Hi I'm not very good with coding but I'm trying to learn as much as I can. I've been looking for tutorials on how to create an internet call app on android studio. So far I haven't found any. If anyone knows a process that could guide me I would very much appreciate it.
You can use android's own implementation
Session Initiation Protocol
from docs
Android provides an API that supports the Session Initiation Protocol
(SIP). This lets you add SIP-based internet telephony features to your
applications. Android includes a full SIP protocol stack and
integrated call management services that let applications easily set
up outgoing and incoming voice calls, without having to manage
sessions, transport-level communication, or audio record or playback
directly.
or other third-party libraries like following.
1.Pjsip
2.Mjsip
3.doubango
4.belle-sip
Hope it helps..
P.S taken from this answer
refer this also..
Happy Coding :)
Related
I am a developer and I have a Bluetooth Lamp that has RGB and Day/Warm Light that has a third party App to control it.
My goal is to do some automation with my Lamp.
Is there a way to read what this app is sending to my Lamp is order to simulate its functionality? The thing is this App is not possible to integrate with my Google assistant so I am trying to find a way to do it my self by making my own mobile application to control my Lamp.
Or maybe my question should be something like: is there a generic App that can control generic Bluetooth Lamps?
Any information is greatly appreciated.
You could either hack the hardware itself, or you could simply snoop on the communications to hack in to the lamp controls.
Otherwise, I would check out integration platforms like IFTTT, which allows some customization of control with proprietary systems.
I'm looking for tutorials or developers guideline docs to develop the chromecast built-in devices. Actually, I want to know software structure and how to get required libraries or sample codes on chromecast built-in in a linux device with screen.
I've already looked at the google cast SDK documentation from https://developers.google.com/cast/. However, it contains content about application and streaming services. I have not found what I want yet.
for example,
required DRM (widevine or playready)
media pipeline integration
device discovery (DIAL???)
application lifecycle management (if needed)
I expect how I can get documentation and resources on what I need to understand for "google cast built-in device" development. Thank you.
I am working on a webRTC application and would like to be able to support multiple calls and be able to call from the browser to legacy VoIP or Videoconferencing systems as well as browser to browser.
now that Asterisk has added websocket in their latest builds would you need SIP and a SIP proxy in order to communicate with VoIP systems or will Asterisk allow this?
now that H.264 has been open sourced by Cisco would you still need a transcoder in order to call a legacy VTC system?
Is Node.js the preferred technology for implementing webrtc client/server deployments? I've looked into Mobicents SIP Servlets a bit but that seems to be the only alternative technology available beside a node.js solution.
If needed I am planning on creating a SIP trunk between an Asterisk server and our Polycom VBP so the webrtc clients should be able to get presence information through that connection so if no media transcoding is required with the recent changes then media should be able to pass directly from polycom endpoint to browser with the asterisk handling the signalling.
Thank you anyone who is able to answer any of these questions, it is still early in the r&d portion of this project for me and i'd like to get as much information as possible.
also: i did see SIP over websockets to true SIP. I understand that "something" needs to stand in between the webRTC client and the VoIP phone or Legacy SIP endpoint. what I would like to know is if that can be just asterisk with the recent update. if asterisk is all that is required, is there a way to include a media transcoder like red5? I haven't seen anything in the webrtc API that would allow you to include a transcoder, asterisk has transcoding mods but none that will do vp8 to h.26x or Opus to anything as far as i know.
Answer on that question higly depend of destination "legacy" system. Cisco "legacy" systems use h323 and sip, which is not compatible with webRTC.
Sure there are alot of ways to setup asterisk, red5, opensips or other as translation level.
Webrtc goal is call from browser. It never supposed have any API for transcode. That have be done by server part(which require special knowledge and experience to be propertly setup)
There are alot of availible documentation in internet, no any way put answer in less then 30 pages of text.
I need to develop a project based on Bluetooth in mobile. Since I am new to j2me I studied some of the articles and run the project until the discovery of devices and services. I need to communicate between devices and transfer the desired files. I search code for client server communication through Bluetooth and got it but I didn't know how to run those code and implement further.
I have go through articles and I can run client server communication. Now I need to transfer the file and communicate to the user which was beyond the limit of my mobile through the another mobile which was within my limit.
JSR82.com has many articles and tutorials about how to use bluetooth from J2ME.
Better you refer the book, "BLUETOTH APPLICATION PROGRAMMING WITH JAVA API" by C.Balakumar. It is helpfull to you.
I want to know that can we make paypel related application in j2me and symbian OS?
Is it secure?
If we can make such application then can i get the api used for that.
Please help me for that.
Thank you
If you are asking whether PayPal offers any SDK for using the service on said platform through library APIs, the answer is no. See https://www.x.com/community/ppx/sdks?bn_r=m
If you are asking whether you could implement the NVP/SOAP APIs on this platform is: you could and it could be secure. It all depends on you ;)