can you really make an anti-cheat software that works (and does not require ridiculous things like kernel modules)?
I did read this questions and some of the answers https://stackoverflow.com/questions/901592/best-game-network-programming-articles-and-books but other answers link to unavailable/restricted content. This is a platform/OS independent question but solutions for specific platforms/OSs are welcome as well.
Thinking cryptography will solve this kind of problem is a very common and very bad mistake: the client itself of course have to be able to decrypt it, so it is completely pointless. You are not adding security, you're just adding obscurity (and that will be cracked).
Cheating is too game specific. There are some kind of games where it can't be totally eliminated (aimbots in FPS), and some where if you didn't screw up will not be possible at all (server-based turn games).
In general network problems like those are deeply related to prediction which is a very complicated subject at best and is very well explained in the famous Valve article about it.
The server can't just process them FCFS because that is unfair against players with higher latency.
Yes it can. Trying to guess exactly how much latency someone has is no more fair as latency varies.
In that case they try to estimate the game state based on the information they already have. How do you know if the estimated state is correct or not after the connection is working again?
The server doesn't have to guess at all - it knows the state. The client only has to guess while the connection is down - when it's back up, it will be sent the new state.
In MMO games the server handles a large amount of clients. What is the best way for distributing the load? Based on location in game?
There's no "best way". Geographical partitioning works fairly well, however.
Can you avoid sending everything through the server?
Only for untrusted communications, which generally are so low on bandwidth that there's no point.
In most RTS games if any player leaves the others can continue playing without him. How is it possible without dedicated server? Does everyone know the full state?
Many RTS games maintain the full state simultaneously across all machines.
Some of the above problems are easy in a trusted client environment but that can not be assumed.
Most games open to the public need to assume a 100% cheater environment.
Bad network
Players with high latency should buy a new modem. I don't think its a good idea to add even more latency because one person in the game got a bad connection. Or if you mean minor latency differences, who cares? You will only make things slower and complicated if you refuse to FCFS.
Cheating: aimbots and similar
Can you really make an anti-cheat software that works? No, you can not. You can't know if they are running your program or another program that acts like yours.
Cheating: access to information
If you have a secure connection with a dedicated server you can trust, then cheating, like seeing more state than allowed, should be impossible.
There are a few games where cryptography can prevent cheating. Card games like poker, where every player gets a chance to 'shuffle the deck'. Details on wikipedia : Mental Poker.
With a RTS or FPS you could, in theory, encrypt your part of the game state. Then send it to everyone and only send decryption keys for the parts they are allowed to see or when they are allowed to see it. However, I doubt that in 2010 we can do this in real time.
For example, if I want to verify, that you could indeed be at location B. Then I need to know where you came from and when you were there. But if you've told me that before, I knew something I was not allowed to know. If you tell me afterwards, you can tell me anything you want me to believe. You could have told me before, encrypted, and give me the decryption key when I need to verify it. That would mean, you'll have to encrypt every move you make with a different encryption key. Ouch.
If your not implementing a poker site, cheating won't be your biggest problem anyway.
With a lot of people accessing games on mobile devices, a "bad network" can occur when a player is in an area of poor reception or they're connected to a slow-wifi connection. So it's not just a problem of people connecting in sparsely populated areas. With mobile clients "bad networks" can occur very very often and it's usually EXTREMELY hard to diagnose.
UDP results in packet loss, but even games that use TCP and HTTP based can experience problems where the client & server communication slows to a crawl while packets are verified to have been sent. With communication UDP compensation for packet loss USUALLY depends on what the packets contain. If you're talking about motion data, usually if packets aren't received, the server interpolates the previous trajectory and makes a position change. Usually it's custom to the game how this is handled, which is why people often avoid UDP unless their game type requires it. Often to handle high network latency, problems games will automatically degrade the amount of features available to the users so that they can still interact with the game without causing the user to get kicked or experience too many broken features.
Optimally you want to have a logging tool like Loggly available that can help you find errors related to bad connection and latency and show you the conditions on the clients and server at the time they happened, this visibility lets you diagnose common problems users experience and develop strategies to address them.
Players leaving
Most games these days have dedicated servers, so this issue is mostly moot. However, sometimes yes, the server can be changed to another client.
