SIP Over WebRTC

Web Real-Time Communication (WebRTC) is an open-source project and technology that enables real-time communication capabilities directly in web browsers without requiring additional plugins or applications.

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?