A live stream moves from a camera or other source through encoding and an ingest connection, then often through transcoding, packaging, and delivery before a viewer’s player displays it. Knowing where each step happens makes terms such as bitrate, RTMP, HLS, CDN, and latency easier to understand—and helps you diagnose quality or delay problems.
How a live stream gets from source to viewer
Streaming is a pipeline, not a single protocol or piece of software. Services can combine or omit stages, so the exact implementation varies. The common path is:
- Capture: A camera, screen capture, or other source produces video and audio.
- Encode: Software or hardware compresses the source into a format suitable for sending.
- Ingest: The encoded stream travels to a platform or media service, often using a contribution protocol such as RTMP or SRT.
- Transcode (sometimes): The service may create multiple encoded versions for different quality levels.
- Package: Encoded tracks and metadata are arranged into a delivery format such as HLS or DASH.
- Distribute: An origin serves the packaged stream, often through a CDN.
- Play: A player on a phone, browser, television, or other device requests and displays the media. With adaptive delivery, it can switch between available quality versions.
Google Cloud’s Live Stream API documents an example of this pattern: it accepts RTMP or SRT input, transcodes into multiple renditions, and publishes HLS or DASH outputs. That is one service’s implementation, not a rule that every platform uses the same stages. Google Cloud Live Stream API overview
Core video and quality terms
Bitrate
Bitrate is the amount of encoded data sent per second, commonly expressed in kilobits per second (kbps) or megabits per second (Mbps). A higher bitrate can preserve more detail, but it requires more upload capacity and delivery bandwidth. It does not guarantee better-looking video on its own: resolution, frame rate, codec, encoding settings, motion, and scene complexity all affect the result.
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Resolution
Resolution is the image’s width and height, such as 1280×720. It describes pixel dimensions, not the amount of data used or the perceived quality. A high-resolution stream can still look poor if it is heavily compressed, while a lower-resolution stream can look clear at an appropriate bitrate.
Frame rate (FPS)
Frame rate is the number of video frames displayed per second. Higher frame rates can make fast movement look smoother, but usually require more data and processing at a given resolution and codec.
Codec
A codec is the method used to encode and decode audio or video. Examples in platform documentation include H.264/AVC, H.265/HEVC, VP9, and AAC. Devices and services differ in which codecs they support. A more efficient codec can reduce the data needed for a given result, but codec choice alone does not determine end-to-end quality.
Adaptive bitrate (ABR)
Adaptive bitrate delivery provides multiple encoded versions, often called renditions. A compatible player can use a higher-bitrate version when conditions allow or switch to a lower one when bandwidth or device conditions worsen, helping to reduce interruptions. The precise switching behavior depends on the platform and player. IETF RFC 9317
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A rendition is one encoded version of a stream, commonly identified by its resolution and bitrate. A service that creates several renditions can offer choices for ABR playback.
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Buffer and buffering
A buffer temporarily holds media data so playback can continue through small variations in delivery. Waiting for more data can make playback steadier but add delay. If the player runs out of buffered media, playback pauses to buffer again.
Protocols and formats: identify the direction
Protocol means rules for sending data; codec means how media is encoded and decoded. A creator’s contribution connection and a viewer’s delivery format are different parts of the pipeline. RTMP or SRT may carry a creator’s stream to an ingest service, while HLS or DASH may deliver packaged segments to viewers. Some platforms support formats in more than one role, so avoid assuming a name always describes the same direction.
RTMP and RTMPS
RTMP (Real-Time Messaging Protocol) is commonly used to send a live contribution stream to a platform. RTMPS is its secure counterpart. YouTube says RTMPS encrypts the contribution connection. In YouTube’s published comparison, RTMP and RTMPS are listed for normal, low, and ultra-low latency; the comparison’s support details are specific to YouTube and its documented configurations. YouTube: Live encoder settings, bitrates, and resolutions
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SRT (Secure Reliable Transport) is a contribution protocol. Google Cloud recommends SRT over RTMP when possible for its Live Stream API input endpoints, citing packet-drop recovery, forward error correction, support for multiple audio elementary streams, and higher bandwidth support. This is guidance for that API context, not a universal ranking: the encoder or transcoder and destination must support SRT for it to be usable. Google Cloud Live Stream API best practices
HLS
HLS (HTTP Live Streaming) is an HTTP-based, segmented delivery format developed by Apple and widely used for playback across devices. Low-latency variants can reduce delay, but the outcome depends on configuration and implementation.
