09 Jan
What is FreeSWITCH?
FreeSWITCH is free and open-source software that can be used to build and deploy private branch exchange (PBX) systems, IVR services, videoconferencing platforms, and other applications that require real-time communication. It is a powerful and flexible platform that can be used for a variety of purposes, including VoIP (Voice over IP) telephony, messaging, and presence.
FreeSWITCH is written in C and C++ and has a modular architecture. It is made up of a number of independent modules that can be used to add new features or to customize the software. FreeSWITCH can support a large number of concurrent users because it is very scalable.
What is WebRTC ?
WebRTC stands for Web Real-Time Communication. WebRTC is a free and open-source project that enables real-time communication (RTC) in web browsers and mobile applications using application programming interfaces (APIs). It allows audio and video communication to work within web pages without the need for plugins or native apps, by enabling direct peer-to-peer communication.
WebRTC is based on a number of open standards, including the Real-time Transport Protocol (RTP), the Session Initiation Protocol (SIP), and the Secure Real-time Transport Protocol (SRTP). All major web browsers, including Chrome, Firefox, Opera, and Edge, support it.
WebRTC and FreeSWITCH are a dynamic duo that revolutionize real-time communication. WebRTC is a free and open-source project that enables real-time communication in web browsers without plugins. FreeSWITCH, on the other hand, is a versatile and scalable telephony platform that provides robust call control and routing capabilities. When combined, WebRTC and FreeSWITCH offer a powerful solution for building advanced communication applications, such as voice and video conferencing, contact center solutions, and unified communication systems.
With WebRTC and FreeSWITCH, businesses can enhance customer engagement, enable seamless collaboration, and deliver exceptional communication experiences. WebRTC and FreeSWITCH help businesses connect people and improve productivity, whether for peer-to-peer or large-scale communication.
How WebRTC is Revolutionizing Browser-based VoIP Calling
Business users find browser-based VoIP calling popular because they can use it while they are connected to their systems and screen sharing or referring files. Thus, using a system is more convenient for communication than using a smartphone. Moreover, browser based calling makes the whole business communication experience flexible unlike using a PC dialer or similar app because it does not involve the installation of the same VoIP software at both ends. There are multiple other advantages related to using browser based communication tools, which has increased its user base. Build Powerful VoIP Applications with FreeSWITCH and WebRTC. Thanks to the versatile WebRTC development company, this has become possible and achievable without much fuss.
The top WebRTC SIP.js development company builds different WebRTC based apps that can be used for browser to browser calling. In the past, companies needed to install a native extension to provide browser-based calling functionality. However, thanks to WebRTC app development, it is possible to provision browser calls without native extensions.
Any VoIP development company with the right skills can help you with its WebRTC development services to build an app that supports browser to browser VoIP calls with multiple amazing features. Here are the steps to develop browser calling functionality with FreeSWITCH and WebRTC if you have WebRTC knowledge.
How to Develop Browser Calling Functionality with FreeSWITCH and WebRTC
Step #1: Install FreeSWITCH
- The first step is to install FreeSWITCH on Windows, Mac, or Linux systems.
Step #2: Edit WebSocket
The next step is to edit the WebSocket support by following the below mentioned steps:
- Open the configuration file, which you can find here: \conf\sip_profiles\internal.xml
- Make this piece of code uncommented: <param name=”ws-binding” value=”:5066″/>
Step #3: Restart FreeSWITCH
Once you open the WebSocket, you need to restart the FreeSWITCH service.
Step #4: Enabling browser calling
A browser based SIP client is needed to enable browser calling. The latest version of Chrome will always make this job easier, but you can also use any other browser SIP client to make this possible according to the best WebRTC development company.
Access the browser SIP client and provide the following information:
– Name: Your Name
– SIP URI: sip:1000@xxx
– SIP Password: 1234
– WS URI: ws://xxx
After you have provided all the required information, click enter to complete the process. The system will take you to the SIP dialer and you will be able to make calls with FreeSWITCH and WebRTC without investing in WebRTC app development.
Concluding notes
WebRTC makes it easy to build browser-based calling apps if you know VoIP and SIP. You can also take advantage of WebRTC SIP.js development company to build different real time communication apps by using their WebRTC development services.
We are renowned as the top WebRTC development company and we can help you build browser based calling facilities with FreeSWITCH and WebRTC. We can also help you with FreeSWITCH setup or any other VoIP related requirements you may have. To discuss your requirements, contact us NOW!
