Walkie-Talkie | Android app to enable infrastructure | Wifi library
kandi X-RAY | Walkie-Talkie Summary
kandi X-RAY | Walkie-Talkie Summary
An Android app to enable infrastructure-less communication using WIFI-Direct.
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Top functions reviewed by kandi - BETA
- Connects to the view
- Checks if the location is enabled
- Discover devices
- Gets the index of the device in a custom_peer list
- Creates a new device
- Generates a random position within the screen
- Checks if a position overlap another
- Initialize the activity
- Get the permissions for this application
- Initialize the WIFManager
- Initializes the audio stream
- Gets the socket
- Handles WIFP message
- Clear all device icons
- Starts the streaming session
- Start audio stream
- Override this method to toggle WiFi state
- Toggle the wifi state
- Called when the sound is destroyed
- Runs the audio record
- On createOptionsMenu
- Loads the first View
- Set the send button
- Check if a device has a specific device
Walkie-Talkie Key Features
Walkie-Talkie Examples and Code Snippets
Community Discussions
Trending Discussions on Walkie-Talkie
QUESTION
Building my first React.js app and I can't seem to get the app to redirect.
I am using the Twilio Voice TwiML (here) inside my React app. I have the frontend and server.
I can record what is said then transcribe it. Then redirect with an action:
to a URL.
Below is my call.js Twilio function (server). The /Omg
redirect isn't working.
ANSWER
Answered 2021-Jul-12 at 19:37React is a client-side application, not an HTTP server. It can't accept a POST request because no request will be made to it in the first place.
You need to write actual server side code to handle this the POST request and then redirect to a URL that serves up your React application to the browser.
QUESTION
I need to transform this code using a functional approach in the Kotlin language, while it is forbidden to use mutable collection and var variables, except for input and output
The attached code does the following
Schaeffer's bitwise stroke of the penultimate digit of all numbers (opera- walkie-talkies are performed from right to left)
...ANSWER
Answered 2020-Sep-26 at 13:10You can do it like this:
QUESTION
This is very early stages therefore no code, but some architecture questions maybe.
Im looking into trying to create a walkie-talkie functionality from ex a desktop-application or android application can send its audio to a server and that server then distributes the stream of audio to all clients.
My issue is that we are talking about both WIFI and LTE/4G network so has to work over the internet, and in theory should be possible to push audio from 1 to 1000 clients ( or select clients )
A small delay from speaker to its distributed isnt a big problem since its only one way communication ( not a like a phone with two way communication ).
Alot of questions arise here, mostly about size and speed :
primary thing im considering if i need to talk out to 1000 clients ( as in many many clients ) at the same time, i assume those 1000 clients all need a few sockets that is connected to the server, and therefore probably have to split it over more than one server to handle such kind of load ? ( i dont know ).
signalling part - would it be possible at all to have a signalling service handling that many clients ? ( assume the signalling needs also a constant connection to the server to be able to react when theres a audio-stream coming out, as it has to be fairly quick to react when someone speaks )
the protocols i have looked into on an overall perspective are SIP for signalling and RTP over TCP for transport, alternatively looked briefly at XMPP and fun-XMPP for inspiration, and in theory i can see it work on a small scale but my brain breaks when i try to imagine it on a large scale.
Core architecture was ex having a server handling SIP for signalling and keeping track of clients ( if SIP is fast enough/real-time enough which i am in doubt of ) - and then a transport server that the client would connect to with its audio-stream and then that server would channel that data out to all clients through the connection made when SIP recognizing someone is 'calling' them.
Side-question to this would be that i would like to have Go as back-end to manage this routing of data to all clients as a kind of media streaming server - but not sure if its fast enough ?
Maybe i am am totally wrong with the approach and protocols - maybe a way better approach then also interested in what you would suggest instead.
...ANSWER
Answered 2020-Jul-08 at 16:52You can explore this idea, as you are not bothered about the end to end delay.
Sender - RTSP/RTMP streaming from the Talker to the Server (1->1 streaming).
Receiver - DASH/HLS streaming from the Server to the multiple client/Receiver ( 1 ->Many streaming).
Sever Takes care of Buffering , transcoding or transrating if any thing is required.
for more details let me know
QUESTION
I am developing a push to talk app and I searched for possible protocols I can use. Those were H.323, MGCP and SIP. Also I came to know that the SIP is dominant over H.323 when it comes to the scalability but, I could not find anything that compares SIP and MGCP such that we can decide a clear winner. Since this is a mobile app, the scalability should be there as well as the security. Can you help me providing your valuable thoughts?
- Implementation of channel concepts.
- Notification generation to the user about channel activity.
- Implementing the push button activity(Walkie-Talkie style communication).
- Containerized server deployment.
are the main requirements of this App. Thank you!
...ANSWER
Answered 2020-Jun-05 at 22:33MGCP is all about a controlling media gateways, hence - Media Gateway Control Protocol.
If you're transcoding a large volume of calls, or moving media traffic from one medium to another, MGCP is the perfect protocol, it's very basic call control + SDP for defining the media attributes.
MGCP has 3 "verbs" in IETF speak (actions it can perform):
SIP is all about setting up and tearing down sessions, in your context, this means phone calls, it too uses SDP for defining the media attributes.
SIP has a plethora of "METHODS" - same as Verbs in MGCP - actions it can perform:
- INVITE
- CANCEL
- UPDATE
- REFER
- NOTIFY
- MESSAGE
- SUBSCRIBE
- REGISTER
- PUSH
- OPTIONS
and a pile more that can be added through extensions to the protocol, these methods open up features like call transfer, call hold, failure management, presence, SIP Registration, etc. You'll find these features in SIP but not MGCP.
On the specifics:
- Implementation of channel concepts. - SIP will be far easier to do this with, lots of open source options.
- Notification generation to the user about channel activity. - Again SIP has the NOTIFY method for this
- Implementing the push button activity(Walkie-Talkie style communication). - SIP UPDATE will allow you to start and stop media stream.
- Containerized server deployment. - All depends on what stacks you use.
I'd suggest you take a look at 3GPP's Mission Critical Push to Talk protocol as a reference of how do to this over SIP - That's essentially what MCPPT is.
QUESTION
i have the following sample app here: Github repo
It uses vuefire in ChatList.vue
...ANSWER
Answered 2020-May-25 at 17:30Since you want to check, in your Security Rules, if a given value (the user uid
in this case) is contained in a field of type Array in your document, you can use the in
operator of the List
type.
So, the following should do the trick:
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