A browser technology that enables real-time audio, video, and data communication directly between users without plugins.
WebRTC (Web Real-Time Communication) is a set of browser APIs that enable peer-to-peer audio, video, and data transfer directly between browsers without requiring plugins, downloads, or intermediate servers for the media itself. It is the technology behind video calls in your browser — when you join a Google Meet or a Zoom web session, WebRTC is doing the heavy lifting. Think of it as a direct phone line between two browsers, skipping the switchboard.
WebRTC establishes connections through a process called signaling, where two browsers exchange information about how to reach each other (IP addresses, supported codecs, etc.) through a signaling server. Once the connection details are exchanged, the actual media streams flow directly between the browsers — peer to peer.
In practice, direct peer-to-peer connections are not always possible because of firewalls and NAT (Network Address Translation). WebRTC handles this with STUN servers (which help discover public IP addresses) and TURN servers (which relay traffic when direct connections fail). The protocol also includes built-in encryption, echo cancellation, and bandwidth adaptation.
WebRTC is optimized for real-time communication with sub-second latency, which makes it ideal for video conferencing, live support, and interactive applications. However, it is not designed for on-demand video streaming at scale — it lacks the efficient caching and CDN delivery that protocols like HLS provide for pre-recorded content.
WebRTC opened up real-time communication to every web developer without the need for proprietary plugins like Flash. It powers video conferencing, telehealth, customer support, and live auction platforms. For businesses evaluating video technology, understanding WebRTC helps clarify the boundary between real-time communication (WebRTC) and video hosting and streaming (HLS/DASH).
host.video focuses on on-demand video hosting and streaming using HLS, which is optimized for pre-recorded content delivered at scale through CDNs. WebRTC solves a different problem — real-time, interactive communication — and the two technologies are complementary rather than competing.