WebRTC Selective Forwarding and Mixing
for all Web, Mobile and Native Platforms
Combining Peer-to-Peer Audio/Video with Server-Based Selective Forwarding & Mixing
LiveSwitch – using IceLink and WebSync as engines – extends peer-to-peer audio/video transmission with server based audio/video capabilities for applications that require selective forwarding (SFU), mixing (MCU), recording, and telephony integration. From three participants to three hundred, 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.
Selective Forwarding for Video Conferencing
Unlike mesh peer-to-peer networks, 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. A peer-to-peer architecture requires participants to upload their media several times - once for each remote peer.
This reduction in upstream bandwidth and client load means you can scale your application out much further on the client. Because the server forwards the media packets without decoding or re-encoding them, it also keeps the server load minimal, allowing you to make the most of each instance.
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 call participant, this is especially useful for legacy and resource-constrained devices. The server handles all the mixing automatically, and the output of that stream is delivered to each user 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 then undergo any post-processing required by your application to mix, modify, or archive.
The WebRTC SFU and MCU
Solution that Plays Nice With Everyone
- 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.
- 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.
- 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.
- 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.