I want to make a webhook that can be triggered by either get/post and triggers connected IoT devices to a WebSocket.
So, I thought Azure might help to automize this process, instead of writing everything from scratch and run it on a webserver.
I am very new in the Azure world, I found it very complicated to make it working on Azure.
Can you point me to any simple to make it work?
The first thing to do is to decide where you want to connect your devices to. Generally, you'll either use Azure IoT Hub or IoT Central (which uses an IoT Hub anyway). Your question doesn't include any details about your devices, or whether you're developing them yourself, but I'll assume you can connect to either.
If you go for Azure IoT Central, the easiest way to get a webhook going is to create a Logic App and link nodes together to trigger a Command to your device. You can use an HTTP request as the trigger, and use the built-in IoT Central node to trigger your command, example:
If you end up connecting your devices to IoT Hub, you can use Direct Methods, Cloud to Device messages or even Device Twins to communicate with your devices. You can leverage these by creating an Azure Function with an HTTP trigger, and you program the logic to call IoT Hub, which will communicate with your devices. There are a lot of samples on the web that show how to create Functions, or control your device remotely.
There are a lot of ways to go about your scenario, if you need some help from the Microsoft community, you can get a lot of help on Microsoft Q&A.
Update based on the comments:
In this case, the devices are connected to IoT Hub. You can use the Service SDK to run a job that fires a Direct Method to all currently connected devices. This doc describes the process.
Is there a simple guide from where I can start creating a stun / turn and signaling server ?
I spend over a week searching for those things and couldn't find any guide where I can say:
okay, I am on the right track now - this is clear.
So far, everything is so abstract without any examples.
This is what I'm trying to achieve: a simple video stream on my local network where I'll have a server with installed usb camera on it, and an application on my iis which will connect to the usb camera and stream it to the clients, and every time when a client opens the application, will see the video stream from the server camera.
Note: since I want to use it on my local network do I really need a stun/turn server, or is there a guide that shows how to avoid it ?
Media streamed over dedicated servers HTTP/HTTPS rarely needs a NAT traversal solution. Instead, just have your web server with camera attached, on the public Internet or behind your NAT with port-forwarding enabled.
There are LOTS of streaming media solutions available as open source, free downloads, or commercially sold. A good list is here:
https://en.wikipedia.org/wiki/List_of_streaming_media_systems
I am a new user of NS-3. Can you help me in writing a code of handover in between nodes in Wi-Fi environment?
I'm gonna base my response on the little information you have provided.
General idea of Wi-Fi (as the technology which we use on daily basis to connect to internet and stuff like that :D) is based on IEEE 802.11 standard which does not support handover between nodes. Therefor you either are asking the wrong question or you want to implement the handover support to a IEEE 802.11 standard.
In case you want to do the latter I strongly suggest you to do the following steps (NS-3 tutorial in case needed):
1- Install NS-3 simulator on a Linux operating system, preferably Ubuntu 14.04.
2- Browse wifi/examples folder and try to fully understand basic wifi examples.
3- Implement your proposed algorithm in proposed scenario (Which obviously requires you to understand how user equipment connects to a Wi-Fi access point automatically and how APs can interact with each other and etc.).
Is WebRTC going to be free for web developers to set up video calls on web pages?
why does Twilio having pricing 25c per mins for video calls,
is it going to be too expensive for the small guy to mange video calls on web hosting servers?
any advice from anyone deep into WebRTC already?
Some of the comments above are not well informed.
Someone wrote, since the bandwidth needed in case of media relay is higher as well. This is not entirely true, transmission happens between Peers(Browsers), servers are used just for signalling(relaying IP addresses of connecting peers and some more info), you can ROUTE your transmission from central server(for fail overs), but can surely do without it for free.
WebRTC is Free and you can setup the whole thing on your own without having to shell out even a penny. It is a bit hard and mitigating fail-overs is really difficult, but you can certainly do it for free.
Tokbox or Twilio charge money because these tools abstract some very rigid complexities of setting up, running and managing fail-overs in a WebRTC application.
In TokBox's Case:
You don't need to setup STUN, TURN servers, you don't have to worry
about integration with android or IOS clients, they provide a plugin
for IE too, so out of box you get everything and you just have to
concentrate on your application logic rather than WebRTC nuances.
This is a big plus.
Both RELAY and ROUTED schemes came in the box hence you can write
fail-over scenarios if RELAY communication fails. Although there are
some good JavaScript based frameworks that do this in a much cleaner
manner.
It adds slew of other goodies which help in building android and IOS
clients without any pain.
