Measure perf - components of a roundtrip MDX query - iis

I would like to decompose the performance of a round-trip MDX query from a client to Analysis Services and back. In particular, I'm looking to identify/distinguish individual queries and record the time each query takes for:
the XMLA over HTTP message from client to IIS
the XMLA over TCP/IP message from the Data Pump to Analysis Services
the response from Analysis Services to the Data Pump
the response from IIS to the client
I am open to other data-points that would be beneficial to identify bottlenecks in the lifecycle of a query.
My company has tested a mix of software including: Periodic SSAS DMV data collection, PerfMon, Flight Recorder, Splunk and SQL Sentry. We are having trouble tying it all together.

One of the main problems that you have is that there probably are overlaps in time: msmdpump in IIS can start sending the first bytes to the AS server as soon as it has available the first few bytes of the XMLA from the http request, and vice versa, it probably starts sending the message as soon as the first few bytes from the response from the AS server is available.
Actually, the communication between msmdpump and the AS server is a binary version of the XML that is sent between msmdpump and the client, and hence easy to translate without knowing information later in the message. See http://sqlblog.com/blogs/mosha/archive/2005/12/02/analysis-services-2005-protocol-xmla-over-tcp-ip.aspx for some details about the protocol.
To track the times, my approach would a low level one: I would run Wireshark (http://www.wireshark.org/) on on the computer running IIS, and filter to only the http frames between the client and IIS and the frames between the IIS computer and the AS server. The contents of the frames would be more or less irrelevant, but you could see the time stamp of the first and last package of a request, giving you an rough estimate about the durations of the different communications. And staying on one computer for all network traffic logging avoids the need to have the clocks of all computers exactly synchronized.

Related

Send HTTP request at exact time in future with Nodejs

I need to make POST http request at exact timestamp in future, as accurate as possible, down to milliseconds. But there is network latency as well. How can I achieve such a goal?
setTimeout is not enough here, because it always takes some time resulting in latecomer request due vary network latency. And firing this request before target timestamp may result in early coming request.
My goal is to make request guaranteed came to server after target timestamp, but as soon as possible after it. Could you suggest any solutions with Nodejs?
The best you can do in nodejs (which is not a real-time system) is to do the following:
Premeasure the expected latency so you know about how much to presend the request.
Use setTimeout() to schedule the send at precisely the one-way latency time before your target time. There is no other mechanism in nodejs that would be more precise.
If your request involves a DNS lookup, you can prefetch the TCP address for your hostname and take the DNS lookup time out of your request cycle or at least prime the local DNS cache.
Create a dedicated nodejs program that does nothing else - so its event loop will not be doing anything else at the time the setTimeout() needs to run. You could run this as a child_process from your larger program if desired.
Run a number of tests to see how the timing works and, if you are consistently off by some margin, then adjust your latency offset.
You can develop a regular latency test to determine if the latency changes with time.
As others have said, there is no way to predict what the natural response time will be of the target server (how long it takes to start processing your request from the moment your network packets arrive there). If lots of incoming requests are all racing for the same time slot, then your request will get interleaved in among all the others and served in some order that you do not control.
Other things you can consider. If the target server supports the latest http specifications, then you can have a pre-established http connection with the host (perhaps targeting some other endpoint) that will be kept alive for you to send your precise timing request on. This would take some experimentation to figure out what the target host supports and if this would work.