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DASH
DASH (Dynamic Adaptive Streaming over HTTP), also called MPEG-DASH, is another HTTP-based segmented streaming format. Google Cloud documents it as an output option. In YouTube’s ingestion comparison, DASH ingestion supports H.264 and VP9 but is not suitable for ultra-low-latency use in the described configuration. YouTube’s ingestion comparison
WebRTC and RTP
WebRTC and RTP are associated with real-time communications and workflows that need very low delay, such as interactive exchanges. RFC 9317 notes that many ultra-low-latency IP applications use RTP or WebRTC with related protocols. Sub-second delivery is challenging when networks vary; reducing delay can mean accepting visible media artifacts. IETF RFC 9317
Protocol comparison at a glance
| Term | Common role | Useful distinction |
|---|---|---|
| RTMP / RTMPS | Contribution to an ingest service | RTMPS encrypts the contribution connection in YouTube’s documented workflow. |
| SRT | Contribution to an ingest service | Google Cloud favors it for its API context when the encoder and destination support it. |
| HLS / DASH | Often segmented viewer delivery; some platforms also accept them as ingest | HTTP-based formats that use manifests and media segments; latency depends on configuration. |
| WebRTC / RTP | Real-time communications | Used where very low delay matters; network variation may affect media quality. |
Packaging, origins, and delivery
Manifest or playlist, and segment
In segmented HTTP streaming, a manifest or playlist describes the available media segments and renditions. The player requests segments as it plays, and may choose among renditions. RFC 9317 describes streaming as continuous transmission of media segments from a server and their simultaneous consumption by a client. IETF RFC 9317, §1.1
Packaging and packager
Packaging arranges encoded tracks and metadata into a delivery format. A packager can change how media is organized or multiplexed without re-encoding it. That is different from transcoding, which decodes and re-encodes media.
Origin
An origin is the service or storage location that serves packaged content before downstream distribution. It is an upstream source for delivery infrastructure, not the camera or encoder.
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CDN
A content delivery network (CDN) is a distributed set of servers that helps deliver content closer to viewers and distribute load. CDN servers sit downstream of the origin or packager in AWS’s description; a CDN distributes media but does not encode the creator’s source video. AWS: What is a CDN?
Transcoding
Transcoding decodes and re-encodes media into another codec, bitrate, resolution, or set of renditions. It can support ABR by creating multiple versions. Packaging, by contrast, can change delivery arrangement without changing the encoded media.
Live-to-VOD
Live-to-VOD is a service feature that turns a live event into video on demand during or after the broadcast. Whether it is available and how it works depends on the service; Google Cloud lists live-to-VOD among its Live Stream API features. Google Cloud Live Stream API overview
Independent reader supportYour contribution helps us test, update, and keep practical guides available for everyone.Latency: what the delay includes
Latency is the time between capture and playback. Glass-to-glass latency is the end-to-end delay from the real-world event being captured to the corresponding media playing on a viewer’s device. It can include encoding and decoding, ingest, distribution, and buffering—not just time spent crossing the network. IETF RFC 9317, §4
RFC 9317 (2022) gives rough categories: ultra-low latency is under 1 second; low-latency live is under 10 seconds; non-low-latency live is 10 seconds to a few minutes; and on-demand is hours or more. These are broad terminology categories, not guarantees for a particular product or stream.
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Choosing sensible settings and protocols
There is no universal bitrate
Bitrate needs depend on resolution, frame rate, codec, encoding parameters, and how much motion or detail is in the scene. Google Cloud publishes the following H.264 source-stream recommendations for its Live Stream API input context. They are not general requirements for every encoder, platform, or viewer-delivery rendition. The documentation is undated and was accessed in 2026.
| Input resolution and frame rate | Google Cloud Live Stream API H.264 input recommendation |
|---|---|
| 720p, 25/30 fps | 8 Mbps |
| 720p, 50/60 fps | 15 Mbps |
| 1080p, 50/60 fps | 20 Mbps |
| 2160p, 50/60 fps | 50 Mbps |
Use the destination’s current recommendations and verify that your upload connection can sustain the chosen stream. A single bitrate number cannot account for every source, codec, or network.
Choose a protocol for the destination and use case
- Check the destination’s supported ingest options before choosing RTMP, RTMPS, or SRT.
- For YouTube contribution, RTMPS encrypts the connection. YouTube’s published comparison also documents HLS and DASH ingestion, with trade-offs in codec support, resolution suitability, and latency.
- Consider SRT where both encoder and destination support it and the recovery characteristics suit the connection; Google Cloud’s preference applies to its API, not every service.
- For interactive, very-low-delay communication, WebRTC or RTP may be relevant, but very low delay can be harder to maintain across variable networks.
Common misunderstandings
- “Higher resolution means higher quality.” Resolution is only one factor; bitrate, codec, frame rate, encoding, and content complexity matter too.
- “A CDN encodes my stream.” A CDN distributes packaged media. Encoding happens earlier in the pipeline.
- “RTMP, HLS, and H.264 are interchangeable.” RTMP and HLS describe transport or delivery formats in common workflows; H.264 is a codec.
- “Latency is just network ping.” Glass-to-glass delay includes media processing and buffering as well as network transit.
- “SRT is always better than RTMP.” Google Cloud recommends SRT in its own Live Stream API context. A different destination may not support it, and the right choice depends on the full workflow.
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