Testing multiple mac/ip adresses connections - http

I'm building client-server software (some sort of http Rest API)
And there is a throttling module that prevents one IP to do more then N requests/second
I want to do stress testing of the system - so that I can emulate 100-1000 connections from different IP(or MAC) addresses (from hardware I have only boxes in one network)
What tools or scripts can I use for this case ?

You need certainly a traffic generator, try using this one: http://robert.rsa3.com/traffic.html

Related

Need to setup a RMTP stream from our server with multicast

I have a client with a 1-2 thousand viewer audience, with everyday streams, same concurrent number of viewers.
Ive got a server set up for their website etc, but am in the process of figuring out the best way to stream with OBS onto that server, and than re-distribute that stream to clients (as an embed on the website).
Now from the calculations i did, running that kind of concurrent viewers is very problematic, as it forces you into a 10gbit link - which is very expensive, and i would ideally like to fit within 1-2gbps, if possible.
A friend of mine recommended to look into "Multicast" which supossedly uses MUCH less bandwith than regular live streaming options. Is multicast doable? Ive had a NGINX live stream set up on my server by a friend before, but never looked into the config and if multicast is supported within that. Are there any other options? What would you recommend?
Also, the service of that live stream isnt a high profit / organisation type of deal, so any pre-made services just dont make sense, as it would easily cost 40+ dollars per stream, which is just too much for my client.
Thank you for any help!
Tom
Rather than Multicast, P2P is more practical solution on Internet, to save money not bandwidth.
Especially for H5 browser, it's possible to use WebRTC DataChannel to transport P2P data.
But Multicast does not work on internet routers.
Multicast works by sending a single stream across the network to edge points where clients can 'join' the multicast to get an individual stream for them.
It requires that the network supports multicast protocols and the edges align with your users.
It is typically used when an operator has their own IP network for service like IPTV, rather than for services over the internet.
For your scenario, you would usually use an organ server and a CDN - this will usually reduce the load on your own server as the video will be cached on the network and multiple user can access the same 'chunks' of the video.
You can see and AWS example for on demand video here - other vendor and cloud providers have solutions too so this is just an example:
https://docs.aws.amazon.com/AmazonS3/latest/userguide/tutorial-s3-cloudfront-route53-video-streaming.html
You can find more complex On Demand and Live tutorial also but they are likley more that you need: https://aws.amazon.com/cloudfront/streaming/
Exploring P2P may be an option also as Winton suggests - some CDN may also leverage P2P technology internally.

Limits when running zerotier

We want to use zero tier to connect from one cloud machine to multiple remote machines. We do not want remote machines to access each other. What would be a good approach?
Use a single network and set rules based on tags to restrict access
Run multiple networks, each having cloud machine and a remote machine
Are there limits to
Number of members in zerotier network
Number of zerotier networks a machine can connect to at a time - tun interfaces, ip conflicts or performance impact
I would use a single network and use rules to prevent peering between the machines. For instance, you could set the 192.168.141.0/25 portion of the network to prevent peering, and allow only defined network paths between hosts.
Just a personal rant here: You don't want to do that. Really. You're going to make a headache for yourself when you have to scale horizontally (which you will if you're successful). I would STRONGLY recommend taking a mTLS approach to service authentication instead. Somewhat more work at the start, but a lot easier in the long run.

Why some internet providers close certain ports?

We published the game on russian server and 1% of people couldn't connect to server on 46xx port through raw TCP while they can load it's HTML page (through HTTP). Most of such people live in Germany, Israel....
Why is it so? What's the politics decisions lay behind it? We discovered that their such ports (which are free on IANA) are closed. Does it mean that such people cannot run Steam (and, then, play all games which you can buy through it), play WoW and many other modern games which use TCP through 4xxx ports?
Thank you.
ISPs have been known to filter certain ports for various reasons. Users should complain loudly to them (or switch) in order to send a signal that such is not to be tolerated. You can encourage them to do so but of course that doesn't solve your problem (or really answer your question).
Common reasons are:
- trying to block bittorrent traffic
- limit bandwidth usage (largely related to previous reason)
- security (mistaken)
- control (companies often don't want employees goofing off)
The easiest thing for you to do is run your game over port 443 (perhaps as an alternate). That's HTTPS and so will not generally be blocked. However, because HTTPS is encrypted, there's no way to inspect the stream to know if its web traffic or something else and thus you can run any data stream (encrypted or not) that you wish over it.
That's precisely correct. In fact every public web site would by default block all ports except the ones they expect to be running some traffic they would want to.
This is the reason many applications often try to encapsulate their programs to use port 80 which can't be blocked as long as some one wants http traffic to run.
They simply don't want any application that they haven't approved to run through their servers. If you have a sensitive server in public you surely won't want any one to use your machine for any apps that you don't allow. A common reason is applications that eat up bandwidth such as bittorent, edonkey, gnutella as well as streaming, voip and other high bandwidth consuming apps

Can two or more SNMP agents be run on the same port (on the same machine)?

