This article elaborates on the functional model of a web browser-based video conferencing system utilizing WebRTC technology.

WebRTC is committed to facilitating peer-to-peer (P2P) audio, video, and data transmission on the browser side. Conversely, a video conferencing system based on WebRTC technology necessitates server-side media processing, with the related specifications still under investigation. The current specifications of WebRTC technology primarily concentrate on the following aspects:
1. ICE, employing the IETF‘s ICE specifications for NAT traversal of media.
2. Utilizing RTP and its secure version SRTP for media transmission.
3. Media negotiation adopting the SDP protocol RFC 3264.
4. Recommendations for the signaling part.

Traditional video conferencing systems only support SIP and H323. A WebRTC technology-based video conferencing server needs to support its specifications, including ICE for media traversal, SRTP, and WebSocket. Such a video conferencing system is required to support the following points:
Network adaptability.
Support for packet loss retransmission and automatic bandwidth adjustment.
Recording: Support for conference recording, with file formats including AVI and MP4.
Support for SIP, H323, and WebRTC client access.