WebRTC (Web Real-Time Communication) technology is an advanced solution for voice, video and data communication between browsers and mobile applications without the need for additional plug-ins or applications.
Support for all major browsers and mobile OS, ideal for cross-platform applications.
Compatibility and usability for a wide audience of users.
Utilizes state-of-the-art codecs (Opus for audio and VP8/VP9 for video) for high quality communication at low network bandwidth.
Adaptive bitrate and error correction to minimize latency and data loss.
Direct connection between devices to reduce latency and load on servers.
Use STUN/TURN servers for initial connection setup and work behind NAT and firewalls.
Data protection using DTLS-SRTP, which provides a high level of security and privacy.
Powerful APIs for JavaScript and mobile platforms that simplify integration and customization.
Ability to quickly implement video calls, voice calls and data sharing.
Easily scale the application as the number of users grows.
Alarm and load balancing servers for stable operation under high loads.
A signaling mechanism for exchanging session data between users, using SDP and ICE protocols to find the optimal connection path.
Features capture, encode and decode audio and video streams using WebRTC API.
Adaptive bitrate control and data stream prioritization mechanisms for stable connectivity.
Measures to protect user data, including encryption and authentication of connections.