Just a technical question -
Can two or more SNMP agents be run on the same port (on the same machine)?
My first instinct would be no since host:port identifies an instance of an application but I'm not sure.
Thank you!
Technically, if the OS supports it, the SO_REUSEADDR SO_REUSEPORT options may be set on a socket to allow other processes to bind to the same address/port and thus allow multiple processes to receive messages on the same address/port. But both processes would have to set the option, and I doubt any agent implementations do that because it would not make sense to do so--it would just cause headaches having both agents potentially responding to a single request. Managers won't be equipped to handle it.
However, you can instead run an SNMP proxy in the primary address/port, configured to forward requests to one of multiple agents based on query, security, or (with SNMPv3) context/engine ID parameters, and forward responses back.
Also, using AgentX, you have an SNMP master agent running on the primary address/port, and one or more SNMP sub-agents connected to the master agent. The master agent dispatches requests to the sub-agents as appropriate, merging the results into a single response, so that to the outside world it appears as a single agent. Each sub-agent typically handles a different branch of OID space (one sub-agent implementing certain module(s), another sub-agent implementing other module(s)).
But taking two agents intended to own the address/port exclusively, and forcing them to share through the REUSE options, while it may be possible, would not be wise.
You can run multiple agents on the same host and with the same port if they have differents ip address (can use a netsh script for that).
Personnaly I use the nsoftware ddl : SecureSNMP V8 edition .NET to do this.
You can look at this post : Multiple SNMP Agents with nsoftware dll
No, two agents cannot both run on the same port as seperate applications for the reasons you assumed (except with a brittle packet sniffing hack, which we'll not go into).
However, 2 agents can be accessed through the same port if there is some mechanism that handles the actual port and distributes requests based on MIB. For example the Windows SNMP service does this, allowing any number of SNMP agents to be added as "extensions" through the registry (HKEY_LOCAL_MACHINE\SYSTEM\CurrentControlSet\Services\SNMP\Parameters\ExtensionAgents) by writing them as DLLs and using the snmp.h headers in the platform SDK.
You are correct: ports can't be shared.
If both the agents were designed by you, then the answer can be different.
Consider the HTTP and FTP cases, we can use host names to distinguise multiple sites on the same port, then why can't we do it for SNMP?
We can create a dispatcher who monitors port 161 for incoming traffic. Then use multiple real agents to handle those traffic behind. We can feel free to design how to distinguise them. Personally I prefer the FTP virtual host name manner and use | to distinguise agents.
Maybe I can create a demo for #SNMP Suite in the future.
But if you need to work with existing agents on the same server, then such flexibility is lost.

Developing Serverless Lan Chat Program Help!

I want to develop simple Serverless LAN Chat program just for fun. How can I do this ? What type Architecture should I use?
Last year I have worked on TCP,UDP Client/ Server application Project.It was simple (Server listens to certain port/socket and Client connect to server's port etc..) But I have no idea about how to develop "Serverless" LAN Chat program. How can I do this? UDP,TCP,Multicast,Broadcast? or Should program behave like both server and client?
The simplest way would be to use UDP and simply broadcast your messages all over the network.
A little bit more advanced version would be to only use the broadcast to discover other nodes in the network.
Every node maintains a list of known peers.
Messages are sent with TCP to all known peers.
When a node starts up, it sends out an UDP broadcast to discover other nodes.
When a node receives a discovery broadcast, it sends "itself" to the source of the broadcast, in order to make it self known. The receiving node adds the broadcaster to it's own list of known peers.
When a node drops out of the network, it sends another broadcast in order to inform the remaining nodes that they should remove the dropped client from their list.
You would also have to consider handling the dropping out of nodes without them informing the rest of the network.
The spread toolkit may be a bit overkill for what you want, but an interesting starting point.
From the blurb:
Spread is an open source toolkit that provides a high performance messaging service that is resilient to faults across local and wide area networks. Spread functions as a unified message bus for distributed applications, and provides highly tuned application-level multicast, group communication, and point to point support. Spread services range from reliable messaging to fully ordered messages with delivery guarantees.
Spread can be used in many distributed applications that require high reliability, high performance, and robust communication among various subsets of members. The toolkit is designed to encapsulate the challenging aspects of asynchronous networks and enable the construction of reliable and scalable distributed applications.
Spread consists of a library that user applications are linked with, a binary daemon which runs on each computer that is part of the processor group, and various utility and demonstration programs.
Some of the services and benefits provided by Spread:
Reliable and scalable messaging and group communication.
A very powerful but simple API simplifies the construction of distributed architectures.
Easy to use, deploy and maintain.
Highly scalable from one local area network to complex wide area networks.
Supports thousands of groups with different sets of members.
Enables message reliability in the presence of machine failures, process crashes and recoveries, and network partitions and merges.
Provides a range of reliability, ordering and stability guarantees for messages.
Emphasis on robustness and high performance.
Completely distributed algorithms with no central point of failure.
Apples iChat is an example of the very product you are envisioning. It uses Bonjour (apple's zero-conf networking protocol) to identify peers on a LAN. You can then chat or audio/video chat with them.
I'm not entirely sure how Bonjour works inside, but I know it uses multicast. Clients "register" services on the LAN, and the Bonjour protocol allows for each host to pull up a directory of hosts for a given service (all without central management).

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