Iodine is extremely slow - proxy

Finally got iodine working for DNS tunneling but the transfer speed is only 2-3KiB/s.
I testing this on multiple networks and iodine is the bottleneck.
I was expecting something closer to 200-300kbps so is the first figure more accurate or has something gone wrong?

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Docker service poor network performance

I've got a go service (http.ListenAndServe) that simply echoes 'hello world' (the most basic service in order to not introduce overhead to the benchmark). The question is, if I run the service with go run server.go and run a performance test (using wrk https://github.com/wg/wrk) on my local machine (Macbook pro), I got a performance of
Requests/sec: 59336.07
Transfer/sec: 8.66MiB
but if I just run the service in docker (docker run -p8080:8080 periket2000/goku) and run the performance test again, I got:
Requests/sec: 4743.77
Transfer/sec: 0.69MiB
Does it make sense? It's docker so poor in performance?
I've tested it with several services/stacks and got the same results.
The service you describe does extremely little, and from looking at its GitHub page it looks like wrk goes out of its way to avoid things like HTTP persistent connections, so your benchmark basically only tests how fast the combined system can set up and tear down TCP connections. Note that the data rate, even in the loopback case, is really low: there's a lot of handshaking and IP and TCP overhead and HTTP headers before you get to say "hello world" and hang up.
In the loopback case it's highly likely that the MacOS kernel can funnel data almost directly from one process to the other.
Process A |--> Local loopback ----> | Process B
In the Docker case, traffic needs to go over a virtual network interface, into a virtual machine, getting received by a Linux-kernel network interface, traversing a NAT layer into Docker space, and then gets received by the process; replies need to go through this stack in reverse. It's not surprising that this has more overhead, and the tiny packets involved make it worse (see the discussion of connection establishment in the Wikipedia page on TCP; there is waiting for each side to complete the SYN-ACK and FIN-RST sequences and those don't carry data IIRC). So, these results aren't surprising.
Process A |--> Virtual interface
Linux VM eth0
iptables NAT
Linux VM docker0
Container eth0 ----> | Process B
In absolute numbers, in systems I've worked with in the past, getting over 4,000 requests per second definitely gets into the "respectable performance" space. (The loopback-only 8 MB/s data rate doesn't look stellar, because of the nature of the workload.) It might be worth re-running the benchmark with a more realistic workload (mix of GETs and POSTs; non-trivial data responses; some computation or external I/O required to produce results). I wouldn't expect Docker for Mac to hit 100% of the performance of a purely-native no-external-network setup, but I would expect larger data-to-overhead ratios and things like HTTP persistent connections to help.

Simulate slow speed for TCP sockets in Windows

I'm building an application that uses TCP sockets to communicate. I want to test how it behaves under slow-speed conditions.
There are similar question on the site, but as I understand it, they deal with HTTP traffic, or are about Linux. My traffic is not HTTP, just ordinary TCP sockets, and the OS is Windows.
I tried using fiddler's setting for Modem Speed but it didn't work, it seems to work only for HTTP connections.
While it is true that you probably want to invest in an extensive set of unit tests, You can simulate various network conditions using VMWare Workstation:
You will have to install a virtual machine for testing, setup bridged networking (for the vm to access your real network) and upload your code to the vm.
After that you can start changing the settings and see how your application performs.
NetLimiter can also be used, but it has fewer options (in your case, packet loss is very interesting to test and is not available in netlimiter).
There is an excellent utility for Windows that can do throttling and much more:
https://jagt.github.io/clumsy/
I think you're taking the wrong approach here.
You can achieve everything that you need with some well designed unit tests. All of the things that a slow network link causes can be simulated in a unit test environment in controlled conditions.
Things that your code MUST handle to deal with "slow" links are just things that you should be dealing with anyway, including:
The correct handling of fragmented messages. All of your network reading code needs to correctly assume that each read will return between 1 byte and the size of your read buffer. You should never assume that you'll get complete 'messages' as TCP knows nothing of your concept of messages.
TCP flow control causing either your synchronous sends to fail with some form of 'try later' error or your async sends to succeed and potentially use an uncontrolled amount of resources (see here for more details). Note that this can happen even on 'fast' links if you are sending faster than the receiver is consuming.
Timeouts - again this isn't limited to "slow" links. All of your timeout handling code should be robust and tested. You may want to make sure that any read timeout is based on any read completing rather than reading a complete message in x time. You may be getting your data at a slow rate but whilst you're still getting data the link is alive.
Connection failure - again not something specific to "slow" links. You need to know how you deal with connections being reset at any time.
In summary nothing you can achieve by running your client and server on a simulated slow network cannot be achieved with a decent set of unit tests and everything that you would want to test on such a link is something that could affect any of your connections on any speed of link.

