RTMP

What is RTMP?

RTMP (Real-Time Messaging Protocol) is the standard protocol used for sending live video and audio streams to streaming platforms. Originally developed by Macromedia (later Adobe) for Flash Player streaming in the early 2000s, RTMP became the de-facto standard for live broadcasting because it's reliable, supports continuous streams, and is widely supported across encoding software and platforms. Major streaming platforms (YouTube Live, Twitch, Facebook Live, LinkedIn Live, custom streamers) all accept RTMP ingest as the primary upload method.

RTMP works by establishing a persistent TCP connection from the encoder (OBS, FFmpeg, hardware encoders) to the platform's ingest server. The encoder packages video/audio in chunks and pushes them continuously over the connection; the platform ingests, transcodes (multiple resolutions for adaptive playback), and distributes to viewers in near-real-time. Latency is typically 5-30 seconds end-to-end depending on platform configuration.

RTMP vs WebRTC vs HLS

Three live-streaming protocols, used differently. RTMP — encoder-to-platform ingest (push direction). WebRTC — browser-based peer-to-peer streams, ultra-low latency (sub-second), used for video calls and interactive streaming. HLS (HTTP Live Streaming) — platform-to-viewer playback (chunks served via HTTP), used for adaptive-bitrate viewer-side streaming. The typical broadcast pipeline: encoder → RTMP → platform → HLS → viewer browser. RTMP is the upload protocol; HLS is the playback protocol.

RTMP has been declared 'deprecated' by some observers because Adobe stopped formal development and Flash Player itself was retired. In practice, RTMP remains the most reliable and most-supported live ingest protocol in 2026. Platforms accept it; encoders push it; the protocol works. Replacements (SRT, WHIP/WebRTC ingest) exist but adoption is uneven.

Frequently asked questions