March 2023 Summaries

Cloud infrastructure has made streaming video content easier than ever, but developers looking to add live video creation into their products have limited choices. Web Broadcast is a new tool that allows developers to build live video layouts right in the browser using HTML and CSS they already know. The server does all of the rendering and compositing, producing high-definition live streams regardless of the user's browser size or bandwidth. Changes to the layout are automatically kept in sync with the live stream in real time. Web Broadcast is still being actively developed and users can sign up for early access or email for more information.

Mux, a company that focuses on developer experience, faced issues with their documentation feedback form being exploited by users attempting to break the intended usage. They initially implemented sanity checks and protections against SQL and code injection but were still receiving hundreds of invalid submissions per hour. To address this issue, they used Vercel's Edge Config feature to block repeated bad feedback and IP addresses without requiring a full deploy each time an update was needed. This solution allowed them to maintain their developer-focused product while ensuring the quality of user feedback.

Mux Real-Time Video has introduced a new Crop Broadcast layout, which aims to maximize the use of pixel real estate in livestreams with fewer participants. This feature is particularly useful for sessions with only two or three people on-stream and works by cropping the input video on the fly. The Crop mode supports 16:9, 1:1, and 9:16 aspect ratios and is available to all accounts at no additional cost.

Real-time communication has come a long way, drawing inspiration from traditional methods such as walkie-talkies. WebRTC is a collection of protocols and APIs that enable real-time media (audio and video) with low latency. It differs from chunked streaming protocols like HLS and DASH, which prioritize high quality and cost-effective distribution but have higher end-to-end latency. WebRTC is designed to optimize for low latency above all else, making it ideal for applications such as video chat or teleconferencing. The main challenge with WebRTC interoperability lies in signaling, which involves exchanging key information between participants to establish connectivity and start sending media back and forth. While there are standards for content and order of SDPs, the transport method is left open-ended. This can lead to proprietary methods of exchanging SDPs, making certain types of interoperability challenging. To address this issue, WHIP (WebRTC-HTTP Ingest Protocol) aims to provide a convenient ingest workflow similar to RTMP while maintaining the benefits of WebRTC. By narrowing the scope to only ingest or video contribution workflows, WHIP can make useful simplifying assumptions and use an HTTP POST request to send an SDP offer and receive an SDP answer in the response. This allows for a well-defined endpoint for WHIP client configuration and enables media to be ingested using WebRTC. WHIP is still a work in progress, but it has the potential to benefit various applications and services that require real-time communication with low latency.