The Mongo C Sharp Driver (at least the 1.9.2 version) has a setting for MaxConnectionLifeTime. From looking at the code, it looks like connections are removed from the pool when their age exceeds that lifetime. The default is set to 30 minutes.
Why?
Do connections somehow degrade in performance the more times they are used?
We have received anecdotal reports that in some scenarios connections die after a certain amount of time. This is presumably because some firewall/router along the way is periodically dropping connections that have reached a certain age.
By having the driver periodically close connections and open new ones we can avoid being affected by this.
Most users are not affected by this and could use any value they want for this setting.
Related
I am new to the topic. Having read a handful of articles on it, and asked a couple of persons, I still do not understand what you people do in regard to one problem.
There are UI clients making requests to several backend instances (for now it's irrelevant whether sessions are sticky or not), and those instances are connected to some highly available DB cluster (may it be Cassandra or something else of even Elasticsearch). Say the backend instance is not specifically tied to one or cluster's machines, and instead its every request to DB may be served by a different machine.
One client creates some record, it's synchronously of asynchronously stored to one of cluster's machines then eventually gets replicated to the rest of DB machines. Then another client requests the list or records, the request ends up served by a distant machine not yet received the replicated changes, and so the client does not see the record. Well, that's bad but not yet ugly.
Consider however that the second client hits the machine which has the record, displays it in a list, then refreshes the list and this time hits the distant machine and again does not see the record. That's very weird behavior to observe, isn't it? It might even get worse: the client successfully requests the record, starts some editing on it, then tries to store the updates to DB and this time hits the distant machine which says "I know nothing about this record you are trying to update". That's an error which the user will see while doing something completely legitimate.
So what's the common practice to guard against this?
So far, I only see three solutions.
1) Not actually a solution but rather a policy: ignore the problem and instead speed up the cluster hard enough to guarantee that 99.999% of changes will be replicated on the whole cluster in, say, 0.5 secord (it's hard to imagine some user will try to make several consecutive requests to one record in that time; he can of course issue several reading requests, but in that case he'll probably not notice inconsistency between results). And even if sometimes something goes wrong and the user faces the problem, well, we just embrace that. If the loser gets unhappy and writes a complaint to us (which will happen maybe once a week or once an hour), we just apologize and go on.
2) Introduce an affinity between user's session and a specific DB machine. This helps, but needs explicit support from the DB, and also hurts load-balancing, and invites complications when the DB machine goes down and the session needs to be re-bound to another machine (however with proper support from DB I think that's possible; say Elasticsearch can accept routing key, and I believe if the target shard goes down it will just switch the affinity link to another shard - though I am not entirely sure; but even if re-binding happens, the other machine may contain older data :) ).
3) Rely on monotonic consistency, i.e. some method to be sure that the next request from a client will get results no older than the previous one. But, as I understand it, this approach also requires explicit support from DB, like being able so pass some "global version timestamp" to a cluster's balancer, which it will compare with it's latest data on all machines' timestamps to determine which machines can serve the request.
Are there other good options? Or are those three considered good enough to use?
P.S. My specific problem right now is with Elasticsearch; AFAIK there is no support for monotonic reads there, though looks like option #2 may be available.
Apache Ignite has primary partition for a key and backup partitions. Unless you have readFromBackup option set, you will always be reading from primary partition whose contents is expected to be reliable.
If a node goes away, a transaction (or operation) should be either propagated by remaining nodes or rolled back.
Note that Apache Ignite doesn't do Eventual Consistency but instead Strong Consistency. It means that you can observe delays during node loss, but will not observe inconsistent data.
In Cassandra if using at least quorum consistency for both reads and writes you will get monotonic reads. This was not the case pre 1.0 but thats a long time ago. There are some gotchas if using server timestamps but thats not by default so likely wont be an issue if using C* 2.1+.
What can get funny is since C* uses timestamps is things that occur at "same time". Since Cassandra is Last Write Wins the times and clock drift do matter. But concurrent updates to records will always have race conditions so if you require strong read before write guarantees you can use light weight transactions (essentially CAS operations using paxos) to ensure no one else updates between your read to update, these are slow though so I would avoid it unless critical.
In a true distributed system, it does not matter where your record is stored in remote cluster as long as your clients are connected to that remote cluster. In Hazelcast, a record is always stored in a partition and one partition is owned by one of the servers in the cluster. There could be X number of partitions in the cluster (by default 271) and all those partitions are equally distributed across the cluster. So a 3 members cluster will have a partition distribution like 91-90-90.
