If you are developing a VoIP system for yourself or your business, you could have seen SIP.js and FreeSWITCH. They allow you to have reliable voice calls over the internet, without having to sign up for a phone line.
How do they actually join forces? What capabilities do they have to support your system when your experience evolves?
To make it easier, let’s explain what it consists of.
Introduction to What Is SIP.js
SIP.js is a library for JavaScript that makes it possible to build voice and video calling apps right in your browser through WebRTC (Web Real-Time Communication). More simply, you can use your browser to call anyone, so you don’t have to worry about downloading additional software or apps.
You are able to use SIP.js WebRTC for:
Talk and receive videos calls directly from your browser.
Try out FreeSWITCH and Asterisk as your VoIP systems.
Ensure you add calling options into your web or mobile apps.
What is the FreeSWITCH project?
The FreeSWITCH platform is designed to support open-source communication. It’s the main part that runs your VoIP system like a real engine. It handles:
Routing calls
Running audio conferences
Recording calls
Voicemail and IVR are just two of the services they provide.
With SIP.js, FreeSWITCH takes charge of handling all the call logic.
Why Should You Use SIP.js Compatible With FreeSWITCH?
By using SIP.js and FreeSWITCH together, you get the best results.
You get an easy-to-use browser interface by using SIP.js.
The complicated work is done by FreeSWITCH on the server.
They support you in building web-based VoIP applications that can support many users — all without the expense of buying pricey equipment.
In what ways is Slides different from using SIP.js together with Asterisk?
SIP.js Asterisk is used by many as a popular VoIP platform as well. While Asterisk is good for most PBX systems, FreeSWITCH is more effective for those handling many calls or demanding conference calls.
When designing a system that can expand, especially if it will use video or WebRTC features, SIP.js FreeSWITCH offers more choices.
The advantages of constructing a VoIP system using SIP.js and FreeSWITCH.
Let me tell you why this pair is so great together.
100% of calls happen over the web using SIP.js WebRTC.
Can be used on mobile, desktop computers or in browsers
Can be used by hundreds or thousands of people at once
It is both open source and able to be customized by anyone.
You don’t have to spend a fortune on phone equipment.
Final Thoughts
Making a VoIP system that can grow isn’t always costly or complicated. You are able to make a smart, modern calling platform using SIP.js and FreeSWITCH and it will run right on the user’s browser.
This configuration allows you to expand as little or as much you like, since you aren’t limited by standard phone systems. If you have worked with Asterisk using SIP.js, you’ll find that FreeSWITCH is a robust choice for bigger or more complicated projects.
Do you want to test it yourself? Make sure to start with a configuration powered by SIP.js, WebRTC and FreeSWITCH. After that, keep working on your features and adding more users.