Expected performance with getstream.io

The getstream.io documentation says that one should expect retrieving a feed in approximately 60ms. When I retrieve my feeds they contain a field named 'duration' which I take is the calculated server side processing time. This value is steadily around 10-40ms, with an average around 15ms.
The problem is, I seldomly get my feeds in less than 150ms and the average time is rather around 200-250ms and sometimes up to 300-400ms. This is the time for the getting the feed alone, no enrichment etc., and I have verified with tcpdump that the network roundtrip is low (around 25ms), and that the time is actually spent waiting for the server to respond.
I've tried to move around my application (eu-west and eu-central) but that doesn't seem to affect things much (again, network roundtrip is steadily around 25ms).
My question is - should I really expect 60ms and continue investigating, or is 200-400ms normal? On the getstream.io site it is explained that developer accounts receive "Low Priority Processing" - what does this mean in practise? How much difference could I expect with another plan?
I'm using the node js low level API.
Stream APIs use SSL to encrypt traffic. Unfortunately SSL introduces additional network I/O. Usually you need to pay for the increased latency only once because Stream HTTP APIs supports HTTP persistent connection (aka keep-alive).
Here's a Wireshark screenshot of the TCP traffic of 2 sequential API requests with keep alive disabled client side:
The 4 lines in red highlight that the TCP connection is getting closed each time. Another interesting thing is that the handshaking takes almost 100ms and it's done twice (the first bunch of lines).
After some investigation, it turns out that the library used to make API requests to Stream's APIs (request) does not have keep-alive enabled by default. Such change will be part of the library soon and is available on a development branch.
Here's a screenshot of the same two requests with keep-alive enabled (using the code from that branch):
This time there is not connection reset anymore and the second HTTP request does not do SSL handshaking.

Designing a message processing system

I have been asked to create a message processing system as following. As I am not sure if this is the right place to post this, feel free to move it to any other appropriate SC group.
Problem
Server have about 100 to 500 clients connected at every moment. When a client connects to server, server loads part of their data and cache it in memory for faster access. Server will receive between 200~1000 messages per second for all clients. These messages are relatively small (about 500 bytes). Any changes to data in cache should be saved to disk as soon as possible. When client disconnects all their data is saved to disk and removed from cache. each message contains some instruction and a text message which will be saved as file. Instructions should be executed as fast as possible (near instant) and all clients using that file should get the update. Only writing the modified message to disk can be delayed.
Here is my solution in a diagram
My solution consists of a web server (http or socket) a message queue and two or more instances of file server and instruction server.
Web server grabs client messages and if there is message available for client in message queue, pushes it back to client.
Instruction processor grabs instructions from queue and creates necessary message to be processed by file server (Get/set file) and waits for the file to be available in queue and more process to create another message for client.
File server only provides the files, either from cache or physical file depending on type of file.
Concerns:
There are peak times that total connected clients might go over 10000 at once and total messages received from clients increase to 10~15K.
I should be able to clear the queue and go back to normal state as soon as possible (with processing requests obviously).
I should be able to add extra instruction processors and file servers on the fly without having to shut down the other instances.
In case file server crashes it shouldn’t lose files so it has to write files to disk as soon as there are any changes and process time is available.
File system should be in b+ tree format so some applications (local reporting apps) could easily access files without having to go through queue server
My Solution
I am thinking of using node.js for socket/web server. And may be a NoSQL database for file server and a queue server such as rabbitMQ or Node_Redis and Redis.
Questions:
Is there a better way of structuring this system?
What are my other options for components of this system?
is it possible to run all the instances in same server machine or even in same application (in different thread)?
You have a couple of holes here, mostly around the web server "pushing" the message back to the client. That doesn't really work in a web-based world. You can try and use websockets, but generally, this ends up being polling based.
I don't know what the "instructions" are to be executed, but saving 1000 500byte messages is trivial. Many NoSQL solutions boast million+ write per second capacity. Especially if you let committing to disk to lag.
Don't bother with the queue for the return of the file. A good NoSQL solution will scale better. Build out a Cassandra cluster, load test it until it can handle your peak load.
This simplifies your architecture into a 1 or more web servers, clients polling that server for file updates, a queue for submitting "messages" to the "instruction server" (also known as an application server in web-developer terms), and a no-sql database for the instruction server to write files to.
This makes scaling easy, you can always add more web servers, and with a decent cluster size for your no-sql server, you should get to scale horizontally there as well. Your only real bottleneck is your instruction server queue, which you could always throw more instruction servers at.

