Here is my use-case :
I'me developping Static Site Generator and stuff for them.
I would like to host on static pages (like gitlab-pages or github-pages) some demo of my stuff.
So I can't use server-side tricks to simulate slow connexion.
My demo page should look like this :
see my stuff in normal mode
see my stuff before optimisation with a simulated 56k connection
see my stuff after optimisation with a simulated 56k connection
I've not found any service-worker built for this.
Any Idea where i can found one or how i can build one for this ?
The tc or Traffic Control command should be able to handle the job for you. tc can do a lot but it sounds like you just need to to 'shape` traffic:
SHAPING
When traffic is shaped, its rate of transmission is under control. Shaping may be more than lowering the available bandwidth - it is also used to smooth out bursts in traffic for better network behaviour. Shaping occurs on egress.
Simulating something like cell delay is pretty easy, in the below example I inject a delay of 100 +/- 10ms on interface eth1.
tc qdisc add dev eth1 root netem delay 100ms 10ms
Naturally this is annoying because it is on a primary interface and it does not handle throughput limiting. To do that you need to go into how tc works more (parent/child queues, etc). This writeup explains how to throttle the bandwidth, so if you do that and then add the delay mentioned above, then you should have a pretty solid emulation.
Related
So i started making test streams on my youtube channel using Streamlabs OBS.
I turned on performance mode, looked at the stream but it was like 2 fps.
I looked at task manager and my network usage was NEVER going past 1%, and streamlabs's network usage rarely went past 0.1 mbps.
This happens with other things too, and i don't like it since it makes my internet so slow. Internet (if you're wondering): Verizon Fios 5Ghz connection.
The percentage simply shows the relation between the current network usage and the link speed of your network adapter. For example, if your link speed is 1 Gbps and you're transferring data at 10 Mbps, the network usage will be 1%.
When transferring data over the Internet, the percentage is generally not very useful, because the maximum speed is defined by your ISP and, in your case, is likely to be a lot lower than your adapter speed. Furthermore, your Internet speed can also be degraded by a poor Wi-Fi signal, by other users in the same network, etc.
What you should actually be looking at is the actual speed (in bit/s) at which you are sending data (look at the "Send" field in the performance tab of the task manager) and compare that to your Internet speed (which you can learn by doing a speed test).
I am designing an iOS app for a customer who wants to allow real-time (with minimum lag, max 50ms) conversations between users (a sort of Teamspeak). The lag must be low because the audio can also be live music, played with instruments, so all the users need to synchronize. I need a server, which will request audio recordings to every client and send to others (and make them hear the same sound at the same time).
HTTP is easy to manage/implement and easy to scale, but very low-performing because an average HTTP request takes > 50ms... (with a mid-level hardware), so I was thinking of TCP/UDP connections kept open between clients and server.
But I have some questions:
If I develop the server in Python (using TwistedMatrix, for example), how are its performance ?
I can't develop the server in C++ because it is hard to manage (scalable) and to develop.
Anyone used Nodejs (which is easy to scale) to manage TCP/UDP connections?
If I use HTTP, will it be fast enough with Keep-Alive? Becuase usually the time required for an HTTP Request to be performed is > 50ms (because opening-closing connection is hard), and I want the total procedure to be less than that time.
The server will be running on a Linux machine.
And finally: which type of compression can you suggest me? I thought Ogg Vorbis would be nice, but if there's anything better (and can be used in iOS), I am open to changes.
Thank you,
Umar.
First off, you are not going to get sub 50 ms latency. Others have tried this. See for example http://ejamming.com/ a service that attempts to do what you are doing, but has a musically noticeable delay over the line and is therefore, in the ears of many, completely unusable. They use special routing techniques to get the latency as low as possible and last I heard their service doesn't work with some router configurations.
Secondly, what language you use on server probably doesn't make much difference, as the delay from client to server will be worse than any delay caused by your service, but if I understand your service correctly, you are going to need a lot of servers (or server threads) just relaying audio data between clients or doing some sort of minimal mixing. This is a small amount of work per connection, but a lot of connections, so you need something that can handle that. I would lean towards something like Java, Scala, or maybe Go. I could be wrong, but I don't think this is a good use-case for node, which, as I understand it, does not do multithreading well at this time. Also, don't poo-poo C++, scalable services have been built C++. You could also build the relay part of the service in C++ and the rest in whatever.
Third, when choosing a compression format, you'll have to choose one that can survive packet loss if you plan to use UDP, and I think UDP is the only way to go for this. I don't think vorbis is up to this task, but I could be wrong. Off the top of my head, I'm not sure of anything that works on the iPhone and is UDP friendly, but I'm sure there are lots of things. Speex is an example and is open-source. Not sure if the latency and quality meet your needs.
