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When Hulu’s Oscars stream went dark during the Best Picture reveal in March 2025, millions missed the moment. Netflix’s live boxing event? Buffering chaos. Even the biggest platforms are learning live streaming at scale is a different beast.
Audiences expect real-time, high-quality video with zero tolerance for delays or errors. Whether it’s a global esports final or a regional football match, one playback hiccup is all it takes to lose trust.
This guide breaks down what it takes to deliver reliable live streams at scale across devices, networks, and geographies. From capture and encoding to delivery and playback, we’ll walk through the modern live streaming workflow and where protocols like RTMP, HLS, WebRTC, and SRT fit into the picture.
Because once you go live, there’s no “retry.” We will talk about how Live streaming actually happens, the protocol and how to live stream with FastPix. But before that lets talk about why high traffic events are a real challenge to do…
Running a live stream for thousands or millions of viewers isn’t just about turning on a camera. It’s about handling unpredictable spikes, keeping latency low, and making sure the stream doesn’t fail when it matters most.
Scalability under pressure: Traffic doesn’t ramp up gradually. It spikes right as a goal is scored, a headline drops, or the artist walks on stage. Without dynamic scaling or a multi-CDN setup, your origin can get overloaded, leading to buffering or total failure.
Latency that breaks the moment: When you're streaming esports, sports betting, or interactive formats, even a few seconds of delay can ruin the experience. Chat reactions arrive late. Bets close too slowly. It kills engagement. Low-latency protocols like SRT and WebRTC help, but only if your full pipeline is tuned for it.
Quality vs. bandwidth tradeoffs: Viewers expect 1080p or 4K. But not everyone has the connection to support it. Without properly tuned ABR (adaptive bitrate), you’re either wasting bandwidth or frustrating users with buffering. The challenge is delivering just enough quality without pushing users over the edge.
Network instability: You can’t control the network, especially at outdoor events or remote locations. Drops, jitter, and congestion all happen sometimes mid-stream. If your protocol can’t recover from that (like RTMP often can't), viewers experience stalls or blackouts. Error-correcting protocols like SRT can help maintain uptime.
All these challenges buffering, delays, quality drops—don’t happen by accident. They’re usually caused by issues in the live streaming pipeline. To fix them, you need to understand how a stream actually works, from camera to screen. Because what looks like a small glitch on the viewer’s end often starts much earlier in the workflow.
Let’s break down how live streaming works, step by step.
At its core, every live stream follows a five-stage pipeline: Capture → Encode → Process & Package → Distribute → Playback. It sounds simple, but each step is doing a lot of heavy lifting ensuring the stream is high quality, low latency, and stable across devices and regions.
Each stage in this workflow is optimized for a specific goal: reducing latency, maintaining visual fidelity, scaling under traffic spikes, and delivering a smooth viewer experience. Let’s talk about them in detailed.
Before encoding or delivery, every live stream begins with how well you capture the raw signal. If audio and video aren’t clean and synced at the source, there’s no fixing it downstream.
Take the 2023 Coachella Festival, streamed live on YouTube. The production team used professional 4K cameras for stage and crowd shots, drones for aerials, and directional microphones to isolate vocals and instruments. All video feeds were synced using genlock, and audio was time-aligned with LTC timecode to keep everything in perfect sync.
To ensure reliability, they built redundancy into every layer backup cameras, duplicate audio paths, and failover encoders. Even drone feeds were stabilized using bonded cellular links, so a dropped signal wouldn’t interrupt the stream.
In short, the Coachella stream wasn’t just about high quality it was about engineering for resilience at scale.
Once video and audio are captured, they need to be compressed into streamable formats. This step called encoding reduces file size while preserving quality, making it possible to deliver content smoothly across different devices and network conditions.
Most live streams use H.264 for compatibility or H.265 for better efficiency, especially for 4K content. To support fluctuating bandwidth, encoding creates multiple renditions of the same stream at different qualities. This process, known as adaptive bitrate encoding, ensures viewers always get the best possible experience without buffering.
The encoded stream is then pushed to a streaming server using a contribution protocol. RTMP is still common, though it has higher latency. SRT is preferred for more reliable, lower-latency delivery. WebRTC is used when real-time interaction is needed, such as during live Q&A or interactive events.
To avoid failures during live broadcasts, most production teams run at least two encoders in parallel and carefully test their bitrate ladders ahead of time. Some workflows also use AI-powered encoding to adjust bitrate dynamically and improve quality without increasing file size.
Real-world example
For the 2022 FIFA World Cup, broadcasters like FOX Sports used cloud encoding to handle live feeds from multiple stadiums. Services like AWS Elemental MediaLive allowed them to process streams in real time, generate adaptive renditions, and deliver them globally with minimal delay. While AWS provided the scale and reliability needed for an event of this size, the setup required deep expertise, custom workflows, and careful configuration to handle region-level failover and dynamic load across encoding nodes.
After encoding, the video stream is prepared for delivery. This involves segmenting it into smaller chunks, applying adaptive bitrate profiles, and packaging it into formats compatible with different devices and players. This stage also adds essential enhancements like subtitles, metadata, and DRM to improve the viewing experience and protect content.
During processing, each stream is broken into short segments, typically 2 to 6 seconds in length, and organized into manifest files that guide playback. HLS (used by Apple devices) and MPEG-DASH (used widely across modern platforms) are the most common packaging formats. Both support adaptive streaming, but MPEG-DASH offers broader codec flexibility, especially for H.265 and VP9.
