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World desk8 min

Live Streaming Glossary: Essential Terms Explained

A plain-language guide to how live video travels from capture to playback—and what bitrate, codecs, RTMP, HLS, SRT, CDNs and latency mean.
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A live stream moves through a chain: a camera or other source produces audio and video, an encoder prepares it for transmission, an ingest service receives it, and delivery systems carry it to a player. Terms such as bitrate, codec, RTMP, HLS, CDN and latency describe different parts of that chain—not interchangeable settings.

How a live stream gets from capture to playback

The exact components vary by service, but this is a useful map of the typical workflow. Google Cloud’s Live Stream API, for example, accepts RTMP or SRT input, can transcode it into multiple renditions, and can publish HLS or DASH outputs. AWS describes a CDN as a downstream network distributing content from an origin or packager. Those are product-specific examples, not a requirement that every stream use identical stages.

  1. Capture: A camera, screen capture, or another source produces video and audio.
  2. Encode: Software or hardware converts and compresses the source into a format suitable for sending.
  3. Ingest: The encoded stream is sent to a platform or media service using a contribution method such as RTMP or SRT.
  4. Transcode, if needed: A service may decode and re-encode the input into other formats, resolutions, bitrates, or renditions.
  5. Package: A packager arranges the encoded tracks and metadata for delivery, often as HLS or DASH segments and a manifest.
  6. Distribute: An origin serves the packaged stream; a CDN can distribute it through servers closer to viewers.
  7. Play: A player on a phone, browser, television, or other device requests and plays the stream. With adaptive delivery, it may switch among available renditions as conditions change.

Core video and audio terms

Encoder

An encoder is software or hardware that prepares a source signal for transmission, usually by encoding and compressing its audio and video. FFmpeg is one example of an encoder program cited in Google Cloud’s documentation. “Encoder” can refer to the software, the device, or the encoding component inside a larger system.

Codec

A codec is a method for encoding and decoding audio or video. H.264/AVC, H.265/HEVC and VP9 are video codecs; AAC is an audio codec. A source, service and playback device must support compatible codecs. A more efficient codec may reduce data needs at comparable quality, but codec choice alone does not determine end-to-end quality.

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Resolution

Resolution is the image’s pixel dimensions, such as 1280×720. It describes the size of the picture, not the amount of data sent or how good the picture will look. A high-resolution stream can still look poor if it is heavily compressed, and a lower-resolution stream can look clean under suitable conditions.

Frame rate (FPS)

Frame rate is the number of frames displayed per second. Higher frame rates can make motion look smoother, but typically require more data and processing at a given resolution and codec.

Bitrate

Bitrate is the amount of encoded data sent each second, commonly expressed in kilobits per second (Kbps) or megabits per second (Mbps). Higher bitrate can preserve more detail, but requires enough upload capacity and does not guarantee better perceived quality. Resolution, frame rate, codec, encoding settings, motion and scene complexity all affect what bitrate is appropriate.

There is no universal bitrate target. As configuration-specific examples, Google Cloud’s current Live Stream API input recommendations for H.264 are 8 Mbps for 720p at 25/30 fps, 15 Mbps for 720p at 50/60 fps, 20 Mbps for 1080p at 50/60 fps, and 50 Mbps for 2160p at 50/60 fps. These are recommendations for source input to that API, not general requirements for every platform or encoder.

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Rendition

A rendition is one encoded version of a stream, often defined by a resolution and bitrate. A service that creates multiple renditions can offer a player more than one quality option.

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Adaptive bitrate (ABR)

Adaptive bitrate delivery makes multiple encoded versions available so playback can use a suitable rendition as network and device conditions allow. More available bandwidth may support a higher-bitrate version; a weaker connection may require a lower one to reduce interruptions. The exact switching behavior depends on the player and platform.

Ingest and delivery: protocols, formats and packaging

When comparing streaming terms, first ask which direction they describe. A creator might send RTMP or SRT to an ingest service, which then prepares HLS or DASH for viewers. But a format’s role is not universal: YouTube’s published ingestion comparison also documents HLS and DASH ingest. Check the destination’s supported workflow rather than assuming a protocol is only used one way.

Protocol

A protocol is a set of rules for sending data. RTMP, RTMPS, SRT, HLS, DASH, RTP and WebRTC have different roles and characteristics. “Protocol” does not mean “codec”: a protocol governs communication, while a codec describes how audio or video is encoded and decoded.

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RTMP and RTMPS

RTMP (Real-Time Messaging Protocol) is commonly used to contribute a stream to a platform. RTMPS is its secure counterpart. YouTube documents RTMPS as encrypting the live contribution connection. The supported options and features depend on the receiving platform.

SRT

SRT (Secure Reliable Transport) is a contribution protocol. Google Cloud’s Live Stream API guidance prefers SRT over RTMP in that service context, citing packet-drop recovery, forward error correction, support for multiple audio elementary streams, and higher bandwidth support. Using it requires an encoder or transcoder that supports SRT; verify that both your sending software and destination accept it before switching.

