Combining Peer-to-Peer Audio/Video with Server-Based Selective Forwarding & Mixing

LiveSwitch is a software-based server that extends traditional WebRTC peer-to-peer audio/video to challenging applications that require selective forwarding (SFU), mixing (MCU), recording, and telephony integration. From three participants to three thousand, LiveSwitch makes WebRTC- and SIP-compatible audio/video conferencing scalable, efficient and truly cross-platform.

LiveSwitch provides unparalleled flexibility to combine P2P-, SFU-, and MCU-based media flows in a single session and switch dynamically while the session is live. 





Take It For a Test Drive

View a basic online demonstration of the LiveSwitch example source code that is provided with the JavaScript SDK.

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Selective Forwarding for Video Conferencing

Selective forwarding use a one-up, many-down architecture that lets participants send their media once to the server where it is distributed out to connected downstream clients. 

This reduction in upload bandwidth and client CPU usage allows each client to scale further.  Additionally, with our exclusive SFU transcoding feature, the server  directly forwards the media packets without decoding/encoding, yet is smart enough to transcode when codec incompatibility is detected. 

Mixing for Multiparty Audio & Video

LiveSwitch also functions as a multipoint control unit, or MCU, and supports mixing audio and video together into a single stream based on standard or user-defined video templates.

With just one upload stream and one download stream for each participant, this is especially useful for legacy and resource-constrained devices. The server handles all the mixing automatically, and each output stream is delivered to to the client in the format their device requires.

SIP Connection for VOIP/PSTN Integration

LiveSwitch provides a SIP connector that can be used to directly access SIP trunks or integrate with VOIP/PSTN virtual PBXs such as FreeSwitch and Asterisk. Learn more

Recording Audio/Video Streams

LiveSwitch can record individual SFU or mixed MCU streams out to Matroska containers in real-time. These files can undergo any post-processing required by your application to mix, modify, or archive. 

The WebRTC Peer-to-Peer, SFU and MCU
Solution that Plays Nice With Everyone

Microsoft Windows .NETLinuxmacOSApple iOSAndroidUWP HololensJavaXamarinFirefoxExplorerChromeSafari


  • Implements the WebRTC standard, including all required specifications and some optional ones.
  • Completely interoperable with other WebRTC implementations, like Chrome, Firefox, and Edge (ORTC).
  • Support for Internet Explorer through ActiveX.


  • Supports peer-to-peer communications via WebRTC.
  • Selective forwarding (SFU) and mixing (MCU) capabilities.
  • End-to-end encryption that can be customized to meet application requirements.
  • Applications can change media flows at will.

Cost Effective

  • Reduces developer costs by increasing productivity due to it's simple, intuitive client APIs and extensive documentation.
  • Reduces short-term costs by providing complete examples and demo applications to get your developers working quickly.
  • Can be run on your own internal or cloud infrastructure to reduce costs.

Dynamic Architecture

  • LiveSwitch can easily switch between SFU and MCU architectures on a per-session basis instantly, based on the needs of your application.


  • Uses WebSync as a signaling engine also capable of text chat and browser synchronization.
  • Supports integration with SIP through an included connector service.
  • Supports integration with third-party signaling through custom connector services.
  • Customizable layouts for mixed video streams.

Signaling Agnostic

  • LiveSwitch supports any signaling system. It is designed to operate with WebSync out-of-the-box, but can be extended to support third-party signaling protocols such as SIP and XMPP.

Download the LiveSwitch Datasheet


Download our LiveSwitch Whitepaper


LiveSwitch Getting Started Documentation

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