Live streaming API
A live streaming API lets developers broadcast live video from an encoder or app to viewers on any device through code. The API creates a stream with a stream key and ingest URL, receives the feed over a protocol such as RTMP, RTMPS or SRT, transcodes it into a bitrate ladder as it arrives, packages it as HLS or LL-HLS and delivers it through a CDN. Most also record the broadcast for replay and report stream health while it runs.
How does a live streaming API work?
Your backend calls the API to create a live stream and gets back an ingest URL and a stream key, a secret that identifies the stream and authorizes the encoder. The encoder, such as OBS, a hardware unit or a phone app, pushes a continuous feed to that URL, usually H.264 video and AAC audio at a single high bitrate.
The service decodes the feed as it arrives and re-encodes it into a ladder of lower renditions on the fly. Each rendition is cut into segments and appended to live HLS or DASH playlists, which players reload to find new segments. CDN edges cache the segments, so thousands of viewers draw from the edge instead of the origin. Webhooks tell your app when the encoder connects, disconnects or errors. When the stream ends, many APIs turn the recorded segments into an on-demand asset, so a replay exists without a second upload.
How is live streaming API latency measured?
Latency is measured glass to glass: the time from light hitting the camera to the frame appearing on a viewer's screen. It adds up across the chain: encoder buffering, the ingest hop, transcoding, segment duration, CDN propagation and, often the largest part, how far behind the live edge the player chooses to sit.
Standard HLS uses segments several seconds long, and players buffer a few of them, so it runs well behind real time. Low-latency HLS (LL-HLS) splits segments into partial segments that players fetch as soon as they exist, which brings delay down to a few seconds while keeping CDN caching. WebRTC, the protocol behind video calls, goes lower still. It was built for two-way interaction, and it is harder to scale to large audiences than HTTP delivery that CDNs cache. Choose by use: a sports stream with live chat needs low latency, a lecture or a faith service values stability, and a two-way video call is a different category of product.
RTMP vs. SRT ingest for a live streaming API
Ingest is the leg from the encoder to the service. RTMP is the long-standing protocol here. Almost every encoder supports it, and it runs over TCP, which retransmits lost packets but stalls the stream while it waits. RTMPS is RTMP wrapped in TLS, so the stream and the stream key are encrypted in transit, and many destinations now require it.
SRT (Secure Reliable Transport) runs over UDP and adds its own selective retransmission inside a configurable latency window, plus optional AES encryption. On lossy or long-distance links, such as a venue's shared uplink, bonded cellular or a contribution feed crossing continents, SRT tends to hold quality where RTMP would stutter. The usual pattern is RTMPS for software encoders and broad compatibility, and SRT for professional and remote contribution. Whatever the protocol, a backup ingest, a second encoder or path that takes over if the first drops, protects events that cannot go dark.
What does a live streaming API include?
Beyond ingest and delivery, the features that matter sit around the broadcast. Recording turns each stream into an on-demand asset when it ends. DVR, or time-shift, lets viewers pause, rewind and return to the live edge, so late joiners catch what they missed. Live clipping cuts a highlight from an active stream. Live captions add speech-to-text while the event runs.
Distribution features decide reach. Simulcast, or multistreaming, sends one incoming feed to several destinations, such as social platforms, while the main stream plays in your own player. Access control uses signed playback URLs or DRM, and that policy should carry over to the replay. Operations features keep events safe: stream health data such as bitrate and dropped frames, webhooks on connect and disconnect, and capacity that scales with the audience without pre-provisioning. Some APIs can also play a recorded file out as a scheduled live event, called simulive.
How does the FastPix live streaming API work?
FastPix ingests RTMP, RTMPS and SRT from OBS, vMix, Wirecast, FFmpeg or hardware encoders, and iOS and Android live SDKs publish from a phone camera. Each stream has an authenticated stream key and supports backup ingest for redundancy. FastPix transcodes the feed into a ladder and delivers LL-HLS at 4 to 7 seconds glass to glass across a multi-CDN network of 300+ points of presence, with edge routing chosen per viewer. Concurrency scales with your plan, so there is nothing to provision before an event or tear down after. Sub-second interactive delivery over WebRTC is on the roadmap and not available today, and FastPix does not run two-way video calls.
While a stream runs, you get real-time bitrate and dropped-frame health, webhooks on connect, disconnect and error, DVR so viewers can rewind and return to the live edge, live clipping and real-time captions. Simulcast sends the same stream to YouTube, Twitch and Facebook Live at once. Playback can be signed with expiring tokens or protected with Widevine, FairPlay and PlayReady DRM. With auto_archive, every broadcast becomes an on-demand asset when it ends, with the same playback ID and policy. The free, open-source FastPix Player plays LL-HLS on web, Android (built on Media3), iOS and tvOS (built on AVPlayer) and Flutter, and video can be served from your own custom domain. Video Data covers live and replay views, and a VOD file can be played out as a scheduled live event. Oceaniek streams live sports on it.
Live encoding is $0.0324 a minute ingested, at any resolution up to 1080p. Viewers are billed per viewer-minute, and simulcast and live captions per minute. The free plan includes 30 live minutes, with no card.
