RTMP Explained: Latency, Bitrate and Why 60fps Matters

Every live stream is a trade between three things: quality, latency and stability. Understanding how they interact is the difference between a crisp broadcast and a pixelated slideshow. Let's decode the stack.

RTMP: the delivery truck of live video

Real-Time Messaging Protocol is the 20-year-old workhorse that still moves most live video from encoders to platforms. Your phone encodes the video, RTMP carries it to the platform's ingest server, and the platform repackages it for viewers (usually as HLS). It's reliable, universally supported, and — with hardware-accelerated encoders like King Push's — fast enough for interactive streaming.

Bitrate: how much picture per second

• 1080p60: 4,500–8,000 Kbps. The sweet spot for action and IRL content.
• 1080p30 / 720p60: 3,000–6,000 Kbps. Safer on crowded 4G.
• 720p30: 2,500–4,000 Kbps. The "hostile network" setting.

Too high a bitrate on a weak uplink causes buffering; too low causes blocky artifacts. This is why adaptive bitrate matters — King Push re-negotiates per destination in real time, holding a stable ~7.8 Mbps master feed while platforms get what they can handle.

Latency: glass-to-glass

Latency is the delay between your camera sensor ("glass") and your viewer's screen ("glass"). Under one second feels face-to-face; 5–30 seconds (typical HLS) makes chat interaction feel like shouting down a well. Sub-second RTMP — under 380ms in v2.0 — is what makes live Q&As actually live.

Frame rate: why 60fps is worth the cost

30fps is fine for talking heads. The moment you move — walking streams, sports, gaming — 60fps halves motion blur and dramatically reduces viewer fatigue. Modern mobile chipsets encode 1080p60 with headroom to spare; there's rarely a reason to drop anymore.

The one-table cheat sheet

Set it once in King Push, let auto-adaptive transcoding handle the rest, and spend your attention on the audience — not the encoder.