Cheating
It's extremely hard to anticipate how players will cheat and create a cheat-proof system no one can hack. These days, a lot of cheat detection strategies are based on heuristic analysis of logging and behavioral analytics information data to spot abnormalities when they happen and flag it for review. You definitely should try to cheat-proof as much as is reasonable, but you also really need an early detection system that can spot new flaws people are exploiting.
I'm profiling a asp(classic) web service. The web service makes database calls, reads/writes to files, and processes xml. On a windows server 2003 box(2.7ghz, 4 core, 4gb ram) how many requests per second should I be able to handle before things start to fail.
I'm building a tool to test this, but I'm looking for a number of requests per second to shoot for.
I know this is fairly vague, but please give the best estimate you can. If you need more information, please ask.
95% of the performance of any data-driven app is dependent on the database: 1) the way you do your calls, 2) the indexes, 3) the hardware under the database (disk subsystem in particular).
I have seen a machine, like you are describing, handle 40 requests per second (2500/minute), but numbers like 10 per second (600/minute) are more common. I would expect even lower if you are running your DB on the same machine, and even lower still if that DB is SQLExpress or MSAccess.
Also, at capacity, your app will probably not fail, but IIS will Queue requests, once it is saturated, and may timeout some of those requests if it can't service them before the timeout expires.
Btw, instead of building a tool to test your app, you may want to look into using a test tool such as Microsoft WCAT. It is pretty smooth and easy to use.
How fast should it be? Fast enough.
How fast is fast enough? That's a question that only you and your users can answer. If your service is horrifically inefficient and keeps up with demand, it's fast enough. If your service is assembly-optimized, lightning-fast, and overwhelmed with requests, it's not fast enough.
If the server is handling its actual workload, then don't worry about how fast it "should" be. When the server is having trouble, or when you anticipate that it soon will, then you should look at improving the code or upgrading the hardware. Remember Knuth's Law – premature optimization is the root of all evil. Any work you do now to make it faster may never pay off, and you may be forced to make compromises with flexivility or maintainability. Remember, too, an older adage – if it ain't broke, don't fix it.
Yes I would also say 10 per second is a good benchmark. For a high performance app you would want to get more than this, but if you have no specific goal you should generally be able to get at least 10 requests per sec for a general web page with a bunch of database queries.
If you're trying to build an application that needs to have the highest possible sustained network bandwidth, for multiple and repetitive file transfers (not for streaming media), will having 2 or more NICs be beneficial?
I think your answer will depend on your server and network architecture, and unfortunately may change as they change.
What you are essentially doing is trying to remove the 'current' bottleneck in your overall application or design which you have presumably identified as your current NIC (if you haven't actually confirmed this then I would stop and check this in case something else restricts throughput before you reach your NIC limit).
Some general points on this type of performance optimization:
It is worth checking if you have the option to upgrade the current NIC to a higher bandwidth interface - this may be a simpler solution for you if it avoids having to add load balancing hardware/software/configuration to your application.
As pointed out above you need to make sure all the other elements in your network can handle this increased traffic - i.e. that you are not simply going to have congestion in your internet connection or in one of your routers
Similarly, it is worth checking what the next bottle neck will be once you have made this change, if the traffic continues to increase. If adding a new NIC only gives you 5% more throughput before you need a new server anyway, then it may be cheaper to look for a new server right away with better IO from new.
the profile of your traffic and how it is predicted to evolve may influence your decision. If you have a regular daily peak which only exceeds your load slightly then a simple fix may serve you for a long time. If you have steadily growing traffic then a more fundamental look at your system architecture will probably be necessary.
In line with the last point above, it may be worth looking at the various Cloud offerings to see if any meet your requirements at a reasonable cost, possibly even as temporary resource every day just to get you through your peak traffic times.
And finally you should be aware that as soon as you settle on a solution and get it up and running someone else in your organization will change or upgrade the application to introduce a new and unexpected bottle-neck...
It can be beneficial, but it won't necessarily be that way "out of the box".
You need to make sure that both NICs actually get used - by separating your clients on different network segments, by using round robin DNS, by using channel bonding, by using a load balancer, etc. And on top of that you need to make sure your network infrastructure actually has sufficient bandwidth to allow more throughput.
But the general principle is sound - you have less network bandwidth available on your server than disk I/O, so the more network bandwidth you add the better, up until it reaches or exceeds your disk I/O, then it doesn't help you anymore.
Potentially yes. In practice, it also depends on the network fabric, and whether or not network I/O is a bottleneck for your application(s).