Now when a client sends a record to store in Hazelcast cluster, it already knows which partition does the record belong to by using consistent hashing algorithm. And with that, it also knows which server is the owner of that partition. Hence, the client sends its operation directly to that server. This approach applies on all client operations - put or get. So in your case, you may have several UI clients connected to the cluster but your record for a particular user is stored on one server in the cluster and all your UI clients will be approaching that server for their operations related to that record.
As for consistency, Hazelcast by default is strongly consistent distributed cache, which implies that all your updates to a particular record happen synchronously, in the same thread and the application waits until it has received acknowledgement from the owner server (and the backup server if backups are enabled) in the cluster.
When you connect a DB layer (this could be one or many different types of DBs running in parallel) to the cluster then Hazelcast cluster returns data even if its not currently present in the cluster by reading it from DB. So you never get a null value. On updating, you configure the cluster to send the updates downstream synchronously or asynchronously.
Ah-ha, after some even more thorough study of ES discussions I found this: https://www.elastic.co/guide/en/elasticsearch/reference/current/search-request-preference.html
Note how they specifically highlight the "custom value" case, recommending to use it exactly to solve my problem.
So, given that's their official recommendation, we can summarise it like this.
To fight volatile reads, we are supposed to use "preference",
with "custom" or some other approach.
To also get "read your
writes" consistency, we can have all clients use
"preference=_primary", because primary shard is first to get all
writes. This however will probably have worse performance than
"custom" mode due to no distribution. And that's quite similar to what other people here said about Ignite and Hazelcast.
Right?
Of course that's a solution specifically for ES. Reverting to my initial question which is a bit more generic, turns out that options #2 and #3 are really considered good enough for many distributed systems, with #3 being possible to achieve with #2 (even without immediate support for #3 by DB).
We have an issue, more often than I would like, where whether worker or client sessions crash and these sessions were in the process of using a number sequences to create a new record, but they end up blocking that number sequence literally and anyone else trying to create a record using the same sequence will have its client frozen.
When this happens, I usually go in the NUMBERSEQUENCELIST table, I spot the correct DataAreadId and the user, and delete the row whose Status = 1.
But this kind of annoying really. Is there anything, any way I can configure the AOS server to release number sequence when client/workers crash ?
For the worker sessions, I guess we can fine tweak the code which runs in them, but for the client sessions crashing, not much we can do...
Any ideas ?
Thanks!
EDIT: Turns out that in this situation, after restarting the AOS server, you can go in List in the number sequence menu, and clean it up. Prior to the restart, my client would freeze trying to do that. So no need to do it directly through SQL.
Continuous numbers in NumberSequenceList are automatically cleaned up every 24 hours (or as set up on the number sequence). The cleanup process is quite slow if there are many "dead" numbers (hundreds or thousands). This may be considered as a hang, but is not.
Things to consider:
Is a continuous number sequence needed?
Do the cleanup more frequent (say every half hour instead of the default 24 hour)
Setup the cleanup process as a batch process
Fix the bug in the client code using the number sequence
Also avoid reserving the number, just use it. Instead of the anti-pattern:
NumberSeq idSequence = NumberSeq::newGetNum(IntrastatParameters::numRefIntrastatArchiveID(), true);
this.IntrastatArchiveID = idSequence.num();
idSequence.used();
Just use the number:
this.IntrastatArchiveID = NumberSeq::newGetNum(IntrastatParameters::numRefIntrastatArchiveID()).num();
The makeDecisionLater parameter should only be used in forms, where user may decide not to use the number (by delete or by escape). And in that case the NumberSeqFormHandler class should be used anyway.
I have a a Nagios configuration which is performing a number of tests on a few hundred nodes; one of these is a variant of check_http. It's not configured to --enable-embedded-perl (ePN) but we'll be changing that soon. Even with ePN enabled I'm concerned about the model where each execution of this Perl HTTP+SSL check will be handling only a single target.
I'd like to write a simple select() (or poll() / epoll()) driven daemon which creates connections to multiple targets concurrently, reads the results and spits out results in a form that's useable to Nagios as if it were results from a passive check.
Is there a guide to how one could accomplish this? What's the interface or API for providing batched check updates to Nagios?
One hack I'm considering would be to have my daemon update a Redis store (with a key for each target, and a short expiration time) and replace check_http with a very small, lightweight GET of the local Redis instance on the key (the GET would either get the actual results for Nagios or a "(nil)" response which will be treated as if the HTTP connection had timed out.
However, I'm also a bit skeptical of my idea since I'd think someone has already something like this by now.
(BTW: I'm ready to be convinced to switch to something like Icinga or Zabbix or Zenoss or OpenNMS ... pretty much anything that will scale better).
As to whether or not to let Nagios handle the scheduling and checks, I'll leave that to you as it varies depending on your version of Nagios (newer versions can run these checks concurrently), and why you want a separate daemon for it. egarding versioning of Nagios, version 3 IIRC uses concurrent checks, and scales thusly to larger node counts than you report.
