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How a live stream gets from source to viewer
A useful way to understand streaming terms is to follow the media in the direction it travels. A typical workflow has these stages, though services can combine or omit steps.
- Capture: A camera, screen capture, or other source produces video and audio.
- Encode: Software or hardware called an encoder prepares and compresses the source into data that can be sent. FFmpeg is one example of an encoder program in Google Cloud’s documentation (Google Cloud Live Stream API overview).
- Ingest: The encoded stream is sent to a platform or media service through an input endpoint. RTMP, RTMPS, and SRT are common terms at this stage.
- Transcode, if needed: The service may decode and re-encode the incoming media into different sizes, bitrates, or codecs. These alternate versions are called renditions.
- Package: A packager organizes encoded audio and video into a delivery format, often with a playlist or manifest and media segments.
- Distribute: An origin serves the packaged stream. A content delivery network (CDN) can distribute it through servers closer to viewers.
- Play: A player on a browser, phone, TV, or other device requests and plays the stream. With adaptive delivery, it can switch between renditions as conditions change.
For example, Google Cloud’s Live Stream API describes ingesting an input, transcoding it into multiple renditions, and publishing HLS or DASH outputs. Its documented configurations support SRT or RTMP inputs and HLS or DASH outputs. This is one service’s implementation, not a rule that every streaming workflow uses the same stages (Google Cloud Live Stream API overview).
Terms that describe video and audio
Bitrate
Bitrate is the amount of encoded data sent per second, usually expressed in kilobits per second (Kbps) or megabits per second (Mbps). More bitrate can preserve more detail, but it also demands more upload capacity from the source and more delivery capacity along the route. A high bitrate by itself does not guarantee a better-looking stream: resolution, frame rate, codec, encoder settings, and the complexity and motion in the scene all matter.
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Resolution
Resolution is the width and height of the video image, such as 1280×720. It is not the same as bitrate or perceived quality. A higher-resolution stream can show more image detail, but only if the source, encoding, available bitrate, network, and viewer’s screen support it.
Frame rate (FPS)
Frame rate is the number of video frames displayed each second. A higher frame rate can make fast movement look smoother, which may matter for gameplay, but it typically requires more data and processing at a given resolution and codec.
Codec
A codec is a method for encoding and decoding audio or video. H.264/AVC, H.265/HEVC, VP9, and AAC are examples found in the cited platform documentation. A codec’s efficiency does not determine end-to-end quality on its own, and support varies among platforms, encoders, and playback devices (Google Cloud Live Stream API overview; YouTube live encoder settings).
Rendition
A rendition is one encoded version of a stream, often defined by a particular resolution and bitrate. A service can produce several renditions so viewers have alternatives when their connection or device cannot handle the highest-quality version.
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Adaptive bitrate (ABR)
Adaptive bitrate delivery offers multiple encoded versions and lets playback use a suitable rendition as network and device conditions allow. More bandwidth may support a higher-bitrate version; when conditions worsen, switching to a lower one can help avoid interruptions. The selection behavior depends on the platform and player (IETF RFC 9317; Google Cloud Live Stream API overview).
Terms that describe how a stream is sent and delivered
Protocol
A protocol is a set of rules for how data is sent. RTMP, RTMPS, SRT, HLS, DASH, RTP, and WebRTC have different roles and characteristics. A protocol is not a codec: a protocol governs transport or delivery, while a codec describes how media is encoded and decoded.
Ingest or contribution
Ingest—also called contribution—is the stage when a creator sends an encoded stream to a platform or media service. RTMP, RTMPS, and SRT commonly appear in this context. Be clear about direction: a format used to send a feed into a service is not automatically the format viewers use to play it. Some systems support formats in more than one role, and YouTube documents HLS and DASH ingestion as well (YouTube live encoder settings).
RTMP and RTMPS
RTMP means Real-Time Messaging Protocol; RTMPS is its secure counterpart. Both are commonly used to send a live contribution to a platform. YouTube describes RTMPS as encrypting the live contribution connection. Its supported codecs, latency modes, and other features are specific to YouTube’s documented configurations (YouTube live encoder settings).
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SRT
SRT (Secure Reliable Transport) is another contribution protocol. Google Cloud favors SRT over RTMP for its Live Stream API input endpoints, citing packet-drop recovery, forward error correction, support for multiple audio elementary streams, and higher bandwidth support. That recommendation applies to Google Cloud’s service context; switching requires an encoder or transcoder that supports SRT, as well as support at the destination (Google Cloud Live Stream API best practices).
HLS and DASH
HLS (HTTP Live Streaming) and DASH (Dynamic Adaptive Streaming over HTTP) are HTTP-based formats commonly used to deliver segmented media to viewers. A playlist or manifest describes available segments and renditions; the player requests segments as playback proceeds. HLS was developed by Apple and is widely used for device delivery. Google Cloud documents both HLS and DASH output (Google Cloud Live Stream API overview; IETF RFC 9317).
