Video Call
Streaming
Video Room

Introduction to Streaming Architecture Overview with JanusScaler

JanusScaler is an advanced WebRTC load balancer designed to efficiently manage and distribute WebRTC streaming traffic. This architecture guide provides insights into setting up a robust streaming session with JanusScaler, including its components and their interactions.


Streaming Architecture Overview

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  • Janus Gateway : Acts as the core WebRTC media server that facilitates the communication between clients and streams. It manages the sessions and handles media transmission, ensuring low latency and high performance.
  • Load Balancer : JanusScaler’s load balancer intelligently distributes incoming streaming requests across multiple Janus instances, optimizing resource usage and ensuring high availability. This component also monitors the health of the servers to reroute traffic in case of failures.
  • Clients : The end-user devices (browsers, mobile apps) that initiate the streaming sessions. Clients interact with Janus via WebRTC protocols, sending and receiving media streams in real-time.
  • Media Transcoding Services : For scenarios requiring different codecs or formats, JanusScaler can integrate with transcoding services to convert streams on the fly, ensuring compatibility across various devices and networks.
  • Monitoring and Analytics : JanusScaler includes built-in monitoring tools that track streaming performance metrics such as latency, bandwidth usage, and session stability, helping administrators make informed decisions for optimizing the system.

Key Features


  • Videocall API : Create high-quality, low-latency video call experiences with ease.
  • Videoroom API : Host virtual meetings, conferences, or webinars with support for multiple participants.
  • Streaming API : Deliver live or on-demand video content to your audience, with adaptive bitrate streaming.
  • Audiobridge API : Implement audio conferencing solutions that can scale to thousands of participants.
  • SIP Integration : Seamlessly connect with SIP networks for VoIP services, enabling interoperability with traditional telephony.