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HLS

HLS (HTTP Live Streaming) is an HTTP-based, segment-oriented streaming format developed by Apple and widely used for device playback. Low-latency HLS variants can reduce delay in supported configurations, but actual latency depends on the platform and implementation.

DASH

DASH (Dynamic Adaptive Streaming over HTTP), often called MPEG-DASH, is another HTTP-based segmented format. Google Cloud documents it as an output. In YouTube’s published comparison, DASH ingestion supports H.264 and VP9 and is not suitable for ultra-low-latency use in the described configuration.

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Manifest, playlist and segment

A manifest or playlist describes the available media segments and renditions. The player reads it and requests segments as playback proceeds. HLS and DASH use this segmented delivery model; segments allow a player to receive and play parts of a stream over time rather than requiring one continuous file transfer.

Packaging and packager

Packaging arranges encoded tracks and their metadata into a delivery format. A packager can change how media is organized or multiplexed without re-encoding the video or audio. That differs from transcoding, which decodes and re-encodes media into a different format, codec, bitrate, resolution, or set of renditions.

Delivery and playback terms

Origin

An origin is the service or storage location from which packaged content is served before downstream distribution. In AWS’s description of CDN delivery, the origin or packager is upstream of the CDN.

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CDN (content delivery network)

A CDN is a distributed network of servers that delivers content closer to viewers and helps distribute delivery load. It is part of the distribution path, not the creator’s video encoder: it does not encode the source video. AWS describes CDN servers as downstream from an origin or packager.

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Player

A player is the software on a viewer’s device that requests, decodes and displays the stream. It may choose among available renditions, but selection behavior depends on that player and the delivery service.

Buffer and buffering

A buffer temporarily holds media data to smooth out variations in delivery. Waiting for more buffered data can make playback more continuous but add delay. If data arrives too slowly to keep the buffer supplied, playback may pause or stall.

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Latency: what the delay measures

Latency and glass-to-glass latency

Latency is the delay between capture and playback. Glass-to-glass latency measures the time from the real-world event being captured to its corresponding media playing on the viewer’s device. It includes more than network transit: encoding, ingest, distribution, buffering and decoding can all contribute.

The IETF’s RFC 9317 (2022) gives 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 content may have a delay of hours or more. These are broad terminology categories, not service guarantees.

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Choosing a latency trade-off

Near-real-time interaction calls for a different design than a buffered broadcast where continuity matters more than immediacy. Real-time communication systems such as WebRTC and RTP are often used when very low delay matters. The IETF cautions that network variation makes sub-second delivery challenging and may require accepting visible media artifacts. Segment-based delivery such as HLS or DASH can provide a different balance, with latency depending on configuration.

RTMPS, HLS and DASH: why the direction matters

YouTube’s published ingestion comparison describes RTMP and RTMPS as suitable for normal, low or ultra-low latency; RTMPS adds encryption for contribution. It describes HLS and DASH as supporting additional codec choices and being better suited to high-resolution use cases in that comparison, while typically incurring greater latency because they are segment-based. These are platform-specific distinctions, not a universal ranking.

Term Common role in the workflow What to check
RTMP / RTMPS Often creator-to-platform contribution Destination support, encryption needs and latency mode
SRT Contribution / ingest Support in both encoder and destination; recovery features depend on the implementation
HLS / DASH Often packaged viewer delivery; some platforms also accept them for ingest Platform-specific codec support, latency configuration and device compatibility
WebRTC / RTP Real-time communication where very low delay is important Network variation, implementation support and tolerance for possible media artifacts

Live-to-VOD

Live-to-VOD is a service feature that turns a live event into video on demand during or after the live event. It is separate from the live delivery path itself; availability and behavior depend on the service. Google Cloud lists live-to-VOD among its Live Stream API features.

Use the glossary to diagnose common questions

  • “Why does high resolution still look blurry?” Resolution is only one part of quality. Check bitrate, codec, motion, scene complexity and encoding settings; a high pixel count does not ensure sufficient detail.
  • “Why does a stream keep buffering?” The player may not be receiving data quickly enough to replenish its buffer. The cause could involve the connection, available rendition, delivery path or playback device; bitrate alone does not identify it.
  • “Why is the viewer behind the event?” Look at the complete glass-to-glass path, including encoder processing, ingest, packaging, segment duration, distribution and player buffering—not just network latency.
  • “Should I switch from RTMP to SRT?” Only if your encoder and destination both support SRT and its capabilities suit your route. Google Cloud’s preference applies to its Live Stream API guidance, not every streaming workflow.
  • “Is the CDN doing the encoding?” Usually that is a separate stage. The CDN distributes content downstream from an origin or packager; encoding and any transcoding occur elsewhere in the workflow.

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