I want to implement my own encryption rule before the call data go into GSM network i.e. I want the call stream in the form of bits, I will implement my own encryption algo, and then send on to the network, my app on the other side(reciever's end) will recieve the data, decrypt it and make it into audio.
I want to know is it feasible, if it is how? I mean I want to use cell phone network as in like Airtel, Vodafone etc.
If it is not possible It will be of great help, if I can do it using internet (2G or 3G) ?
Any guidance in this, I want just direction.
Thanks in advance.
You can quickly create a chat application using Adobe Flex which will create an Adobe Air app that can run on Android (and also compile an iOS version if desired). The core strength of Adobe Flex is sending audio (and video) data with very little effort on the developers part.
You can configure your application to use SSL using the rtmps protocol if you want the data being transmitted to be encrypted.
This page shows you how you can create a simple video chat app for android using Flex http://coenraets.org/blog/2010/07/video-chat-for-android-in-30-lines-of-code/ - if you specifically don't want video you can send audio only data.
I can't imagine any reason why this wouldn't be possible as the networks are just passing data around, I don't think they care if it's encrypted or not encrypted - it's just a series of 1s and 0s.
As to how, that's a little beyond the remit of Stack Exchange - if you have a specific problems then post them with code.
There are other similar questions which you could look at:
Basic encryption on Android
https://stackoverflow.com/search?q=android+encryption
On Android, calls using the GCM (or other) network are handled by the baseband processor, which you don't have direct access to. You talk to it via the rild (Radio Interface Layer daemon) which uses proprietary library to talk to the actual hardware. So in practice you cannot mess with the mobile network.
A VOIP application would use the data connection and you can send/receive pretty much anything you want. If you use a standard technology such as SIP, there are ways to use TLS for the communication channel(s), so that traffic is encrypted. If you are creating your own, you might do something similar by using SSL sockets.
The 'how' part doesn't really fit the SO format, since it's very open ended and depends on how you decide to implement this.
Related
I wanna try to make an application like Discord (Groups/Add Friends/Voice Call/Video Call/Streaming) and until now I was focus on messages and on an application for Web(ReactJS or any) and PC/Mac/Linux(ElectronJS or any), and working on backend with NodeJS/ExpressJS/Socket.IO/GraphQL/MongoDB. Recently got a solution for Voice Call and Video Call which is PeerJS, but now I asked myself how would this work in Androd/iOS application? Am I choosing well my frameworks and language? Because of course one Android/iOS user have to be able to join to a Voice/Video with users on PC/Web Is there other frameworks? Not just for Voice/Video Call, all the application. Can anyone please share some documentation or tips?
Thanks
A personal tip would be to refrain from using PeerJS, as it doesn’t support VP8 Simulcast. When creating high-load group video calling software like discord, you need VP8 Simulcast because it splits a stream into different bitrates for different users, dependent on their bandwidth and that of the person streaming.
Consider reading about WebRTC in JS WebAPI! Unfortunately, the main issue nowadays is hosting a Signalling server (which essentially manages who is where in terms of rooms) and TURN Servers (which gets information such as the IP address of a user joining a room).
In terms of the iOS app, and Android App, consider implementing with a language of your choice, such as Xamarin.
Personally, I prefer React Native for mobile applications. Take a look at it online!
Good luck in your venture!
I have to build an application for android to stream video and audio to a desktop application through a server. Latency is important. I also have to make sure that android streaming can be controlled from pc (user should be able to switch the camera or turn off the microphone).
I thought to use the WebRTC protocol for communication but it seems I'm gonna have to write signalling server myself to support that requirement mentioned above.
Is there a better way to implement this whole thing? Also, I can't find any good docs or libraries for android streaming (no retrofit analogies obviously).
P.S. I'm thinking about using Javafx via Tornadofx for a desktop application.
You certainly don't need to create your own signaling server. I would suggest using something like Kurento Streaming Server or a derivation of Kurento like OpenVidu. It's open source and free and has lot's of great and active support via google groups. Depending on how much specific customization you may need one or the other might be better for you. OpenVidu allows for less customization since most of the stuff under the hood is already done for you, whereas Kurento allows you to modify and customize almost everything under the hood and on the front end using examples that can be changed at the code level. I have used it extensive on projects on the past and would think it meets most, if not all of your requirements. Scaling can be a bit challenging, but is still mush easier than just P2P webRTC since everything is relayed through a central server and most certainly doable depending on your requirements and implementation. Additionally you can record, process and transcode video server side.