Finally, to be blunt, I think there are som other things you should research a bit more. eg. DNS is usually cached locally and not checked every http call (though it may depend on the system/library. At least most systems cache dns locally). Also, there is no such protocol as TCP/UDP. There is TCP/IP (sometimes just called TCP) and UDP/IP (sometimes just called UDP). You seem to refer to the two as if they are one. The difference is very important for what you are doing. For example, HTTP runs on top of TCP, not UDP, and UDP is considered "unreliable", but has less overhead, so it's good for streaming.
Edit: speex
What concerns the server, the request itself is not a bottleneck. I guess you have sufficient time to set up the connection, as it happens only in the beginning of the session. Therefore the protocol is not of much relevance.
But consider that HTTP is a stateless protocol and not suitable for audio streaming. There are a couple of real time streaming protocols you can choose from. All of them will work over TCP or UDP (e.g. use raw sockets), and there are plenty of implementations.
In your case, the bottleneck with latency is not the server but the network itself. The connection between an iOS device and a wireless access point (AP) eats up about 40ms if the AP is not misconfigured and connection is good. (ping your iPhone.) In total, you'd have a minimum of 80ms for the path iOS -> AP -> Server -> AP -> iOS. But it is difficult to keep that latency stable. (Typical latency of AirPlay on my local network is about 300ms.)
I think live music over iOS devices is not practicable today. Try skype between two iOS devices and look how close you can get to 50ms. I'd bet no one can do it significantly better, what concerns latency.
Update: New research result!
I have to revise my claims regarding the latency of wifi connections of the iDevice. Apparently when you first ping your device, latency will be bad. But if I ping again no later than 200ms after that, I see an average latency 2ms-3ms between AP and iDevice.
My interpretation is that if there is no communication between AP and iDevice for more than 200ms, the network adapter of the iDevice will go to a less responsive sleep mode, probably to save battery power.
So it seems, live music is within reach again... :-)
Update 2
The ping-interval required for keep alive of low latency apparently differs from device to device. The reported 200ms is for an 3rd gen. iPad. For my iPhone 4 it's more like 50ms.
While streaming audio you probably don't need to bother with this, as data is exchanged on a more frequent basis. In my own context, I have sparse communication between an iDevice and a server, but low latency is crucial. A keep alive therefore is the way to go.
Best, Peter
Any suggestions for components to use as a base for a scalable TCP server? I currently have an implementation that uses Indy which works well for say 100 relatively active connections or 1,000 relatively inactive connections, but the one thread per connection model limits the number of concurrent active connections that can be handled.
Let's say my goal might be 1,000 connections each processing 10 messages per second or 10,000 connections each processing 1 message per second on a good server (8-16 cores). Is this realistic? I'd really like to hear of any real-world implementations because I have found that what might work in theory does not necessarily work in practice and I do not want to be chasing a proposed solution that will not work.
Edit: IOCP would be good, but I only want to use commercial-grade classes/components, so they need to be as "professional" as Indy or IP*Works before I would think of using them. Furthermore, I have no intention of "rolling my own" solution - it would take too much time to make it commercial-grade. Lastly, I am looking for a significant improvement on what I already have. I am sure I can squeeze at least 20-50% more out of what I have (based on Indy), but I am never going to be able to handle 10,000 concurrent clients, or 10,000 messages per second, no matter how hard I try. Whether there is something out there that meets these conditions is another matter.
I have decided to accept the answer referring to the IOCP classes, even though I have not used them, because they look like the best path for investigation at this stage.
There is a project at http://voipobjects.com/ which is based on the former iopcclasses project.
It claims to handle thousands simultaneous connections:
IOCP engine is set of classes, components and routines for rapid
creation high scalable and performance TCP/UDP applications.
Application created using IOCP classes can handle thousands
simultaneous connections.
Library is written in Delphi - Delphi 7 - 2010 are supported.
Library uses IO completion ports technology. There is most powerful
technology in Win32 world for creation highly scalable and performance
TCP/UDP applications. This technology is supported in all desktop
Windows OSes except old Win9x/WinME versions.
This library is licensed under MPL1.1. Also It includes some files
from Jedi project (Winsock2 header translation).
https://bitbucket.org/voipobjects/iocpengine
My favorite Delphi network layer is ICS by Francois Piette. It's fantastically easy to understand, very scalable, and ultra-high performance. Free, and open source. Will probably scale to 1000 clients for most people, without significant effort, and without the complexity that gives me trouble when I use Indy.
I got about a 20% scalability/performance boost from switching all my stuff from Indy to ICS.
You should look at RealThinClient SDK http://www.realthinclient.com/about.htm
Well proven solution. Good support. Test results for different server solutions on the home page.
The real deciding fact is what you plan on doing on each of those transactions.
I use Indy with Network Load Balanced windows servers. One of these Delphi applications is serviced by 3 physical servers listening one public IP address where we have received millions of requests since yesterday with zero errors. Load overnight is pretty idle so the actual requests are around 350/second/server during the day and there's plenty of room for growth.
If there's not a lot of CPU/Memory needed per transaction you might get away with it on one box using Indy. It all depends on the load...as you likely can't write to 1000 different files every second.
