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1Fix the driver behind crashes, sound loss and screen glitches2Clear out junk files and repair common Windows errors3Scan for outdated or missing drivers - takes under a minuteThroughput is how much data your connection can deliver per second; latency is how long it takes for a live event to reach a viewer’s screen. A stream needs enough stable upload throughput to carry its encoded bitrate, but faster internet alone does not guarantee a more immediate picture. The delay also comes from encoding, platform processing, and the player’s buffer.
Throughput and latency measure different things
Throughput: how much data can get through
Throughput is the rate at which a connection successfully delivers data, usually expressed in megabits per second (Mbps). For a creator, the key figure is sustained upload throughput: it must be sufficient for the stream’s outgoing bitrate, with room for fluctuations and other network use.
Latency: how long the picture takes to arrive
Live-stream latency is the elapsed time from the real-world event to the moment the media plays on a viewer’s device. RFC 9317 calls this glass-to-glass delay. It is not the same as a network ping: it can include capture and encoding, service ingestion and processing, delivery, player buffering, and decoding. The IETF’s RFC 9317, published in October 2022, uses rough categories of under one second for ultra-low latency, under 10 seconds for low-latency live, and 10 seconds to a few minutes for non-low-latency live. These are descriptive categories, not guarantees from every platform. RFC 9317
Why a stream can lag with good internet
A speed test measures capacity under its test conditions; it does not tell you the complete glass-to-glass delay. A stream can have abundant throughput and still arrive late because the service or player holds data before playback. Congestion or an unstable connection can also cause playback problems even when average throughput appears sufficient.
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The player’s read-ahead buffer is a central tradeoff. It stores incoming video so playback can continue through short variations in delivery speed. YouTube explains: “The lower the latency, the less read-ahead buffer the video player will have.” A smaller buffer can make a stream feel more immediate, but leaves less protection against network variation and can increase buffering. YouTube Help: latency settings
How much upload throughput does your stream need?
Your upload connection must keep up with the encoder’s outgoing bitrate. YouTube’s recommended video bitrates are encoder guidance, not universal internet-speed requirements. Allow additional headroom for bitrate variation and other household or production traffic; the safe margin depends on the stability of your connection and competing use.
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| Output format | AV1 or H.265 recommendation | H.264 recommendation |
|---|---|---|
| 1080p at 30 fps | 10 Mbps | 14 Mbps |
| 1080p at 60 fps | 12 Mbps | 17 Mbps |
| 4K at 60 fps | 35 Mbps | 50 Mbps |
These figures are YouTube Help’s published recommendations, accessed in 2026. Test your upload connection and compare its stable performance with the selected encoder bitrate; a single peak result is not a guarantee that the connection will sustain that rate. YouTube Help: encoder settings and bitrates
Does a higher bitrate increase latency?
Not by itself in every setup. Bitrate describes how much encoded data the stream sends per second; latency describes elapsed time through the full capture-to-playback chain. But if the upload connection cannot reliably carry the configured bitrate, frames may drop or the stream may disconnect, and congestion can make delivery less consistent. A higher bitrate can therefore expose a throughput problem without being the sole cause of delay.
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Codec and protocol also matter. More efficient compression can deliver comparable image quality at a lower bitrate, depending on the encoder and content. YouTube’s HLS ingestion documentation says HEVC generally provides 25% to 50% more data compression than H.264 at the same video quality. That is a codec comparison, not a promise that every stream will achieve a specific bitrate reduction. YouTube HLS ingestion documentation
Choosing a latency mode and ingest protocol
Match the mode to how viewers participate
- Normal latency: YouTube describes this as appropriate for non-interactive streams and says it offers the lowest viewer buffering.
- Low latency: Intended for streams with limited interaction. YouTube says most viewers experience less than 10 seconds in this mode.
- Ultra-low latency: Intended for real-time conversation. YouTube says most viewers experience less than five seconds, but this mode may increase buffering.
The YouTube figures describe typical viewer experience, not a guaranteed maximum. YouTube’s low- and ultra-low-latency modes do not support 4K. Webcam and mobile streams are set up for interactivity and do not expose the same latency selection. Platform behavior and settings can change. YouTube Help: latency settings
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Ingest protocol adds another tradeoff
YouTube supports RTMP and RTMPS for normal, low, or ultra-low latency; RTMPS adds encrypted transmission. HLS and DASH support newer codec options for high-resolution workflows, but typically add latency compared with RTMP because they deliver media in segments. For YouTube HLS, segments of one to four seconds are recommended. Shorter segments can reduce delay, but raise rebuffering risk and reduce encoding efficiency. YouTube protocol comparison · YouTube HLS ingestion documentation
Independent reader supportYour contribution helps us test, update, and keep practical guides available for everyone.Diagnose lag, buffering, or dropped frames
- Measure sustained upload capacity. YouTube recommends testing upload bitrate. Compare stable results with the encoder setting, not just the maximum speed shown by a test. OBS suggests a starting bitrate of 75% of total upload speed in its troubleshooting guidance; treat that as a rule of thumb, not a universal guarantee.
- If frames drop, reduce the stream bitrate. OBS says dropped frames indicate an unstable connection or an inability to keep up with the configured bitrate. Also check the selected streaming server route.
- Try Ethernet if Wi-Fi is unstable. OBS recommends a wired connection for streaming. Ethernet may improve the stability of the local link, but cannot increase your ISP plan’s capacity or guarantee lower glass-to-glass latency.
- Choose latency for the audience’s needs. Use a more buffered mode when interaction is not important and playback resilience matters; use a lower-latency mode when timely interaction matters, while watching stream health and viewer buffering.
- Check protocol and codec together. A more efficient codec may reduce the bitrate needed for a given quality, while segment-based ingestion can add delay. Neither choice removes the need for a stable upload connection.
OBS troubleshooting guidance: Dropped frames and network issues.
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When the stream is prerecorded and meant to run 24/7
Throughput and latency matter differently when you are broadcasting uploaded video continuously rather than interacting live. For a do-it-yourself setup, the streaming computer still needs to stay on and connected, and its upload connection must sustain the outgoing bitrate. You must also choose a latency mode and ingest protocol appropriate to the stream; the same buffering and quality tradeoffs apply.
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