STUN or TURN Servers are used only for Signalling Purpose, and this signalling happens before any actual transmission. This signal is very small and carries the IP address of both the browsers(machines running browsers). For Transmission the communication is done between Browsers(Peer to Peer) themselves, so no server is involved.
Your relay is not happening from a central server so you don't have
to pay for the outgoing bandwidth cost.
To Setup Turn Server,
Use this server, build it and put it into a Rackspace/Amazon Web
Services instance and you are Good with your TURN
Server. That is It, setup your application and have fun with WebRTC
for FREE.
rfc5766-turn-server
If you wish to Use some more free framework to ease yourself more, check out: EasyRTC and PeerJS
Enjoy Developing with WebRTC....
Twilio developer evangelist here.
Your link at the end of your question points to our WebRTC page, which currently talks about the product Twilio Client. Twilio Client briefly is a way that, using WebRTC within browsers and mobile applications you can make phone calls to real phone numbers. This product does not allow you to conduct video calls.
Twilio Client has a cost because of the ability to call out from a browser to a telephone number. The cost is not in the WebRTC portion, but delivering those minutes to the other leg of the call.
Notably, it's not 25 cents ($0.25) a minute, instead it is just a quarter of a cent ($0.0025) a minute.
With regards to video calls with WebRTC, you can now access the public beta of Twilio Video, a platform to make setting up WebRTC calls much easier.
Twilio Video costs for the signalling infrastructure and you can see the prices here. If a WebRTC connection requires a TURN server to relay the media, that also costs per gigabyte of transfer. Usage of the STUN server is free, the costs for the TURN relay are available here.
Please get in touch with me at philnash#twilio.com if you have any other questions about WebRTC.
WebRTC is a technology placed in a browser. It requires backend infrastructure to support it - specifically, STUN and TURN servers as well as signaling servers.
This boils down to the fact that you pay for WebRTC - same as you pay for hosting your website on a server. The price is higher, since the bandwidth needed in case of media relay is higher as well.
To understand more about WebRTC and how it works (as well as why there's a price tag associated with services such as Twilio for it), you can check this free report: https://bloggeek.me/webrtc-business-people/
WebRTC is already free for developers to use. When we added WebRTC to our product, we used this example code, which made it very simple to build a WebRTC client:
https://shanetully.com/2014/09/a-dead-simple-webrtc-example/
Google and Mozilla provide free STUN servers, and it is easy to set up a TURN server. Most clients will be able to connect via STUN, so you won't end up using too much bandwidth on your TURN server.
To set up your own TURN server, coturn seems to be the easiest to set up:
https://github.com/coturn/coturn
Make sure you read the "WEBRTC USAGE" section in the README.turnserver file.
"STUN or TURN Servers are used only for Signalling Purpose, and this signalling happens before any actual transmission. This signal is very small and carries the IP address of both the browsers(machines running browsers). For Transmission the communication is done between Browsers(Peer to Peer) themselves, so no server is involved."
if that is the case, then you should be able to do this on a standard web server using Java/php. PHP will get the IP address of the guys connected to it. Then its just a matter of storing them in MySQL, then making a javascript that would run when the user go to that page in the site.
I've been looking for a solution around using a VPS because running a dedicated server for signaling is like golfing with a Ferrari instead of a golf cart. I still don't think node is efficient. Its single threaded. so node's fararri can only go 5mph.
Since they went to the web site, the php service already can get their ip address what else does it need? All of the above solutions so far require you to pay for a dedicated app to run on a server connected to the web separately for what 5k of data? What a waste of electrons.
But I'm going to start a new thread that is going to be based on getting webrtc without the buy a "VPS" because we want a VPS-less solution.
Need to access Dialup terminal transnational history
I'm a software engineer and thus far my world has been on the internet. So any payment processing I've ever doen has been through Gateways (Auth.net, SecureNet, ...etc) However I have a new project that requires me to deal with physical terminals. Some terminals that are linked to WiFi or Ethernet are exactly the same they just go through the gateway and and the backend API continue to work. Dialup terminals are apparently different and skip a gateway all together.
Does anyone know how I can get a dialup terminal's transactional history via some backend API?
So I called up the gateway (securenet) and they are telling me that I wouldnt get terminal transaction history via the API. This was confusing because the API documentation states if there was a pin pad on site which would imply a terminal.
I also called up a terminal programming firm. They assured me that even if they wrote custom software to the terminal it still wouldnt have access to the transaction unless it was the processing software itself.
so the answer is I'm SOL