Develop flexible live video applications with LiveSwitch Cloud's API and platform.
Take a hands-off approach managing live video platforms. Run it smoothly with reliable cloud hosting and leave the heavy DevOps lifting to our expert team of professionals.
Use LiveSwitch's ultra low-latency video and audio so that your users never miss a moment. When every second counts, trust us to deliver impressive video and audio for limitless numbers of end-users.
Learn the LiveSwitch API once and build for every platform on any device, browser, or hardware. Save time and get your app into the hands of mobile and browser users.
Provide DevOps and Support Teams with access to real-time session analytics and WebRTC statistics. Make evaluating video and audio quality easier than ever before - and deliver better experiences.
Learn the API once and build for every platform - future-proof applications have never been this simple. Plus, never get caught by an unexpected platform vendor update again. Get started fast with our suite of example libraries.
Build platforms that let your audience connect with a simple link. LiveSwitch's API is WebRTC-based, making joining video calls easy - no software downloads or plugins required.
Customize streaming experiences your way - any codecs, any bitrate, any video topology. Manage your users' streaming experiences down to a granular level.
Easily segment your platform's video and audio for all video sources and sinks. Create configurations that deliver the best user experience.
Managing real-time video and audio connection quality is easier than ever before with LiveSwitch Cloud's real-time telemetry dashboard. Get the WebRTC stats your team needs to diagnose and solve connection issues.
Available Stats: Bytes Sent/Received, Packets Sent/Received/Lost, NACK Count, PLI Count, and FIR Count
Spend less time worrying about how getusermedia, rtcpeerconnection, offer/answer, and icecandidates work and spend more time building great apps.
Tap into the embedded TURN Server and automatically spin up one to handle complex NAT traversal.
Seamlessly switch between P2P, SFU and MCU connections within a single conference for optimal performance.
Gain unprecedented API access to the media pipeline (eg. capture, encode, packetize, send, receive, depacketize, decode, display) to create powerful applications.
Capture audio and video streams from any device and integrate them into your application and workflows.
Use webhooks to monitor and intercept messages at any stage of your eventing system.
Reduce upload bandwidth requirements when broadcasting data across multi-party conferences.
Choose TURN-S, TCP/TLS, UDP/DTLS, and JWT-based authentication to achieve full security compliance.
Intelligently transcode media into compatible formats that work on all devices and platforms.
Send media in different qualities simultaneously based on your users' bandwidth.
Provide the best user experience possible for your audience by adjusting their bitrate.
Take control of the back-end of your video conferences with our powerful REST APIs.