I have thesis to do this year. I want to create an android VOIP application. But I need it to be secure. I intent to use the SIP. Maybe I wasn't searching good enough but I have to find some information fast. Do you know some way to encrypt the voice using SIP? A few times I found some information about ZRTP but the information wasn't really useful. Any help would be useful. I have three months to finish the project (but I can't work all the time because I go to school).
Signalling and Media are to be managed differently to build a secure voip application.
Signaling - SIP is a signaling protocol and carried over UDP [usually]. Adding a layer of protection to this can be done by using TLS as the transport to make the message exchange secure. The SIP packets are all encrypted and makes it secure over the transmission.
RTP - Usually the media is carried as RTP. But the secure flavor is the SRTP and in that ZRTP is a particular mechanism to handle encryption of the packets.
Coming to building the VoIP Application, i would recommend with going with a stack like PJSIP or numerous other options and this should help you have the prototype faster and understand the concepts better. Good luck.
I'm working on an Android game based on Playing Cards (Bridge, to be precise), which can be played by four players at time. And there'll be a server available via Web, to which devices will connect, and server will keep track of game progress.
My game is very basic when it comes to graphics that I can attain the UI without using any gaming engine.
While I'm supposed to build the game (the client) for Android, I wanted to develop server which can be RE-USED in future ports of the game, even if it is ported to other mobile platforms or even desktop.
So I thought of first possible candidate for server architecture was having RESTful Web Service so that I can leverage the server with any client as long as the client's programming end supports HTTP methods.
But later I realized that since there'll be persistent connection between devices and the server throughout the game session, would it be okay to have such a server, where connection will terminate after the request is responded (I'm not sure if it is true)?
Or shall I use DatagramSocket and DatagramPacket way of Java to build the server? (will that ensure re-usability of server?)
Any other suggestions or recommendations?
Note: I'm not new to Java or Network Programming in Java, but I'm new to both Android Development and creating RESTful services.
While writing for Android, don't plan for a persistent connection. Connections break very often (and often for good reasons, like switching from GSM to wifi). HTTP is a great, popular and proven choice (you get some lower levels of the stack out of your way and can focus on processing the payload).
BTW: saying "RESTful web service" int this context is meaningless - what you need is a HTTP server that serves data and accepts commands, not a mental framework for structuring your game logic as a set of stateful resources.
I think your HTTP-based plan is appropriate for this situation, I don't think the question of persistence of connection is relevant for a slow turned based game such as bridge.
Edit: as suggested by tdreger almost all Android docs recommend that you plan for routine connection failure and reestablishment through a different channel, as such the html connection seems the most resilient solution.
I think your idea of making it client-side independent is correct and important - in this light the HTTP idea is clearly much better in that it will be much easier to code client-side applications in other languages (which you will probably want - Javascript for a web-client and objective-C for an iOS app).
I also think the Android development will be easier as Android and appache have strong support for these HTTP-like connections.
I'm currently in an early stage of my internship at a company which offer VoIP solutions. I'm basically here to create a custom SIP-client App for iPhone. I told them however, if I were to set up the MVC pattern correctly and more efficient in terms of portability, there would be minimal code to write when porting to different platforms.
I've chose to go with MonoTouch C#.NET, for high portability and productivity (learning Objective-C is too steep for my timeframe + memory management too time consuming). To create even more portability I've been thinking of exposing a C# SIP library as webservice, so when porting to Android there's even less hooking up to different APIs. Also, MonoTouch for compile reasons does not allow usage of Dynamic Libraries.
My app would communicate to the SIP webservice and the webservice in turn to the SIP server.
SIP is very familier to HTTP, but could this solution work? As I'll be facing Realtime Transport Protocol aswell.
Kind regards
As far as I know, it won't work because, as you mentionned, you will face RTP. You'll probably get a lot of lag in your conversations. Also, you'll have to figure out how you are going to stream the data between the clients and the server.
However, to really know if this can be done would be to do a few prototypes to test these kind of issues.