Here you can find technical blog posts about how WebRTC, RTMP, HLS, CMAF, and other protocols are used and integrated. Build your own live streaming platform easily with these tutorials.
In this blog post, we’ll explore the best practices for deploying, configuring, and using Media Push. Media Push is a powerful tool that allows you to record conferences, capture any web page, merge video streams, create custom layouts, and much more. While Media Push is an incredible tool, it does Read more…
Ant Media provides a Unity WebRTC SDK built on top of the WebRTC for Unity packages developed by the Unity team. This SDK enables developers to publish and play ultra-low-latency WebRTC streams directly from Unity applications using Ant Media Server. The SDK is distributed as a GitHub project and includes Read more…
WebRTC has become a critical technology for industries ranging from telehealth and online education to live streaming, enterprise collaboration, and surveillance. By enabling real-time audio, video, and data communication directly in browsers, it eliminates the need for plugins or third-party installations.However, despite its maturity, Official WebRTC specifications support remains inconsistent Read more…
Amazon CloudFront scales HLS live streaming to millions of concurrent viewers by caching MPEG-TS segments at 600+ global edge locations, reducing origin server load by up to 95% compared to direct delivery. Ant Media Server functions as the CloudFront origin, ingesting RTMP or WebRTC streams and generating the HLS manifest Read more…
Video bitrate vs resolution is one of the most common — and misunderstood — concepts in video streaming, directly impacting video quality, bandwidth usage, and viewer experience. Whether you’re streaming a live sports event, producing video content for YouTube, or hosting a business webinar, your audience expects sharp visuals and Read more…
Discover what WebRTC servers are and learn key concepts like SFU, MCU, signaling, transcoding, and simulcast. Plus, get started with your own WebRTC server setup — for free.
WebRTC peer-to-peer (P2P) communication lets two clients exchange audio and video directly — after a short signaling handshake, media travels between peers without touching a media server. That single architectural decision cuts latency to sub-100ms on a good path and eliminates server-side bandwidth costs for the media stream entirely. In Read more…
TensorFlow object detection in Java uses pre-trained neural network models to identify and locate objects within image frames — without requiring Python runtimes or external ML services in your JVM application stack. The TensorFlow Java API loads SavedModel or Protocol Buffer graph definitions, runs inference against preprocessed image tensors, and Read more…
Before the WebRTC signaling server, let’s briefly explain what WebRTC is. WebRTC stands for Web Real-Time Communications. It is a powerful technology and an open-source streaming protocol. WebRTC is HTML5 compatible, and it allows you to add real-time communications to your apps to directly communicate between browsers and devices. You Read more…
HLS (HTTP Live Streaming) is a widely used protocol for live streaming due to its compatibility with Content Delivery Networks (CDNs), making it suitable for streaming to large audiences. Traditionally, HLS streams have a latency of about 8–10 seconds, but with Low Latency HLS (LL-HLS), this can be reduced to Read more…