Many-to-many messaging on local machine without broker - windows

I'm looking for a mechanism to use to create a simple many-to-many messaging system to allow Windows applications to communicate on a single machine but across sessions and desktops.
I have the following hard requirements:
Must work across all Windows sessions on a single machine.
Must work on Windows XP and later.
No global configuration required.
No central coordinator/broker/server.
Must not require elevated privileges from the applications.
I do not require guaranteed delivery of messages.
I have looked at many, many options. This is my last-ditch request for ideas.
The following have been rejected for violating one or more of the above requirements:
ZeroMQ: In order to do many-to-many messaging a central broker is required.
Named pipes: Requires a central server to receive messages and forward them on.
Multicast sockets: Requires a properly configured network card with a valid IP address, i.e. a global configuration.
Shared Memory Queue: To create shared memory in the global namespace requires elevated privileges.
Multicast sockets so nearly works. What else can anyone suggest? I'd consider anything from pre-packaged libraries to bare-metal Windows API functionality.
(Edit 27 September) A bit more context:
By 'central coordinator/broker/server', I mean a separate process that must be running at the time that an application tries to send a message. The problem I see with this is that it is impossible to guarantee that this process really will be running when it is needed. Typically a Windows service would be used, but there is no way to guarantee that a particular service will always be started before any user has logged in, or to guarantee that it has not been stopped for some reason. Run on demand introduces a delay when the first message is sent while the service starts, and raises issues with privileges.
Multicast sockets nearly worked because it manages to avoid completely the need for a central coordinator process and does not require elevated privileges from the applications sending or receiving multicast packets. But you have to have a configured IP address - you can't do multicast on the loopback interface (even though multicast with TTL=0 on a configured NIC behaves as one would expect of loopback multicast) - and that is the deal-breaker.

Maybe I am completely misunderstanding the problem, especially the "no central broker", but have you considered something based on tuple spaces?
--
After the comments exchange, please consider the following as my "definitive" answer, then:
Use a file-based solution, and host the directory tree on a Ramdisk to insure good performance.
I'd also suggest to have a look at the following StackOverflow discussion (even if it's Java based) for possible pointers to how to manage locking and transactions on the filesystem.
This one (.NET based) may be of help, too.

How about UDP broadcasting?

Couldn't you use a localhost socket ?
/Tony

In the end I decided that one of the hard requirements had to go, as the problem could not be solved in any reasonable way as originally stated.
My final solution is a Windows service running a named pipe server. Any application or service can connect to an instance of the pipe and send messages. Any message received by the server is echoed to all pipe instances.
I really liked p.marino's answer, but in the end it looked like a lot of complexity for what is really a very basic piece of functionality.
The other possibility that appealed to me, though again it fell on the complexity hurdle, was to write a kernel driver to manage the multicasting. There would have been several mechanisms possible in this case, but the overhead of writing a bug-free kernel driver was just too high.

Related

Automatic reconnect in case of network failures

I am testing .NET version of ZeroMQ to understand how to handle network failures. I put the server (pub socket) to one external machine and debugging the client (sub socket). If I stop my local Wi-Fi connection for seconds, then ZeroMQ automatically recovers and I even get remaining values. However, if I disable Wi-Fi for longer time like a minute, then it just gets stuck on a frame waiting. How can I configure this period when ZeroMQ is still able to recover? And how can I reconnect manually after, say, several minutes? How can I understand that the socket is locked and I need to kill/open again?
Q :" How can I configure this ... ?"
A :Use the .NET versions of zmq_setsockopt() detailed parameter settings - family of link-management parameters alike ZMQ_RECONNECT_IVL, ZMQ_RCVTIMEO and the likes.
All other questions depend on your code.
If using blocking-forms of the .recv()-methods, you can easily throw yourself into unsalvageable deadlocks, best never block your own code ( why one would ever deliberately lose one's own code domain-of-control ).
If in a need to indeed understand low-level internal link-management details, do not hesitate to use zmq_socket_monitor() instrumentation ( if not available in .NET binding, still may use another language to see details the monitor-instance reports about link-state and related events ).
I was able to find an answer on their GitHub https://github.com/zeromq/netmq/issues/845. Seems that the behavior is by design as I got the same with native zmq lib via .NET binding.

