WebRTC Video Conferencing
WebRTC video conferencing lets users communicate amongst themselves via instant, streaming feeds that rival current video conferencing options in terms of quality and reliability.
WebRTC Video Conferencing Uses APIs Built Into Browser
WebRTC video conferencing takes advantage of three HTML5 APIs - getUserMedia, RTCPeerConnection, and RTCDataChannel - which come together to capture video and audio information, locate other users to communicate with, and transmit the streaming data to another peer (i.e. another browser). These APIs are built into Chrome, Firefox, and Opera. WebRTC uses the VP8 video codec and Opus audio codec to offer industry-leading streaming capabilities.
The VP8 video codec, the backbone of WebRTC video conferencing, has frame rate and resolution limits far higher than any monitor can reach. Users do not have to sacrifice high definition video to take advantage of instantly streaming media. Furthermore, VP8 is free and open source, which means WebRTC video conferencing does not come with any licensing fees.
All WebRTC video conferencing data transmitted through WebRTC is encrypted prior to delivery. This encryption is done using Datagram Transport Layer Security (DTLS), which is a standardized protocol. This encryption method comes with all browsers that support WebRTC, and it’s designed to ensure the data is passed along securely. It offers full encryption with asymmetric cryptography methods, message authentication, and data confidentiality to keep video data exclusively private.
WebRTC Video Conferencing and OnSIP