However, I can answer the Redis route concept as I've done it with Postfix queue stats and TTFB tracking for web sites.
Setting up the check using Python with the curl and multiprocessing modules is fairly straightforward as is dumping it into Redis. An expiration of I'd say no more than the interval would be a solid idea to keep the DB from growing. I'd recommend tis value be no more (or possibly just less than) the check interval to avoid grabbing stale check results. If the currently running check hasn't completed and the Redis-to-Nagios check runs, pulling in the previous check, you can miss failed checks.
For the Redis-To-Nagios check a simple redis-cli+bash scripting or Python check to pull the data for a given host, returning OK or otherwise depending on your data is fairly simple and would run quickly enough.
I'd recommend running the Redis instance on the Nagios check server to ensure minimum latency and avoid a network issue causing false alerts on your checks. I would also recommend a Nagios check on your Redis instance and the checking daemon. Make the check_http replacement check dependent on the Redis and http_check daemons running. THus you have a dependency chain as follows:
Redis -> http_checkd -> http_check_replacement
This will prevent false alerts on http_check_replacement by identifying the problem. For example, if your redis_checkd dies you get alerted to that, not 200+ "failed http_check_replacement" ones.
Also, since your data in Redis is by definition transient, I would disable the disk persistence. No need to write to disk when the data is constantly rotating.
On a side note, I would recommend, if using libcurl, you pull statistics from libcurl about how long it takes to get the connection open and how long the server to to respond (Time To First Byte - TTFB) and take advantage of Nagios's ability to store check statistics. You may well reach a time when having that data is really handy for troubleshooting and performance analysis.
I have a CLI Tool I've written in C which does this and uploads it into a local Redis instance. It is fast - barely more than the time to get the URL. I'm expecting it be open sourced this week, I can add Nagios style output to it fairly easily. In fact, I think I'll do that in the next week or two.
I have a couple of questions
1) How can we define in weblogic configuration how many concurrent users are allowed or can be allowed at a time to a particular application?
2) how can we tell how may threads are being used in a weblogic at a time?
3) How many max jdbc connections should I set so that users are not blocked due to all connections used up. How to keep a balance between number of concurrent user/threads allowed to jdbc connections max?
Thanks
It is different in each use case scenario.
But usually WLS 1 instance can cover 50~100 active user per instance.
The instance has 2 CPU and 1~1.5GB heap.
This document will be useful to your question:
"Planning Number Of Instance And Thread In Web Application Server"
1) You can user Work Managers to do this for managing requests. However, restricting the number of concurrent users will vary application to application. If it is a web app, use the work managers with a max constraint equal to the number of users you want to restrict it to. However, be sure you figure out how to handle overflow - what will you do when you get 100 requests but have a 5-user restriction? Is this synchronous or asynchronous processing?
2) Ideally you would want a 1:1 ratio of threads to connections in the pool. This guarantees that no thread (User Request) is waiting for a connection. I would suggest trying this. You can monitor the JDBC connection pools using the WebLogic console and adding fields to the columns under the 'Monitoring' tab for the connection. If you have a high number of waiters, and/or a high wait time then you would want to increase the number of connections in the pool. You could start with a 1:0.75 ratio of threads:connections, do performance/load testing and adjust based on your findings. It really depends on how well you manage the connections. Do you release the connection immediately after you get the data from the database, or do you proceed with application logic and release the connection at the end of the method/logic? If you hold the connection for a long time you will likely need closer to a 1:1 ratio.
1) If to each user you assign a session, then you can control the max number of sessions in your webapp weblogic descriptor, for example adding the following constraint :
<session-descriptor> <max-in-memory-sessions>12</max-in-memory-sessions> </session-descriptor>
It's more effective (if you mean 1 user = 1session) than limiting the number of requests by work managers.
Another way, when you can't predict the size of sessions and the number of users, is to adjust memory overloading parameters and set :
weblogic.management.configuration.WebAppContainerMBean.OverloadProtectionEnabled.
More info here :
http://download.oracle.com/docs/cd/E12840_01/wls/docs103/webapp/sessions.html#wp150466
2) Capacity of threads is managed by WebLogic through work managers. By default, just one exists : default with unllimited number of threads (!!!).
3) Usually, adapting the number of JDBC connections to the number of threads is the more effective.
The following page could surely be of great interest :
http://download.oracle.com/docs/cd/E11035_01/wls100/config_wls/overload.html
As far as i know you have to control these kind of things in
weblogic-xml-jar.xml
or
weblogic.xml
if you look for weblogic-xml-jar.xml commands you can find your desire .