Monitoring office internet connection for drop outs in Ruby

I am looking for a simple way to monitor our office internet connection for drop outs. A secondary pipe dream is to also monitor for other 'dodgy' behaviour - packet loss, jitter etc. But the primary goal is to watch for dropped connections. Pinging Google every second is great to keep an eye on latency but we have had a few temporary blips which have caused hell with a few streaming services but have not affected connection latency. The IT department also sometimes decide to block outgoing ICMP traffic which doesn't help with the humble ping tool's efforts.
If this is not something available already via an open source, freeware or commercial tool, ideally I would like to be able to come up with something in Ruby (or, if forced, .NET) which will open a 'long' TCP connection to an arbitrary web server on port 80 (i.e. I don't want to have to write something keeping a socket open on a hosted server) and have the program detect and alert the guys in the office if the connection drops out in a "bad" way. With my attempts using Ruby Socket (http://www.ruby-doc.org/stdlib-1.9.3/libdoc/socket/rdoc/Socket.html) I've had trouble extracting an accurate error code here; ideally I want to isolate actual network connectivity issues from the usual connection timeouts. On a timeout, I'll want to restart the connection silently, but on a real drop out, I'll flash something big and obvious up on screen to alert the guys in the office.
I've spent most of the day googling for examples of this kind of monitoring and trying to hack something together but it seems that it is not a common request. 99% of results are forum posts ending with me being authoritatively informed that speedtest.net will do everything I need. My own attempts have all proven futile - no matter which way I've tried, whenever I seem to be getting somewhere even the most basic drop out test (unplugging the network cable from my laptop!) fails to be detected.
Is this something trivial, and if so could anyone point me in the right direction please? Or am I in for a world of pain? (This has been my general experience whenever I've tried to do anything with network programming in the past...)
Alternatively is there anything pre-written (free, commericial, open source all fine) which will do just this?
Thanks!
Smokeping might do what you want. Nagios might as well.
http://oss.oetiker.ch/smokeping/
http://www.nagios.org/

How slow are TCP sockets compared to named pipes on Windows for localhost IPC?