Is there a way asterisk reconnect calls when internet connection is missed

For being specific, I am using asterisk with a Heartbeat active/pasive cluster. There are 2 nodes in the cluster. Let's suppose Asterisk1 Asterisk2. Eveything is well configured in my cluster. When one of the nodes looses internet connection, asterisk service fails or the Asterisk1 is turned off, the asterisk service and the failover IP migrate to the surviving node (Asterisk2).
The problem is if we actually were processing a call when the Asterisk1 fell down asterisk stops the call and I can redial until asterisk service is up in asterisk2 (5 seconds, not a bad time).
But, my question is: Is there a way to make asterisk work like skype when it looses connection in a call? I mean, not stopping the call and try to reconnect the call, and reconnect it when asterisk service is up in Asterisk2?
There are some commercial systems that support such behavour.
If you want do it on non-comercial system there are 2 way:
1) Force call back to all phones with autoanswer flag. Requerment: Guru in asterisk.
2) Use xen and memory mapping/mirror system to maintain on other node vps with same memory state(same running asterisk). Requirment: guru in XEN. See for example this: http://adrianotto.com/2009/11/remus-project-full-memory-mirroring/
Sorry, both methods require guru knowledge level.
Note, if you do sip via openvpn tunnel, very likly you not loose calls inside tunnel if internet go down for upto 20 sec. That is not exactly what you asked, but can work.
Since there is no accepted answer after almost 2 years I'll provide one: NO. Here's why.
If you failover from one Asterisk server 1 to Asterisk server 2, then Asterisk server 2 has no idea what calls (i.e. endpoint to endpoing) were in progress. (Even if you share a database of called numbers, use asterisk realtime, etc). If asterisk tried to bring up both legs of the call to the same numbers, these might not be the same endpoints of the call.
Another server cannot resume the SIP TCP session of the other server since it closed with the last server.
The MAC source/destination ports may be identical and your firewall will not know you are trying to continue the same session.
etc.....
If you goal is high availability of phone services take a look at the VoIP Info web site. All the rest (network redundancy, disk redundancy, shared block storage devices, router failover protocol, etc) is a distraction...focus instead on early DETECTION of failures across all trunks/routes/devices involved with providing phone service, and then providing the highest degree of recovery without sharing ANY DEVICES. (Too many HA solutions share a disk, channel bank, etc. that create a single point of failure)
Your solution would require a shared database that is updated in realtime on both servers. The database would be managed by an event logger that would keep track of all calls in progress; flagged as LINEUP perhaps. In the event a failure was detected, then all calls that were on the failed server would be flagged as DROPPEDCALL. When your fail-over server spins up and takes over -- using heartbeat monitoring or somesuch -- then the first thing it would do is generate a set of call files of all database records flagged as DROPPPEDCALL. These calls can then be conferenced together.
The hardest part about it is the event monitor, ensuring that you don't miss any RING or HANGUP events, potentially leaving a "ghost" call in the system to be erroneously dialed in a recovery operation.
You likely should also have a mechanism to build your Asterisk config on a "management" machine that then pushes changes out to your farm of call-manager AST boxen. That way any node is replaceable with any other.
What you should likely have is 2 DB servers using replication techniques and Linux High-Availability (LHA) (1). Alternately, DNS round-robin or load-balancing with a "public" IP would do well, too. These machine will likely be light enough load to host your configuration manager as well, with the benefit of getting LHA for "free".
Then, at least N+1 AST Boxen for call handling. N is the number of calls you plan on handling per second divided by 300. The "+1" is your fail-over node. Using node-polling, you can then set up a mechanism where the fail-over node adopts the identity of the failed machine by pulling the correct configuration from the config manager.
If hardware is cheap/free, then 1:1 LHA node redundancy is always an option. However, generally speaking, your failure rate for PC hardware and Asterisk software is fairly lower; 3 or 4 "9s" out of the can. So, really, you're trying to get last bit of distance to the "5th 9".
I hope that gives you some ideas about which way to go. Let me know if you have any questions, and please take the time to "accept" which ever answer does what you need.
(1) http://www.linuxjournal.com/content/ahead-pack-pacemaker-high-availability-stack

Multi-client RPC routing daemon

I've spent a few weeks looking for a daemon that is capable of the following:
Handling arbitrary numbers of clients from other network hosts.
Allowing clients to register services under an identity.
Allowing other clients to make RPC calls to these registered services, with reliable error handling (if client registered for that service detaches, the daemon should report an error to the caller, not leave it hanging).
Allowing clients to register to receive events, which other clients can broadcast.
The simpler, the better.
The closest thing I've found in the F/LOSS world is D-Bus, which meets all of these criteria except the first: it cannot reliably work with remote connections. I have spent time looking into other options (ESB and MQ daemons -- spending the most time with 0MQ and RabbitMQ) and they all fall short in one way or another: ESBs tend to be extremely complicated with a high learning curve, and MQ daemons tend to provide half of the solution (routing) while leaving the other half (error recovery) extremely complicated, if not impossible.
I'm not looking for any fantastic routing capabilities beyond one client saying "I provide service 'foo'" and another client saying "I want to invoke method 'bar' on service 'foo'."
It seems like something like this would exist already, and I hesitate to roll my own.
The use case is that I will have many processes across many hosts. Each host will have a governor process/service that will provide control of the lifetimes of these processes from a control panel. The processes themselves will also be services, allowing for direct inquiry about status as well as reconfiguration requests from this panel. The trick is that processes will come and go, and so static configuration of endpoints isn't something I really want to bother with; I would rather have each process be responsible for telling a daemon "I'm service so-and-so" and then let the daemon do the inter-client routing.
The only piece I'm really missing to develop this system is something that can route RPC requests between all of these processes. Is there any such daemon out there, or is there some other model that would better fit my needs?

