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.
This guide details how to set up Asterisk for WebRTC, enabling browser-based voice and video calls. It covers essential Asterisk configurations for WebSocket, DTLS, and SIP, along with SIP.js setup to create a WebRTC client for making and receiving calls.
This guide explores how to integrate WebRTC with OpenSIPS, enabling browser-based voice and video calls. It covers essential OpenSIPS modules, TLS setup, and using SIP.js for WebRTC clients, complete with code examples for making and receiving calls. Perfect for building SIP-compatible, real-time communication in web applications.
WebRTC supports video, voice, and generic data to be sent between peers, building a powerful basis for building real-time communication applications, but how safe is it?
Session Initiation Protocol (SIP) is a signaling protocol used for initiating, maintaining, and terminating real-time sessions that involve voice, video, messaging, and other communications applications and services between two or more endpoints on IP networks.