I'm currently trying to build an application that inherently needs good time synchronization across the server and every client. There are alternative designs for my application that can do away with this need for synchronization, but my application quickly begins to suck when it's not present.
In case I am missing something, my basic problem is this: firing an event in multiple locations at exactly the same moment. As best I can tell, the only way of doing this requires some kind of time synchronization, but I may be wrong. I've tried modeling the problem differently, but it all comes back to either a) a sucky app, or b) requiring time synchronization.
Let's assume I Really Really Do Need synchronized time.
My application is built on Google AppEngine. While AppEngine makes no guarantees about the state of time synchronization across its servers, usually it is quite good, on the order of a few seconds (i.e. better than NTP), however sometimes it sucks badly, say, on the order of 10 seconds out of sync. My application can handle 2-3 seconds out of sync, but 10 seconds is out of the question with regards to user experience. So basically, my chosen server platform does not provide a very reliable concept of time.
The client part of my application is written in JavaScript. Again we have a situation where the client has no reliable concept of time either. I have done no measurements, but I fully expect some of my eventual users to have computer clocks that are set to 1901, 1970, 2024, and so on. So basically, my client platform does not provide a reliable concept of time.
This issue is starting to drive me a little mad. So far the best thing I can think to do is implement something like NTP on top of HTTP (this is not as crazy as it may sound). This would work by commissioning 2 or 3 servers in different parts of the Internet, and using traditional means (PTP, NTP) to try to ensure their sync is at least on the order of hundreds of milliseconds.
I'd then create a JavaScript class that implemented the NTP intersection algorithm using these HTTP time sources (and the associated roundtrip information that is available from XMLHTTPRequest).
As you can tell, this solution also sucks big time. Not only is it horribly complex, but only solves one half the problem, namely giving the clients a good notion of the current time. I then have to compromise on the server, either by allowing the clients to tell the server the current time according to them when they make a request (big security no-no, but I can mitigate some of the more obvious abuses of this), or having the server make a single request to one of my magic HTTP-over-NTP servers, and hoping that request completes speedily enough.
These solutions all suck, and I'm lost.
Reminder: I want a bunch of web browsers, hopefully as many as 100 or more, to be able to fire an event at exactly the same time.
Let me summarize, to make sure I understand the question.
You have an app that has a client and server component. There are multiple servers that can each be servicing many (hundreds) of clients. The servers are more or less synced with each other; the clients are not. You want a large number of clients to execute the same event at approximately the same time, regardless of which server happens to be the one they connected to initially.
Assuming that I described the situation more or less accurately:
Could you have the servers keep certain state for each client (such as initial time of connection -- server time), and when the time of the event that will need to happen is known, notify the client with a message containing the number of milliseconds after the beginning value that need to elapse before firing the event?
To illustrate:
client A connects to server S at time t0 = 0
client B connects to server S at time t1 = 120
server S decides an event needs to happen at time t3 = 500
server S sends a message to A:
S->A : {eventName, 500}
server S sends a message to B:
S->B : {eventName, 380}
This does not rely on the client time at all; just on the client's ability to keep track of time for some reasonably short period (a single session).
It seems to me like you're needing to listen to a broadcast event from a server in many different places. Since you can accept 2-3 seconds variation you could just put all your clients into long-lived comet-style requests and just get the response from the server? Sounds to me like the clients wouldn't need to deal with time at all this way ?
You could use ajax to do this, so yoǘ'd be avoiding any client-side lockups while waiting for new data.
I may be missing something totally here.
If you can assume that the clocks are reasonable stable - that is they are set wrong, but ticking at more-or-less the right rate.
Have the servers get their offset from a single defined source (e.g. one of your servers, or a database server or something).
Then have each client calculate it's offset from it's server (possible round-trip complications if you want lots of accuracy).
Store that, then you the combined offset on each client to trigger the event at the right time.
(client-time-to-trigger-event) = (scheduled-time) + (client-to-server-difference) + (server-to-reference-difference)
Time synchronization is very hard to get right and in my opinion the wrong way to go about it. You need an event system which can notify registered observers every time an event is dispatched (observer pattern). All observers will be notified simultaneously (or as close as possible to that), removing the need for time synchronization.
To accommodate latency, the browser should be sent the timestamp of the event dispatch, and it should wait a little longer than what you expect the maximum latency to be. This way all events will be fired up at the same time on all browsers.
Google found the way to define time as being absolute. It sounds heretic for a physicist and with respect to General Relativity: time is flowing at different pace depending on your position in space and time, on Earth, in the Universe ...
You may want to have a look at Google Spanner database: http://en.wikipedia.org/wiki/Spanner_(database)
I guess it is used now by Google and will be available through Google Cloud Platform.