SIP.js is a JavaScript library that allows web apps to make voice/video calls and send messages using SIP over WebRTC
It’s lightweight, open-source, and supports modern real-time communication directly from browsers without plugins.
Yes, we can build feature-rich, browser-based softphones using SIP.js that replace desktop-based tools.
Absolutely. SIP.js works with WebRTC to handle both voice and video communication.
Yes. SIP.js supports secure signaling (WSS), encrypted media (DTLS/SRTP), and authentication mechanisms.
Yes. We ensure compatibility with FreeSWITCH, Asterisk, Kamailio, and other SIP servers.
Yes, we create modern, responsive interfaces with full call control and status indicators.
Yes. SIP.js apps can run on mobile browsers or be packaged as PWAs for mobile devices.
WebRTC handles media (audio/video), while SIP.js handles signaling to set up and control communication.
Yes, we provide support, upgrades, and enhancements for all SIP.js-based solutions.