High amount of http read timeouts on azure

When we migrated our apps to azure from rackspace, we saw almost 50% of http requests getting read timeouts.
We tried placing the client both inside and outside azure with the same results. The client in this case is also a server btw, so no geographic/browser issues either.
We even tried increasing the size of the box to ensure azure wasn't throttling. But even using D boxes for a single request, the result was the same.
Once we moved out apps out of azure they started functioning properly again.
Each query was done directly on an instance using a public ip, so no load balancer issues either.
Almost 50% of queries ran into this issue. The timeout was set to 15 minutes.
Region was US East 2
Having 50% of HTTP requests timing out is not normal behavior. This is why you need to analyze what is causing those timeouts by validating the requests are hitting your VM. For this, I would recommend you running a packet capture on your server and analyze response times, as well as look for high number of retransmissions; it is even better if you can take a simultaneous network trace on your clients machines so you can do TCP sequence number analysis and compare packets sent vs received. 
If you are seeing high latencies in the packet capture or high number of retransmissions, it requires detailed analysis. I strongly suggest you to open a support incident so Microsoft support can help you investigate your issue further.

Using Fleck Websocket for 10k simultaneous connections

I'm implementing a websocket-secure (wss://) service for an online game where all users will be connected to the service as long they are playing the game, this will use a high number of simultaneous connections, although the traffic won't be a big problem, as the service is used for chat, storage and notifications... not for real-time data synchronization.
I wanted to use Alchemy-Websockets, but it doesn't support TLS (wss://), so I have to look for another service like Fleck (or other).
Alchemy has been tested with high number of simultaneous connections, but I didn't find similar tests for Fleck, so I need to get some real info from users of fleck.
I know that Fleck is non-blocking and uses Async calls, but I need some real info, cuz it might be abusing threads, garbage collector, or any other aspect that won't be visible to lower number of connections.
I will use c# for the client as well, so I don't need neither hybiXX compatibility, nor fallback, I just need scalability and TLS support.
I finally added Mono support to WebSocketListener.
Check here how to run WebSocketListener in Mono.
10K connections is not little thing. WebSocketListener is asynchronous and it scales well. I have done tests with 10K connections and it should be fine.
My tests shows that WebSocketListener is almost as fast and scalable as the Microsoft one, and performs better than Fleck, Alchemy and others.
I made a test on a Windows machine with Core2Duo e8400 processor and 4 GB of ram.
The results were not encouraging as it started delaying handshakes after it reached ~1000 connections, i.e. it would take about one minute to accept a new connection.
These results were improved when i used XSockets as it reached 8000 simultaneous connections before the same thing happened.
I tried to test on a Linux VPS with Mono, but i don't have enough experience with Linux administration, and a few system settings related to TCP, etc. needed to change in order to allow high number of concurrent connections, so i could only reach ~1000 on the default settings, after that he app crashed (both Fleck test and XSocket test).
On the other hand, I tested node.js, and it seemed simpler to manage very high number of connections, as node didn't crash when reached the limits of tcp.
All the tests where echo test, the servers send the same message back to the client who sent the message and one random other connected client, and each connected client sends a random ~30 chars text message to the server on a random interval between 0 and 30 seconds.
I know my tests are not generic enough and i encourage anyone to have their own tests instead, but i just wanted to share my experience.
When we decided to try Fleck, we have implemented a wrapper for Fleck server and implemented a JavaScript client API so that we can send back acknowledgment messages back to the server. We wanted to test the performance of the server - message delivery time, percentage of lost messages etc. The results were pretty impressive for us and currently we are using Fleck in our production environment.
We have 4000 - 5000 concurrent connections during peak hours. On average 40 messages are sent per second. Acknowledged message ratio (acknowledged messages / total sent messages) never drops below 0.994. Average round-trip for messages is around 150 miliseconds (duration between server sending the message and receiving its ack). Finally, we did not have any memory related problems due to Fleck server after its heavy usage.

Resources