Best way to communicate from KEXT to Daemon and block until result is returned from Daemon

In KEXT, I am listening for file close via vnode or file scope listener. For certain (very few) files, I need to send file path to my system daemon which does some processing (this has to happen in daemon) and returns the result back to KEXT. The file close call needs to be blocked until I get response from daemon. Based on result I need to some operation in close call and return close call successfully. There is lot of discussion on KEXT communication related topic on the forum. But they are not conclusive and appears be very old (year 2002 around). This requirement can be handled by FtlSendMessage(...) Win32 API. I am looking for equivalent of that on Mac
Here is what I have looked at and want to summarize my understanding:
Mach message: Provides very good way of bidirectional communication using sender and reply ports with queueing mechansim. However, the mach message APIs (e.g. mach_msg, mach_port_allocate, bootstrap_look_up) don't appear to be KPIs. The mach API mach_msg_send_from_kernel can be used, but that alone will not help in bidirectional communication. Is my understanding right?
IOUserClient: This appears be more to do with communicating from User space to KEXT and then having some callbacks from KEXT. I did not find a way to initiate communication from KEXT to daemon and then wait for result from daemon. Am I missing something?
Sockets: This could be last option since I would have to implement entire bidirectional communication channel from KEXT to Daemon.
ioctl/sysctl: I don't know much about them. From what I have read, its not recommended option especially for bidirectional communication
RPC-Mig: Again I don't know much about them. Looks complicated from what I have seen. Not sure if this is recommended way.
KUNCUserNotification: This appears to be just providing notification to the user from KEXT. It does not meet my requirement.
Supported platform is (10.5 onwards). So looking at the requirement, can someone suggest and provide some pointers on this topic?
Thanks in advance.
The pattern I've used for that process is to have the user-space process initiate a socket connection to the KEXT; the KEXT creates a new thread to handle messages over that socket and sleeps the thread. When the KEXT detects an event it needs to respond to, it wakes the messaging thread and uses the existing socket to send data to the daemon. On receiving a response, control is passed back to the requesting thread to decide whether to veto the operation.
I don't know of any single resource that will describe that whole pattern completely, but the relevant KPIs are discussed in Mac OS X Internals (which seems old, but the KPIs haven't changed much since it was written) and OS X and iOS Kernel Programming (which I was a tech reviewer on).
For what it's worth, autofs uses what I assume you mean by "RPC-Mig", so it's not too complicated (MIG is used to describe the RPC calls, and the stub code it generates handles calling the appropriate Mach-message sending and receiving code; there are special options to generate kernel-mode stubs).
However, it doesn't need to do any lookups, as automountd (the user-mode daemon to which the autofs kext sends messages) has a "host special port" assigned to it. Doing the lookups to find an arbitrary service would be harder.
If you want to use the socket established with ctl_register() on the KExt side, then beware: The communication from kext to user space (via ctl_enqueuedata()) works OK. However opposite direction is buggy on 10.5.x and 10.6.x.
After about 70.000 or 80.000 send() calls with SOCK_DGRAM in the PF_SYSTEM domain complete net stack breaks with disastrous consequences for complete system (hard turning off is the only way out). This has been fixed in 10.7.0. I workaround by using setsockopt() in our project for the direction from user space to kext as we only send very small data (just to allow/disallow some operation).

How can I create a reliable and fast network daemon with Ruby?

I am trying to create a Ruby daemon process which clients will be able to connect to.
I need to ensure that the remote Ruby process always remains up and available for connection, so I need to detect network outages or unreachable errors.
I was thinking of having a heartbeat mechanism at the application level between clients and the server, and a timeout in the client if the connection fails.
I was told the select method in Ruby could be of help as well but not sure.
Can anyone share any good links/resources or impart some general wisdom to create reliable and fast daemon processes in Ruby?
I think a lot of people would use eventmachine for this type of application. At its core, it uses epoll (which is similar to select) to decide which socket to deal with next. There are lots of gems that build on eventmachine to allow you to run different types of servers. One example is em-websocket.

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