End-to-end latency, sometimes called encoder-to-decoder latency, isolates the network and protocol portion of the chain. It is what you measure when you do not care about (or cannot control) the camera, the lens or the display. It is the natural unit for comparing two delivery protocols — HLS vs LL-HLS vs WebRTC — because each side of the comparison faces the same capture and render constraints.
The chain components are: encoder output buffer, contribution protocol (RTMP, SRT, RIST, WebRTC), origin/packager, CDN edge, last-mile network, decoder input buffer. Each adds latency: SRT typically 100–500 ms depending on retransmit window, packager adds a segment duration, CDN edge is bound by RTT and HTTP/2 head-of-line, and the player adds a buffer of 2–10 segments. The sum is what end-to-end measures.
A clean trick to measure it without specialised gear: timestamp every encoded frame with NTP time (most encoders can embed time-of-day metadata or SCTE-104), then in the player read the presentation timestamp and compare it to the wall clock. The difference is end-to-end latency. The result depends on clock sync quality, so most operations teams use a single NTP source for both encoder and player monitoring.