There's other items to worry about too - like the OS supporting this amount of activity. You may need to tweak some registry settings. (see this stackoverflow question)
IOCP is the way to go for ultra-capacity servers. I have used Indy for ease of use in implementation/debugging for a very long time. I have my own IOCP implementation that I wrote years ago but never rolled it out on production as we simply haven't needed to.
My simple advice - I'd highly suggest rolling it out in Indy, using NLB as your crutch for load, and after that if you are still desiring the utmost speed, write your own IOCP implementation so you can craft it towards your specific requirements. Note that this is based on knowing nothing of the actual implementation requirements.
I've tried multiple Delphi solutions to do networking, and found that many if not all solutions add complexity and code which impacts on either performance or footprint or both. So I started searching for the lightest wrapper around the winsock API. The I (re)discovered Delphi's own TTcpClient and TTcpServer components. Used in blocking mode, and overriding TCustomTcpServer overriding the DoAccept method, I've had the best results till now.
If you expect to have a really high number of incoming connections and (small) responses to serve to (small) requests, it's highly advisable to implement I/O completion ports, as this handles incoming requests better.
I have been using ICS for the last 12 years. It is non-blocking. I support upto 2000 concurrent connections, each getting atleast 5000 bytes per second, sent as 1000 bytes every 200 msec. Never faced any problem and cpu usage of the app is very small.
Good support in forum, but not required at all.
Shekar
I'm considering using 2 NSStream for up/down channels. However, it looks somewhat complex. If you know simpler way (or recommendations) to do this, please let me know!
-- edit --
This is a kind of fast prototyping of internal/in-house remote controller. Low latency is best, but not required.
Binary formatted data, but not so heavy. Most of them are short control messages, and sometimes big chunks exceptionally.
On Cocoa / Cocoa touch. Platforms are limited to them.
Two peers are on LAN or at least WiFi network. So I can assume the connection is basically fast.
Compatibility to unknown hosts, high efficiency/performance/reliability and such things are no need to be considered. Just simplicity is most important now.
Without knowing acceptable latency, amount of data, type of data, and/or network topology (same LAN? routing over WAN?), it is impossible to say.
For most purposes, HTTP provides an awfully big and versatile hammer. And HTTP is supported by just about everything.
You want simple? Nothing is as simple as HTTP simply because it is a ubiquitous high level protocol that everyone and there brother has implemented anywhere from high level APIs (like NSHTTP*/NSURL*) down to less-than-$1 embedded chips.
If the devices you want to control have an option for an HTTP server, go for that. It'll be dead simple and debugging is much much easier when working with a high level protocol like HTTP.
At this point, it is hard not to buy a device with a LAN/wLAN port that doesn't also have an HTTP server in it (off the top of my head, my home theater receiver, solar controller, bbq, printer, security camera, PS3, VOIP box, and U-verse router all have HTTP servers).
However, the requirements on your non-Cocoa Touch side may dictate otherwise.
I'm currently in the process of performance profiling. We have a basic client/server application. Would the TCP transfer speed be different if I ran client/server on the same machine (localhost) vs across two computers on a LAN?
TCP transfer speed will be! because if you ran it on same computer it will forward packets locally without even touching LAN and network adapter.
But overall speed of client+server may be better on different machines, especially if you do not communicate with server too often.
When using localhost, local resources are more likely to be the performance bottleneck because of memory, disk, cpu, etc. When using two computers, its more likely the network will be the bottleneck because of latency, bandwidth, throughput, packet loss, etc.
It depends on what your application does and how it uses the network, client, and server.
Yes definitely, the latency of sending it across the network would slow the program down. The throughput wouldn't but if you're waiting for replies before sending data then this builds up because of extra latency.
I just hit this issue on a project at work. Using UDP with localhost is at least an order of magnitude faster than over a network connection (maybe two orders of magnitude), and I believe with localhost there is no MTU ceiling of 1500 as normally exists for network ports.
One unconfirmed suspicion is the built in network ports on PCs are not all the same quality, so even if they claim to be gigabit, you may not be able to really go that fast. But it may also be making lots of Windows system calls (one OS call per packet) may be a significant overhead. With TCP I can hand the OS a large chunk of data to write in a single call. With UDP I have to hand it a packet at a time, limited by the MTU size, resulting in a much larger number of OS calls. But unconfirmed as yet.
Have not tried Linux yet.
It really depends on what your application does....
As an example:
If it transfers 10GB files from client to server, then yes, it will make a difference.
I don't know if it measurable (that also depends on your LAN's speed) but from a logical point of view, of course there is a difference. Localhost will always be the fastest as the data has not be sent through another medium (like air or copper wire).
But depending on what your application does, this might or might not matter.
The transfer times would almost certainly be faster if the client and server were on the same machine. That may not actually matter to the performance of your program as a whole depending on the other resources consumed by the client and server.