WebRTC is an innovative technology that allows customers to quickly deploy audio and video communication functions without installing any plug-ins and communicate through the network anywhere. WebRTC is based on SIP protocol, but it does not fully support SIP protocol. Jet Zhiyun’s media server can be used as a WebRTC convergence gateway to mediate the media plane differences of WebRTC, so that existing PSTN and SIP communication terminals can communicate perfectly with Web and APP, and integrate traditional communication network with advanced WebRTC technology.

The main functions of CloudWebRTC media server include:

Standard SIP Protocol WebScoket JS SIP Protocol Stack
Flexible Dial-up Controller
P2P Pass Through Call
VP8 < H264 Transcoding Call
Multi-party audio conference
Video MCU Function
Video Call Recording
Video File Playback
SIP-based text messaging, offline storage
Perfect compatibility with Chrome, Firefox, Opera, control support IE, Safari, etc.
Real-time call list, call list storage
Real-time call management and extension online information management
Services support Windows, Linux and other mainstream platforms