We are using Foxglove as a visualization tool for our ROS2 Foxy system on Ubuntu 20, but we are running into bandwidth issues with the rosbridge websocket. We have plans to switch to using the foxglove_bridge websockets since they advertise performance improvement, but are waiting until we migrate to ROS Humble.
When a client initiates a subscription to a topic, it can also pass along options to the server to throttle the message rate for each topic.
Where do I change those options? They must be set within the client, but I couldn't find anything within the GUI to set it.
I'm running foxglove-studio from binaries installed through apt. The only source code I have for the foxglove-studio is a few custom extension panels.
My temporary fix is to filter out the topics I want to throttle and hard-code the throttle_rate option within the rosbridge server before the options are passed to the subscriber handler.
This will work for the demo we have coming up, but I'm searching for a better solution.
Foxglove Studio currently uses a hard-coded set of parameters for creating the roslib Topic object and so does not support throttling. To achieve this you'd currently need to either:
patch the Studio source and build it yourself
patch the server as you've currently done, or
create a separate downsampled topic (e.g. using topic_tools/throttle).
Related
I am trying to come up with a monitoring solution for MITRE ATT&CK Technique T1115 (Clipboard Data). The data can be retrieved via Powershell (Get-Clipboard) or by using the Windows API (OpenClipboard() or GetClipboardData). Scriptblock logging will allow me to detect the Powershell use, but how can you monitor for those specific API calls?
I have not been able to come up with a solution to track specific API calls. The deepest I can drill is down to the process level, but tracking specific API calls is a mystery to me.
I am trying to write a small script that will help me automate some of my IT tasks regarding to VLAN management.
I do not want to log-in to my switch via command-line - I want to send commands to it and get response (over the NET).
Are there any alternatives? I have started to search the web but so far I did not found anything.
I know SNMP is an option to gain info but I want to check other alternatives
thanks.
You can try Netconf Configuration Protocol, it is RPC-like management protocol which is supported by Cisco and many other vendors.
SNMP is the only widely and commonly used option here.
You can use WMI to manage Windows-based infrastructure.
There is also legacy SYSLOG protocol (RFC3164) which is UDP based.
For traffic monitoring and billing purposes there are NetFlow,
sFlow, jFlow, IPFIX and RADIUS protocols.
There are some other protocols but mostly proprietary.
So I'd suggest using SNMP which is nowadays a de-facto standard in network monitoring domain.
You might look at Expect as a scripting language solution. It is commonly used to do exactly what you are needing:
log into device (with result cases)
execute commands
save config
logout
As you build out a script library, tasks become simplified as you could do things like run scripts with parameters and have Expect do all the detail work.
See the wikipedia article for an overview.
I have also used SNMP for this kind of thing but the functionality is different because you are using an SNMP read-write privilege to upload new parts or complete configs, saving the running config to flash and/or saving the config off-device.
Try NETCONF+YANG protocol because it is currently the best option for network device configuration. More about SNMP alternatives:
https://bestmonitoringtools.com/top-snmp-alternatives-because-snmp-is-dying/
I want to implement a peer-to-peer video chat feature for a web application I am currently developing. After doing my research, I've decided that using webRTC's Javascript APIs is the way to go. The application uses AngularJS in the front end and Ruby on Rails in the back end. The main issue I'm encountering while conceptualizing this application is linking the front end with the backend, and creating and maintaining the connection between user streams.
For the signaling aspect of the network, I want to utilize ActionController::Live and the Ruby gem em-event source to push live messages from the server to users and indicate which of their connections are online. Then, when they are ready to make a connection, they will create a custom room and the URL will be sent to the user that they wish to connect with, creating their offer. Once the user clicks on the link sent to them, they send back their answer. When the user responds, the ICE candidate process will begin for each of the users. Do you think that this is a sufficient signaling channel to set up the PeerConnection? What other major players am I missing?
From the research that I have done about WebRTC's RTCPeerConnection, once the initial connection is set up, and both users have public IP addresses corresponding to their stream, the connection is sustained through RTCPeerConnection, more specifically getPeerConnection(). Am I wrong? Are there other factors that I am not considering?
WebRTC makes the process of creating MediaStreams very simple with their getUserMedia method. Once these streams are created they can be added to the RTCPeerConnection that was established. Both as local and remote streams.
If you have any other suggestions for me, please let me know. I want to create this feature using webRTC, it seems like so much fun
There are certainly many ways to handle the call signaling so I'm not going to comment specifically on your approach. I will say that if you plan on supporting ICE trickling the ICE candidates will start flowing very early in the process so you really need an open signalling channel between your peers almost immediately when trying to connect to a peer.
