Latency refers to the lapse in time between a camera capturing an action in
real life and viewers of a livestream seeing it happen on their screen.
Ultra-low latency is when that time is short enough, typically 0.5 - 3
seconds, so viewers of a live-streaming application can interact with what’s
happening in the stream in a way that feels natural.
Understanding Protocols
Livepeer Studio delivers video using several protocols. The most common (though not the only) delivery protocols supported by Livepeer Studio are HLS and WebRTC.Delivering video with HLS
HLS (HTTP Live Streaming), initially developed by Apple for iOS devices, serves video with a series of segmented.ts files. It is broadly supported across
many device types and is extremely well-optimized for serving multiple
renditions; these characteristics position HLS as the default choice for many
gateways.
However, HLS has very high overhead and latency. Specifically, its chunked
segment delivery means that viewers must wait for the current segment to finish
downloading before they can start viewing it. Similarly, HLS players often
buffer a few segments in advance to ensure smooth playback and to handle network
fluctuations. This buffer introduces additional latency, as the player waits to
accumulate enough data before starting playback. Buffer behavior is heavily
dependent on client implementation.
HLS delivers video with standard latency of 10-20 seconds. Streaming HLS
with Livepeer Studio’s recommended low-latency settings lead to ~10s
latency.
Delivering video with WebRTC
WebRTC (Web Real-Time Communication) is a streaming protocol known for its low-latency capabilities, making it suitable for real-time communication applications like video conferencing, online gaming, and live streaming. Unlike HLS, which is designed for adaptive streaming and on-demand content, WebRTC focuses on minimizing latency. WebRTC offers low latency by prioritizing real-time communication, direct peer-to-peer connections, UDP transport, and adaptive streaming while minimizing intermediaries. It is a preferred choice for applications where minimizing delay is critical, such as video conferencing and live interactive streaming.WebRTC delivers video with ultra-low latency of 0.5 - 3 seconds.
Optimizing Playback for Low Latency
Lowest possible latency (WebRTC playback)
WebRTC is available as a delivery protocol for all streams regardless of input protocol; to achieve the lowest possible latency of 0.5 - 3 seconds, use of WebRTC delivery is required. We implement the WHIP/WHEP spec with a few minor nuances around SDP negotiation. To play back a WebRTC rendition, you will need to use a WebRTC-compatible player, such as the Livepeer Player.Optimizing for Lower Latency (HLS and WebRTC ingest and playback)
Depending on your goals, you may or may not want to optimize for the lowest
latency; for example, if your user base has a high percentage of low-bandwidth
mobile users, it may be preferable to optimize for bandwidth efficiency over
latency.
Streaming with OBS
Achieving the right balance between low latency and stream quality is essential for providing the best possible user experience. Two settings that significantly impact your stream quality, latency, and user experience are:- Rate Control: This setting dictates the bitrate or “quality” of the stream. A high amount of bandwidth usually means better quality, but keep in mind that your output can never improve the quality of your stream beyond your input.
- Keyframe Interval: Video streams consist of full frames and data relative to the full frames. This setting determines how often a full frame appears, which heavily influences latency.
To allow streamers to easily configure OBS for best performance with Livepeer
Studio, we’ve created an OBS preset that can be selected in the OBS Settings
panel.
In-browser broadcasting
Livepeer Studio allows users to broadcast from within their browser using WebRTC broadcast. These broadcasts are optimized for low latency by default and can play back in HLS (8-10s latency) or WebRTC (0.5 - 3s latency). You can leverage this capability in three ways:- Implement the customizable React component
- Embed a hosted version
- Roll your own broadcasting experience with plain WebRTC
Smoke testing your workflow
When you first implement your livestreaming workflow, you may see higher-than-expected latency. This is common and likely means that a few settings need to be tweaked.Check latency using the Livepeer Player
Go tohttps://lvpr.tv/?v=<playbackId> and observe the latency. This will help
isolate the cause of the latency.
If you are seeing higher-than-expected latency on the Livepeer Player (>15s for
HLS or >4s for WebRTC), it suggests something about the incoming stream is
causing high latency. Check your keyframe interval, bitrate, and b-frame
settings.
Viewers changing the resolution will also impact the latency for WebRTC, since
this will necessarily incur transcode latency.