In our project we send HTTPS requests to Realtime DB (in order to add data to it), using the ESP8266 Wi-Fi component and AT commands.
For months we were able to communicate with Firebase, but last week the communication failed in the AT+CIPSTART phase.
We have been trying to find the solution for the problem for a week without success. We would greatly appreciate any help!
Order of commands:
AT+CWMODE=1
AT+CWJAP="NETWORK_NAME","PASSWORD"
AT+CIPSSLSIZE=4096
AT+CIPSTART="SSL","project-name-default-rtdb.firebaseio.com",443
-----THE ERROR IS HERE-----
The error:
ERROR
CLOSED
Note: We were able to write a TCP message to another site,but for Firebase we need to use SSL
Related
A ftp client program using libcurl is getting timeout. however curl work fine to same server port running vsftpd server.
Inside packet capture, difference i see that ftp client not getting new session ticket . seems like that client is waiting for that and getting timeout.
ftp client program packet capture
curl packet capture(which work fine)
please help in finding out cause of the issue. trying this past two days no progress.
NOTE: server is same in both cases, there is just server interfaces ip difference, which appear in packet captures.
Thanks in advance
Well I am studying about Web-RTC from its official documentation. I need to integrate live streaming in my website but after seeing this and studying about all its documentation. I only learn about streaming on locally like on same browser and same page. But this is not what i want. I want to start stream from my admin panel(This part has been completed) and broadcast that stream whoever has access to my website whatever the website is and whatever device. Whoever open my website if i m streaming then he should see that and if some of u have worked on live streaming then u should tell me. It will be a great help for me. All i did until now is to make a connection between two peers on same page. Now i want to make global peer to peer connection
I have done this implementation using simple-peer. Basically a wrapper for webRTC.
As soon as a new user connects a new webrtc connection should be made between the receiver and the caller. The receiver is firstly initialized and then sends a message to the sender to start the connection. This first connection is all done by your own server you should write.
Here is a working example. And here is the demo. Any connected devices will be automatically connected to the call. Multiple users supported. You'll find all the webrtc code in /public/js/main.js
You have to do signaling which mean that you have to exchange the PEER CONNECTIONS over the server and which required you to build a server page and client page so both of you can exchange the peer connection.
here is the complete procedure of exchanging the peer connection over the server.
find the heading
RTCPeerConnection plus servers
https://www.html5rocks.com/en/tutorials/webrtc/basics/
We started to get failed messages when refreshing extracts based on Bigquery
When looking at the error in the TABLEAU Server we see the next error:
Did you encounter a similar problem?
What did you do to sort this out?
Thanks!
Daniel
Resolution from the Tableau Knowledge Base:
As a troubleshooting step, check whether you can access "https://www.googleapis.com/auth/bigquery" on the affected computer using an internet browser.
Work with your internal IT to ensure that your firewall or antivirus allows communication to "https://www.googleapis.com/auth/bigquery".
The problem was the Big Query API IP being blocked, IT took care
I am coding a mac app, which will be a server that serve files to each user's mobile device.
The issues with this of course are getting the actual ip/port of the server host, as it will usually be inside of a home network. If the ip/port changes, its no big as i plan to send that info to a middle-man-server first, and have my mobile app get the info from there.
I have tried upnp with https://code.google.com/p/tcmportmapper/ but even though I know my router supports upnp, the library does not work as intended.
I even tried running a TURN server on my amazon ec2 instance, but i had a very hard time figuring how what message to communicate with it to get the info i need.
I've been since last night experimenting with google's libjingle, but am having a hard time even getting the provided ios example to run.
Any advice on getting this seemingly difficult task accomplished?
The port of your app will not change. The IP change could be handled by posting your servers IP to a web service every hour or whatever time period you want.
Server should run a URL http://your-web-service.com/serverip.php?ip=your-updated-ip and then have your serverip.php handle the rest (put it into a mySQL db or something)
When your client start it should ask your site for the IP and then connect to your server with that.
This is a pretty common way of handling this type of things.
since few days I'm trying to build a basic webRTC Videochat. I've got some Demos running localy, even via LAN. But now I want to build one by my one at the really basics without so much overload some Demos come with.
But I still don't get a complete peer connection.
Eg. this example seems to be broken, because I can't "createSignalingChannel();" w3.org/TR/webrtc/#simple-example
Some other examples (https://webrtc-experiment.appspot.com/) want me to link their scripts, but I wont do this, because I want to understand the magic of the peer connection and how to get a handshake between 2 browsers.
I also explored examples with the Google App Engine but thats not what I want.
I want to run it in really easy JS and HTML just on the minimum of what is neccessary.
Here is my code:
https://github.com/mexx91/basicVideoRTC EDIT: Should work now
So what will I have to add to get an handshake and peer connection, so that I can send eg. the mediaStream to eachother.
Thanks a lot!
createSignalingChannel() is only pseudo-code to illustrate the existence of a separate channel. You need for the initial connection handling a separate message channel.
You can achieve that with hosted services like Pusher, Brightcontext or PubNub, or you can host your own backend with open-source projects like socket.io or SignalR.
Then you just need to send the offers, answers and iceCandidates through your separate channel.
List of Realtime Services: http://www.leggetter.co.uk/real-time-web-technologies-guide
Imagine a video conferencing web-app, which users A and B originally access from some webserver. Suppose that web app supports presence, so the web server knows who's currently on-line. Imahine the UI allows A to try and place a video call to B. Via say XMLHttpRequest(), A's browser informs the server this is wanted, and B's javascript pops up something saying that A wants to call B. No WebRTC has happened at all yet. But at this stage, A can indirecttly communicated with B by sending messages using e.g. XMLHttpeRequest. In WebRTC parlance, this is the "signalling channel". So, A and B can both interact with their ICE agents to discover candidate addresses, and SDP descriptions, and send these to each ot6her, via the server, over this signallinh channel. E.g. the web app on A calls a WebRTC API to get its ICE candidates, and packages these up as it sees fit, to send to B. B's reader receives this message from the server (e.g over a WebSocket or long poll) and hyence it can unpack this, and format as needed to send to the ICE agent on B, using the RTCPeerConnection object. Similalrly, SDP offer/answer can be sent betweent he two apps, and passe through into the ICE agnet in the browsers, to get agreed media formats etc. At that stage, media connections can get set uo by the browser (meida streams are added to the RTCPeerConnection initially (which aren't communicating, but whihc have attributes that can be queried to describe the codec etc, and when the API is asked to create an SDP description, it does that using these attributes, but adjust the IP address and port based on how the ICE agent on each local browser has figured out what addresses can reach that local browser / port (NAT traversal).