We developed our solution for WebSphere on top of MQTT which is an open, and very simple pub/sub protocol. You can use any open MQTT broker with the protocol and there are a number of open source components available to make WebRTC development extremely easy including an AngularJS WebRTC module (angular-rtcomm), a core pure JavaScript module and much more. We also released a simple JSON based protocol as part of this open source solution. You can take a look at the signaling protocol. You can also read more details about the overall solution here (www.wasdev.net/webrtc). Here you'll find the base JavaScript libraries as well as a number of open source sample solutions. All of these can be forked on github.
In general you want to build your signaling on a protocol that will allow you to grow over time. It should work well for the web and mobile apps. From our experience it took a lot of time to get all this to work well and our goal was to not only support peer-to-peer calls but to support using media resources like Dialogic's XMS PowerMedia server on the backend for multiway support, record/playback and more. We also needed to support federation via SIP trunking so we wanted to make sure the protocol could be easily translated to SIP signaling while also supporting transcoding between media protocols like VP8 and H.264.
Note that if you're looking to only support peer-to-peer calling between WebRTC clients you can do that with these rtcomm open source components only, including an open MQTT broker and save yourself a ton of time. You can literally get something up and running in a matter of hours. The developer version of the WebSphere Liberty beta with the new rtcomm-1.0 service enabled also includes a built in MQTT broker and supports the open WebRTC signaling protocol linked above. You can use WebSphere for development and deploy a single server of this in production for free. You can also use Ruby on Rails with Liberty as well if you'd like.
Even if you decide not to use Liberty you can use all the open source components along with something like Mosquito (which is an open source MQTT broker) to get a solution off the ground quickly. There are also a number of MQTT clients available for many different programming languages including JavaScript, Java, etc. Check out https://eclipse.org/paho/. If you decide to build you're own signaling protocol you might still find these open source components helpful to see how we approached integration with the WebRTC PeerConnection.
I am sorry if this question goes out of topic but i forced to ask here as there is very limited resources found over the net on this.
I am looking to implement system to get real time data from avaya cms server I did lot of RND on JTAPI but it has got some limitations it is not giving all events all data as stored in CMS database. I also tried connecting cms database using Java but no success because it also give historical data in delay of 30 mins.
Is it possible to get the same technically using JTAPI,TAPI anything. Or is there anyone who have used any paid tool by avaya which is cheaper and can solve this purpose.
I saw clint but don't intend to use. Please let me know the ways if anyone had done this.
Your CMS may provide a feature known to me as realtime socket. It is a service pushing data about skills/splits, vdns and vectors over a network socket.
It is virtually the same what you'll find in hsplit and so on but realtime.
Pushed data can be configured by your cms admin.
If you are looking for call data you may take a look at *call_rec* table in cms.
You can use clintSVR which is a high level tool based on CMS CLINT. By using clintSVR, you can use CGI, OCX and C++ interfaces to get the real time data from CMS.
As others have said you can get this from realtime reports. You'll need to scrape them.
RT socket is just a set of wrappers around clint for running reports. It takes the realtime report data and sends to to a socket.
You can roll your own real time reports with clint and feed that to whatever needs to ship the data. A sample realtime report can be run from the command line like:
/cms/toolsbin/clint -u your_user <<EXECUTE_DONE
do menu 0 "cu:rea:Meas"
do "Run"
do "Exit"
EXECUTE_DONE
Here is an example of running a report directly
Run report directly:
/cms/toolsbin/clint -u ini <<EXECUTE_DONE
clear
run gem "r_custom/cr_r_3"
do "Run"
do "Exit"
EXECUTE_DONE
I'm planning on writing a control panel for Minecraft in Rails but I don't have much experience with Java at all, Minecraft seems to have some standard remote connection and query tools, but most conventional panels don't seem to use them. For example with McMyAdmin, I have disabled remote connectiona and the query, but it still seems to be able to communicate with the server after restarting it after I've edited the server configs to disable the settings.
What I'm asking is if anyone knows how McMyAdmin communicates with the Minecraft server, it comes with a plugin, but I've deleted that as well and it still seems to be able to communicate with the server, I know McMyAdmin is written in .NET and I believe it uses Mono as it's server, as it's cross platform.
If anybody could shed some light on this I'd be ever so greatful, just trying to get my head around the communication.
McMyAdmin uses the plugin to open a socket that it can interact with(Not sure which features are provided using this plugin). The rest of the features are just from the Process instance that it creates. It also just edits the config files for a few things as well or runs commands using the input stream of the process.