I am developing a TCP Proxy to be put in front of a TCP service that should handle between 500 and 1000 active connections from the wild Internet.
The proxy is running on the same machine as the service, and is mostly-transparent. The service is for the most part unaware of the proxy, the only exception being the notification of the real remote IP address of the clients.
This means that, for every inbound open TCP socket, there are two more sockets on the server: the secondth of the pair in the Proxy, and the one on the real service behind the proxy.
The send and recv window sizes on the two Proxy sockets are set to 1024 bytes.
What are the performance implications on this? How slow is this configuration? Should I put some effort on changing the service to use Named Pipes (or other IPC mechanism), or a localhost TCP socket is for the most part an efficient IPC?
The merge of the two apps is not an option. Right now we are stuck with the two process configuration.
EDIT: The reason for having two separate process on the same hardware is 100% economics. We have one server only, and we are not planning on getting more (no money).
The TCP service is a legacy software in Visual Basic 6 which grew beyond our expectations. The proxy is C++. We don't have the time, money nor manpower to rewrite and migrate the VB6 code to a modern programming environment.
The proxy is our attempt to mitigate a specific performance issue on the service, a DDoS attack we are getting from time to time.
The proxy is open source, and here is the project source code.
It will be the same (or at least not measurably different). Winsock is smart enough to know if it's talking to a socket on the same host and, in that case, it will short-circuit pretty much everything below IP and copy data directly buffer-to-buffer. In terms of named pipes vs. sockets, if you need to potentially be able to communicate to different machines ever in the future, choose sockets. If you know for a fact that you'll never need to do that, pick whichever one your developers are most familiar or most comfortable with.
For anyone that comes to read this later, I want to add some findings that answer the original question.
For a utility we are developing we have a networking class that can use named pipes, or TCP with the same calls.
Here is a typical loop back file transfer on our test system:
TCP/IP Transfer time: 2.5 Seconds
Named Pipes Transfer time: 3.1 Seconds
Now, if you go outside the machine and connect to a remote computer on your network the performance for named pipes is much worse:
TCP/IP Transfer time: 12 Seconds
Named Pipes Transfer time: 2.5 Minutes (Yes Minutes!)
I realize that this is just one system (Windows 7) But I think it is a good indicator of how slow named pipes can be...and it seems like TCP is the way to go.
I know this topic is very old, but it was still relevant for me, and maybe others will look at this in the future as well.
I implemented IPC between Excel (VBA) and another process on the same machine, both via a TCP connection as well as via Named Pipes.
In a quick performance test, I submitted a message than consisted of 26 bytes from client (Excel) to server (not Excel), and waited for the reply message from the other process (which consisted of 12 bytes in the example).
I executed this a ton of times in a loop and measured the average execution time.
With TCP on localhost (Windows 7, no fastpath), one "conversation" (request+reply) took around 300-350 microseconds. Especially sending data was quite slow (sending the 26 bytes took around 200microseconds via TCP).
With Named Pipes, one conversation took around 60 microseconds on average - so a LOT faster.
I'm not entirely sure why the difference was so large. The corporate environment I tested this in has a strict firewall, package inspections and what not, so I THINK this may have been caused as even the localhost-based TCP connection went through security measures significantly slowing it down, while named pipe ones likely did not.
TL:DR: In my case, Named Pipes were around 5-6 times faster than TCP for small packages (have not tested with bigger ones yet)
http://msdn.microsoft.com/en-us/library/aa178138(v=sql.80).aspx
Let me sum it up for you. If you are worried about performance then use TCP/IP. But if you have a really fast network and your not worried about performance then Named Pipes would be "neat" in that it might save you some code.
Not to mention, if you stick to TCP then you will have something that can be scaled, and even load balanced when the time comes.
Cheers,
In the scenario you describe, the local TCP connections are very unlikely to be a bottleneck. It will introduce some overhead, of course, but this should be negligible unless your CPU is already running hot.
At a guess, if your server's CPU usage is normally below 50% or so (with the proxy in place) it isn't worth worrying about minimizing the overhead associated with the local TCP connections.
If CPU usage is regularly above 80% you should probably be doing some profiling. I'd start by comparing the CPU load (or, better still, the performance, if you can measure it meaningfully) when the proxy is in place to when it isn't. Unless the proxy is doing some complicated processing, the overhead associated with the extra TCP connections is probably a significant fraction of the total overhead introduced by the proxy, so that should give you at least an order-of-magnitude estimate of how much you'd gain by using a more efficient form of IPC.
What is the reason to have a proxy on the SAME machine, just curious?
Anyway:
There are several methods for IPC, TCP/IP, named Pipes are comparable in speed and complexity. If you really want something that scales well and has almost no overhead: use shared memory. Best used in combination with a lock free algorithm for advancing the pointers (or use one buffer for each reader (the proxy/the service) and writer(the service/the proxy)).

Bittorrent protocol 'not available'/'end connection' response?

I like being able to use a torrent app to grab the latest TV show so that I can watch it at my lesiure. The problem is that the structure of the protocol tends to cause a lot of incoming noise on my connection for some time after I close the client. Since I also like to play online games sometimes this means that I have to make sure that my torrent client is shut off about an hour (depending on how long the tracker advertises me to the swarm) before I want to play a game. Otherwise I get a horrible connection to the game because of the persistent flood of incoming torrent requests.
I threw together a small Ruby app to watch the incoming requests so I'd know when the UTP traffic let up:
http://pastebin.com/TbP4TQrK
The thought occurred to me, though, that there may be some response that I could send to notify the clients that I'm no longer participating in the swarm and that they should stop sending requests. I glanced over the protocol specifications but I didn't find anything of the sort. Does anyone more familiar with the protocol know if there's such a response?
Thanks in advance for any advice.
If a bunch of peers on the internet has your IP and think that you're on their swarm, they will try to contact you a few times before giving up. There's nothing you can do about that. Telling them to stop one at a time will probably end up using more bandwidth that just ignoring the UDP packets would.
Now, there are a few things you can do to mitigate it though:
Make sure your client sends stopped requests to all its trackers. This is part of the protocol specification and most clients do this. If this is successful, the tracker won't tell anyone about you past that point. But peers remember having seen you, so it doesn't mean nobody will try to connect to you.
Turn off DHT. The DHT acts much like a tracker, except that it doesn't have the stopped message. It will take something like 15-30 minutes for your IP to time out once it's announced to the DHT.
I think it might also be relevant to ask yourself if these stray incoming 23 byte UDP packets really matter. Presumably you're not flooded by more than a few per second (probably less). Have you made any actual measurements or is it mostly paranoia to wait for them to let up?
I'm assuming you're playing some latency sensitive FPS, in which case the server will most likely blast you with at least 10-50 full MTU packets per second, without any congestion control. I would be surprised if you attract so many bittorrent connection attempts that it would cause any of the game packets to be dropped.

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