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WebRTC For Video Streaming

WebRTC For Video Streaming

Enable real-time communications directly from your browser.

WebRTC (Web Real-Time Communication) is a free, open-source project that provides web browsers and mobile applications with real-time communication (RTC) via simple APIs. It supports video, voice, and generic data to be sent between peers, allowing developers to build powerful voice and video communication solutions.

The technology is available on all modern browsers as well as on native clients for all major platforms. The WebRTC project is open-source and supported by Apple, Google, Microsoft, Mozilla, and others.  

Opus:WebRTC implementations

Opus is a lossy audio coding format developed by the Xiph.Org Foundation and standardized by the Internet Engineering Task Force. Opus is a totally open, royalty-free, highly versatile audio codec.  

The Opus format is based on a combination of the full-bandwidth CELT format and the speech-oriented SILK format. It can handle a wide range of audio applications, including Voice over IP, videoconferencing, in-game chat, and even remote live music performances. It can scale from low bitrate narrowband speech to very high-quality stereo music.

Opus support is mandatory for WebRTC implementations.

ZEGOCLOUD With our fully customizable and easy-to-integrate Live SDK, you can quickly build reliable, scalable, and interactive live streaming into your mobile, web, and desktop apps.

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