PRODUCTS
62% of viewers leave live streams because of delay or buffering
Latency isn’t just a number. It affects how people watch, react, and stay engaged.
If a stream is delayed by even a few seconds, it can break the experience, especially for sports, auctions, events, and anything time-sensitive. Most streaming platforms still use workflows built for video-on-demand, which adds unnecessary delay.
At FastPix, we help developers build low-latency live apps with full monitoring, and API-level control, without managing custom infrastructure.
In this guide, we’ll cover:
If you’re building anything live, latency isn’t a backend detail. It’s a product decision.
Latency is the delay between the time a video is captured by a camera and when it's displayed on a viewer’s screen. In live streaming, low latency means real-time or near-real-time delivery of video content.
Here's a simple breakdown:
.png)
The protocol you choose impacts more than just how fast your stream loads. It defines how interactive your experience can be, how much infrastructure you’ll need, and how reliably you can scale.
Let’s look at three commonly used protocols for low-latency live streaming, and when each one fits.
RTMPS
RTMPS is the secure version of RTMP, the long-standing protocol used to push live video to platforms like Facebook Live, YouTube, and Twitch. It’s a publishing protocol, mainly used to send video from an encoder to an ingest server.
RTMPS is simple to set up, widely supported by encoders (like OBS, Wirecast, or vMix), and integrates easily with streaming platforms. It includes SSL encryption by default, making it safer for public delivery than its predecessor.
However, RTMPS is not designed for playback. It’s meant to push streams into a system, not deliver them to viewers. To reach an audience, you’ll still need a playback protocol like HLS or DASH downstream.
Best for: sending live video from the source to your media pipeline
Not ideal for: playback delivery, large-scale interaction, in-browser viewing
SRT
SRT is built for delivering high-quality video over unpredictable networks. It shines in scenarios like remote production, where you need to move a live feed from the field to a central control room.
It supports packet recovery, encryption, and error correction, making it more resilient than RTMP. But like RTMPS, it’s focused on contribution, not playback. You’ll need dedicated infrastructure and player support to view or distribute streams sent via SRT.
Best for: remote contribution, broadcast workflows, studio ingest
Not ideal for: browser playback, end-user streaming, interactive experiences
LL-HLS
Low-Latency HLS is Apple’s evolution of HLS for live delivery. It keeps the scalability and CDN compatibility of traditional HLS while reducing the delay between stream and playback.
It’s ideal for large audiences, sporting events, concerts, or public broadcasts, where interactivity isn’t the main concern. LL-HLS works with most modern players and supports partial segments for smoother delivery, but it’s not suitable for real-time apps like live chat or auctions.
Best for: wide reach, public streams, consistent playback quality
Not ideal for: real-time feedback loops, small-group interaction
Even with smarter encoders, edge CDNs, and protocol improvements, low-latency streaming remains complex. Here’s why:
Fragmented infrastructure: Most live setups involve multiple vendors, one for ingest, another for encoding, another for delivery. Every integration point adds overhead and delay.
Limited visibility: When latency spikes, it’s hard to pinpoint the cause. Is it the encoder buffer? CDN edge? Player startup? Traditional pipelines don’t expose these details.
Performance tradeoffs: Chasing speed often means cutting corners, lowering bitrate, disabling adaptive quality, or compromising on network resilience.
How APIs help reduce live streaming latency
Reducing latency isn’t just about choosing the right protocol, it’s about controlling the entire workflow. That’s where APIs make a difference.
With an API-based architecture, developers can:
This level of control is hard to achieve with fragmented tools and services. APIs make it easier to build for speed, and maintain it as you scale.
FastPix brings this API-first approach to live streaming. It gives you a complete video pipeline, ingest to playback, accessible entirely through APIs.
Here's how FastPix optimizes for low latency out of the box:
Here’s what a typical low-latency architecture looks like with FastPix:
Camera → FastPix Ingest (RTMPS/SRT) → JIT Encoder → LL-HLS Packager → CDN → FastPix Player SDK → Viewer
Everything from ingest to playback is handled by FastPix, via APIs. You don’t need to provision servers, configure encoders, or manage playback compatibility.
Here’s how to build it:
After signing up, head to your FastPix dashboard and generate a new Access Token from the Organization settings > Access token tab.
You’ll get a Token ID and Secret Key, store them securely. FastPix doesn’t save the full secret.