In YouTube’s published ingestion comparison, RTMP and RTMPS are listed for normal, low, or ultra-low latency, with RTMPS adding encryption. HLS and DASH support additional codec choices and are described as better suited to high-resolution use cases in that comparison, but typically have greater latency because they are segment-based. These are platform-specific tradeoffs, not a universal ranking (YouTube live encoder settings).
WebRTC and RTP
WebRTC and RTP-related technologies are often used when interactive, very low-delay communication matters. IETF RFC 9317 notes that many IP applications requiring ultra-low latency use RTP or WebRTC with related protocols. Keeping delivery below a second can be difficult when network conditions vary and may require accepting visible media artifacts (IETF RFC 9317).
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Packaging, packager, and transcoding
A packager formats encoded tracks and metadata for delivery. It can change how tracks are arranged or presented without re-encoding them. Transcoding is different: it decodes and re-encodes media into another format, codec, bitrate, resolution, or set of renditions. A service may do both, but the terms refer to distinct operations (AWS CloudFront documentation; Google Cloud Live Stream API overview).
Manifest or playlist, segment, origin, and CDN
A manifest or playlist describes the media segments and renditions available to a player. The player requests segments as it plays. The origin is the service or storage location that serves the packaged stream before distribution downstream. A CDN is a distributed network of servers that helps deliver content toward viewers and distribute load. The CDN is part of delivery; it does not encode the creator’s source video (AWS CloudFront documentation; IETF RFC 9317).
Buffer and buffering
A buffer temporarily stores media data to smooth out variation in delivery. Waiting to build a buffer can make playback more continuous but adds delay. If the player runs out of buffered data, playback can pause while it waits for more.
Live-to-VOD
Live-to-VOD is a service feature that turns a live event into video on demand during or after the event. Whether it is available, and how recordings are handled, depends on the service (Google Cloud Live Stream API overview).
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Latency: what the delay actually measures
Latency is the delay between capture and playback. A stream might prioritize near-real-time interaction, low-latency viewing, or a more buffered broadcast experience. “Glass-to-glass latency” describes the time from the real-world event being captured to the corresponding media playing on a viewer’s device. It includes more than network transit: encoding and decoding, ingest, distribution, and buffering can all contribute (IETF RFC 9317).
RFC 9317 (2022) gives these rough categories: ultra-low latency is less than 1 second; low-latency live is less than 10 seconds; non-low-latency live ranges from 10 seconds to a few minutes; and on-demand is hours or more. They are broad descriptions, not guarantees for a particular service or stream (IETF RFC 9317).
Bitrate and resolution: why there is no universal setting
Bitrate needs vary with resolution, frame rate, codec, encoding parameters, and how much motion or detail is in the scene. A fast-moving game, for example, can be harder to encode cleanly than a mostly static scene at otherwise similar settings. The destination platform’s accepted settings and the uploader’s available capacity matter too.
Google Cloud publishes the following H.264 source-stream bitrate recommendations for its Live Stream API input context. These figures are configuration-specific recommendations, not universal requirements for live streaming:
| Input resolution and frame rate | Google Cloud recommendation for H.264 |
|---|---|
| 720p at 25 or 30 FPS | 8 Mbps |
| 720p at 50 or 60 FPS | 15 Mbps |
| 1080p at 50 or 60 FPS | 20 Mbps |
| 2160p at 50 or 60 FPS | 50 Mbps |
Do not treat those values as a target for every platform or encoder. Check the destination’s current specifications and leave enough upload headroom for a stable contribution stream. Google Cloud’s guidance also favors SRT over RTMP for its own input endpoints when possible, but the choice depends on support at both ends (Google Cloud Live Stream API best practices; YouTube live encoder settings).
Choosing terms and settings for a real stream
- If sending a feed into a platform: Look at its ingest options and distinguish contribution protocols such as RTMP/RTMPS or SRT from viewer-delivery formats such as HLS or DASH.
- If viewers need interactive timing: Ask what latency the platform and player actually support. A “low-latency” label does not by itself establish glass-to-glass delay.
- If playback must adapt to changing connections: Check whether the workflow provides multiple renditions and adaptive bitrate playback.
- If selecting a bitrate: Consider resolution, frame rate, codec, content motion, encoder settings, upload capacity, and the destination’s limits together.
- If choosing a protocol: Confirm both the encoder and destination support it. Protocol advantages are context-dependent; for example, Google Cloud’s SRT recommendation is for its Live Stream API input endpoints.
- If diagnosing poor playback: Separate encoding problems from transport, packaging, CDN delivery, buffering, and device/player compatibility. A CDN cannot repair a source that was poorly encoded.
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