RTMP vs WebRTC vs HLS : Live Video Streaming Protocols Compared

Video streaming has become an essential part of our daily lives, with the demand for video increasing exponentially in recent years. To facilitate this, various video streaming protocols have been developed, including RTMP, WebRTC, and HLS. Each protocol has its own advantages and disadvantages, making it crucial to choose the right one based on specific business requirements and customer expectations. RTMP (Real-Time Messaging Protocol) was developed by Adobe for high-performance live streaming of audio, video, and data between a dedicated RTMP streaming media server and Adobe Flash Player. It offers low latency and reliable communication but has lost its popularity due to the decline of Flash support across clients. HLS (HTTP Live Streaming Protocol) was launched by Apple in 2009 as an open specification for delivering media over HTTP, which is then playable with HTML5 players. HLS supports adaptive bitrate streaming (ABR), enhancing viewer experience and stream quality, and DRM encryption, making it ideal for last-mile content delivery. However, its main drawback is latency, which can peak at 20-30 seconds in some cases. WebRTC (Web Real-Time Communication) is a free and open-source technology that enables real-time video, audio, and data communication without any plugins. It supports ABR to a limited extent and works on a long stack of protocols to abstract the media engine, codec, and transport layer into APIs. WebRTC's main challenge lies in scaling due to the intense bandwidth configuration required for multiple peer connections. In conclusion, there is no single winner among video streaming protocols that can solve all challenges of enabling live video at present. A combination of RTMP, WebRTC, and HLS works best by bridging gaps in the entire lifecycle of video transmission - from ingest to delivery.