Make a POST request to /live/streams using your Access Token credentials. You can define settings like resolution, reconnect window, and metadata.
curl -X POST https://v1.fastpix.io/live/streams \
--user {ACCESS_TOKEN_ID}:{SECRET_KEY} \
-H 'content-type: application/json' \
-d '{
"playbackSettings": {
"accessPolicy": "public"
},
"inputMediaSettings": {
"maxResolution": "1080p",
"reconnectWindow": 60,
"mediaPolicy": "public",
"metadata": {
"livestream_name": "live_demo"
}
}
}'
Response includes: streamKey, playbackId, RTMPS/SRT endpoint, and stream status
Use the streamKey and FastPix RTMPS or SRT endpoint in your preferred encoder:
OBS Example:
Server: rtmps://live.fastpix.io:443/live
Stream Key: (from API response)
FFmpeg example:
ffmpeg -re -i input.mp4 -c:v libx264 -f flv rtmps://live.fastpix.io:443/live/<stream-key>
As soon as the broadcast starts, FastPix transitions the stream status to active and begins packaging video for playback.
<script src="https://cdn.jsdelivr.net/npm/@fastpix/player@1.0.1/dist/player.js"></script>
<fp-player
stream-type="live-stream"
playback-id="{playbackId}"
primary-color="#F5F5F5"
accent-color="#5D09C7"
secondary-color="#000000"
/>
The player supports LL-HLS by default and includes built-in QoE tracking and analytics support.
When the broadcast ends, stop your encoder. FastPix will automatically update the stream status to idle. If the reconnect window expires or the duration limit is hit, the stream will be marked as disabled.
You can also subscribe to live stream webhooks to get real-time updates on stream status and viewer activity.
Real-time sports streaming with instant highlights
A live football app used FastPix’s Live API and Key Moments detection to clip and surface goals within 30 seconds of broadcast. Fans stayed engaged, highlights were shareable immediately, and editorial teams didn’t need to cut manually.
EdTech that feels like a real classroom
One education platform replaced its video stack with FastPix to support live classes that feel live. With low-latency delivery, automatic Live-to-VOD archiving, and speech-to-text transcripts, students could review sessions instantly, without waiting for uploads or edits.
Smarter live commerce with stream-aware optimization
A retail brand used FastPix to simulcast product drops to both Instagram and their app. By feeding FastPix’s stream health data into their A/B testing engine, they identified device-level buffering issues and improved viewer-to-buyer conversion during peak sales moments.
.png)
Ready to build?
We hope this guide helped you better understand how to approach low-latency live streaming, without the complexity.
If you're building something live, FastPix gives you the tools to move fast and scale with confidence. Sign up today and get $25 in free credits to start streaming. Have questions? Reach out to our team, we’re happy to help.
To maintain quality while keeping latency low, you need to make trade-offs that favor stability and responsiveness. This means using adaptive bitrate streaming with shorter segment durations, optimizing your encoder settings for speed over compression, and choosing a delivery protocol like LL-HLS that supports fast segment transfer without sacrificing consistency. Real-time playback monitoring also plays a key role, by tracking stalls, bitrate shifts, and device-level buffering, you can adapt the experience before users notice a drop in quality.
Reconnect windows act as a buffer zone for stream interruptions. When a broadcast source temporarily goes offline, due to encoder crashes, network instability, or brief hardware issues, the reconnect window keeps the stream session alive for a predefined period. This avoids stream termination and preserves the playback session for viewers. It’s especially useful for interactive formats or long-running events where continuity is critical, as it allows broadcasters to resume without starting from scratch.
Simulcasting is most valuable when you want to centralize your stream control but distribute content to multiple platforms like YouTube, Twitch, or your own app simultaneously. It reduces the strain on your encoder, simplifies workflow management, and gives you full control over quality and performance monitoring. Direct platform streaming might still make sense if you rely heavily on native features like built-in audience engagement tools, platform-specific monetization, or proprietary ad integrations.
For true real-time interaction, most traditional protocols fall short. WebRTC offers the lowest latency and supports two-way communication, making it ideal for use cases like video calls, live auctions, and gaming. However, for broader reach and better compatibility with browsers and mobile devices, a hybrid approach using LL-HLS with fast playback initialization and lightweight signaling can strike a practical balance between speed and scale.
Buffering is usually the result of delayed segment delivery, poor network performance, or slow player initialization. To reduce it, your app should use a CDN with efficient edge caching and a protocol like LL-HLS that supports partial segment fetching. Optimizing encoder output for consistent bitrate and keeping startup delays low helps too. Monitoring playback events in real time allows you to identify exactly where buffering occurs whether it’s at the edge, in the player, or due to last-mile network issues and adjust accordingly.
.png)
.png)
