July 2022 Summaries

Internet users today demand seamless online experiences, including video communication. To meet these expectations, video conferences must offer sub-second latency and high-quality audio/video transmission. WebRTC is often chosen to build such video experiences. While some WebRTC calls are possible without an external server, a signaling server is required to establish the connection. Most calls would fail on a direct connection, and issues arise as more peers join in. To optimize call performance, different servers can be used within the WebRTC ecosystem. This article discusses various elements on the server side that must be considered when building a WebRTC solution, including signaling servers, STUN/TURN servers, and media servers like MCU and SFU.

The world of live streaming is undergoing significant changes in user behavior that are driving the need to reevaluate the live streaming tech stack. Traditional methods have involved combining RTMP encoded media streams at the streamer's end and HLS streams at the viewer's end, with an industry of media servers in the middle for transcription. However, new user behaviors such as democratization, interactivity, and creator collaboration are driving a shift towards WebRTC becoming an alternative to RTMP in the live streaming tech stack. This modern protocol built over UDP can adjust encoding bitrates based on network feedback, is more widely available, supports interactivity, and enables creator collaboration out of the box. The evolution in live video will leverage conferencing tech to introduce interactivity and other benefits while conferencing will leverage streaming tech to scale video calls to many viewers in near real-time.

100ms has introduced Starter Kits to simplify the process of building live audio or video applications. These kits are proof-of-concept versions of use cases constructed on real-life interactions built with the 100ms SDK. They enable users to experience a working, proof-of-concept version of whatever they are imagining and serve as building blocks for implementing their own ideas. The starter kits include Instant Demos, Open Source Github Repositories, and Deploy to Vercel options. Currently, five starter kits with various use cases have been rolled out, including Video Conferencing, Virtual Events, Slack Huddle Clone, Discord Starter Kit, and Live Audio Apps.

HTTP Live Streaming (HLS) is a media streaming protocol designed to deliver audio and video content over the internet. It allows for adaptive bitrate streaming, enabling viewers to experience high-quality video playback even with varying network conditions. HLS has become the default way to play video on demand and is widely supported across platforms, devices, and browsers. The protocol consists of three major components: the media server, content delivery network (CDN), and client-side video player. In 2019, Apple introduced Low Latency HLS (LL-HLS) to significantly reduce latency while maintaining compatibility with Apple devices. Advantages of using HLS include transcoding for adaptive bitrate streaming, efficient delivery and scaling, and support for advanced features such as ad insertion, digital rights management (DRM), closed captions, and more.

This article discusses the network reliability of the 100ms React SDK, which is crucial for creating engaging live experiences in applications. The tests simulate common scenarios such as low bandwidth, network blips, and network switching to evaluate how the SDK handles these situations. The results show that the 100ms SDK performs well under regular usage conditions and gracefully handles degradation issues like bandwidth drops.

HTTP Live Streaming (HLS) is a widely used media streaming protocol designed to deliver audio-visual content over the internet. It facilitates content transportation from media servers to viewer screens across various devices, including mobile, desktop, tablets, and smart TVs. HLS allows distributing content and ensuring excellent viewing experiences across devices, playback platforms, and network conditions. Adaptive Bitrate Streaming (ABR) is a key feature of HLS that enables video players to choose which segments to download and play based on the available bandwidth and network strength. Low Latency HLS (LL-HLS), an extension of HLS, significantly reduces latency by dividing segments into parts and delivering them as soon as they're available.