Browser Phone
SIPERB began as a fork from the original Browser Phone project, a web-based softphone designed to facilitate real-time communication directly from a web browser.
SIPERB began as a fork from the original Browser Phone project, a web-based softphone designed to facilitate real-time communication directly from a web browser.
This article explains how to configure Siperb as a softphone for an existing Asterisk extension or endpoint.
SIPERB, an acronym for “Session Initiation Protocol Endpoint Relay Bridge,” is a transformative solution designed to merge traditional VoIP systems seamlessly with advanced WebRTC technology.
Web Real-Time Communication (WebRTC) is not inherently bound to the Session Initiation Protocol (SIP); it’s a versatile set of technologies designed for peer-to-peer media communications across web browsers.
In the evolving landscape of telecommunications, softphones have become a pivotal tool for businesses and individuals alike. A softphone is a software application that enables voice and video calls over the internet using a computer or mobile device, rather than through traditional hardware like a desk phone.
STUN (Session Traversal Utilities for NAT) is a protocol that assists in establishing peer-to-peer (P2P) connections over the Internet, particularly in scenarios involving Network Address Translators (NATs).
SDP, or Session Description Protocol, is a format used primarily in multimedia communications and applications to describe the details of media sessions. These sessions often involve real-time protocols like SIP (Session Initiation Protocol) and WebRTC (Web Real-Time Communication).
Web Real-Time Communication (WebRTC) is an essential technology that powers seamless voice, video, and data sharing over the internet in real-time. It’s commonly used in applications such as video calls, online meetings, and peer-to-peer (P2P) file sharing. However, a known concern surrounding WebRTC is its ability to potentially reveal users’ IP addresses, raising privacy issues for some.
A WebSocket connection is a communication protocol that provides a persistent, two-way connection between a client and a server
This guide provides a detailed setup for enabling WebRTC with FreeSWITCH, allowing for browser-based voice and video calls. It covers FreeSWITCH configuration for WebSocket and SRTP support, along with SIP.js setup for making and receiving WebRTC calls.