Session Initiation Protocol (SIP) is a signaling protocol that plays a crucial role in establishing, managing, and terminating real-time communication sessions over IP networks.

A SIP transaction is a core element of the SIP protocol, encompassing a single request made by a client and the corresponding responses from a server.

In SIP communications, passwords play a key role in the authentication process, typically using a method known as Digest Authentication.

DTLS, or Datagram Transport Layer Security, is a protocol that provides privacy and data integrity for communications over datagram protocols, which are typically used for applications that require real-time communication and low latency, such as streaming media, voice over IP (VoIP), and online gaming.

WebRTC (Web Real-Time Communication) is an innovative technology that enables direct communication between browsers and devices.

Transcoding is a crucial function within the SIPERB platform, especially when dealing with legacy PBX systems that do not natively support WebRTC's modern codecs and protocols.

SIPERB provides robust WebRTC client formats to meet diverse communication needs across various devices. Our solutions include clients tailored for web, tablet, and mobile interfaces, each designed to optimize your communication experience regardless of your device type.

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.

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?

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.

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.

A WebSocket connection is a communication protocol that provides a persistent, two-way connection between a client and a server

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.

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).

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).

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.

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.