Enhancements like multilingual subtitles and metadata markers are added here as well. Subtitles help make content accessible to wider audiences, while event markers allow for features like instant replays, chapter navigation, or highlights. DRM is applied at this stage too using protocols like Widevine or FairPlay to prevent piracy and unauthorized distribution, especially for paid or premium content.
Testing for playback consistency across devices phones, browsers, smart TVs is critical. Automated tools can also help tag metadata or generate subtitles using speech recognition or computer vision, reducing the need for manual post-processing.
Real-world example
The 2024 Paris Olympics used HLS packaging with multi-language subtitle support to deliver streams at scale. Global broadcasters ensured that viewers could access content in their preferred language with minimal buffering, while DRM protected premium live feeds. While the architecture was robust, it also required significant setup, tuning, and device testing to meet global standards.
Once streams are packaged, they need to be delivered quickly and reliably to viewers around the world. This is where CDNs (Content Delivery Networks ) come in. CDNs cache video segments on edge servers, placing the content physically closer to users to reduce latency, improve load times, and absorb traffic spikes during peak moments.
Most video delivery today uses adaptive streaming formats like HLS and MPEG-DASH, which CDNs are optimized to serve. Each segment is requested as needed, based on the viewer’s bandwidth and device. While formats like WebRTC or SRT are also used in special low-latency cases, they’re less common for mass delivery and more often used during contribution or for interactive layers like live Q&A.
For events with global viewership, a single CDN is often not enough. Multi-CDN strategies combine providers across regions to ensure consistent performance and failover. A stream might use Akamai for North America, Cloudflare in Europe, and FastPix’s own CDN nodes for Asia automatically routing traffic to the fastest and most available edge point. This helps minimize buffering, avoids regional congestion, and ensures streams stay resilient even under sudden surges in demand.
To make this work smoothly, edge caching is critical. Segments must be pre-positioned or quickly fetched to avoid stressing the origin server. Many teams also rely on real-time traffic analytics to anticipate spikes—like a concert finale or game-winning play—and scale CDN resources in advance. Monitoring tools track delivery metrics like startup time, stall ratio, and throughput to identify issues before they affect users.
Real-world example
Twitch’s broadcast of the 2023 Dota 2 International reached millions of simultaneous viewers. To keep latency low and playback smooth, Twitch leaned heavily on a distributed CDN architecture. Edge caching allowed content to load instantly, while multi-region routing ensured viewers in different parts of the world experienced consistent quality—even during critical, high-traffic moments like championship matches.
The final stage of the streaming pipeline is where everything comes together for the viewer. Whether through a website, mobile app, or smart TV platform, playback is where performance, quality, and engagement are felt most directly. If anything goes wrong here buffering, sync issues, latency spikes the user blames the product, not the pipeline.
Most modern platforms rely on HLS for playback, thanks to its adaptive bitrate support and broad compatibility across browsers and devices. Regardless of protocol, the goal is to deliver smooth playback while adapting to each viewer’s device and network conditions in real time.
User experience is no longer just about watching video it’s about interaction. Features like live chat, polls, and real-time stats have become standard in esports and live sports. Behind the scenes, observability tools monitor playback health tracking buffering, stall rates, and drop-offs to help teams detect issues and optimize performance.
Playback optimization also depends heavily on device testing. Mobile viewers now make up a majority of stream traffic, especially in emerging markets. Ensuring fast startup time, responsive UI, and low memory usage on phones and smart TVs is as critical as CDN performance or encoder tuning.
Real-world example
Super Bowl LVIII, streamed on Paramount+ in 2024, reached over 123 million viewers. The stream used HLS for playback and supported features like multi-angle views, live stats, and instant replays creating a deeply interactive experience. Paramount combined a high-redundancy delivery pipeline with real-time monitoring to maintain quality across all devices, setting a new bar for large-scale sports streaming.
Now that we understand the full streaming process from capture to playback let’s move on to the protocols. Each one brings different strengths to the table, and choosing the right set is key to delivering a reliable, high-quality stream.
Now that we understand the full streaming workflow, it’s time to look at the protocols that move video from camera to viewer. Each protocol is built for a specific part of the pipeline whether it's contribution, distribution, or playback and each comes with trade-offs in latency, compatibility, and scalability.
RTMP remains one of the most common protocols for pushing video from encoders to servers. It’s supported by most tools like OBS and Wirecast, making it easy to set up for live events. RTMPS, its secure version, adds encryption during transmission.
While both offer convenience and broad compatibility, they typically introduce 1–3 seconds of latency and aren’t ideal for unstable networks or interactive use cases. Still, they’re a solid choice for concerts, webinars, and one-way live broadcasts.
HLS and MPEG-DASH are the backbone of large-scale video delivery. These protocols segment streams into small chunks and support adaptive bitrate playback across a wide range of devices. HLS is widely supported in Apple ecosystems, while MPEG-DASH offers broader codec flexibility. Standard latency for HLS is around 5 to 10 seconds, though low-latency HLS can bring it down to 2–3 seconds. These protocols power everything from Disney+ Hotstar’s IPL streams to Netflix’s global content delivery.
SRT (Secure Reliable Transport), developed by Haivision, is designed for contribution not playback. It offers low latency (200–400ms), strong error recovery, and encryption over unpredictable networks. SRT excels in remote production environments like mountain biking races or live field reporting, where internet conditions can be inconsistent. It’s increasingly used to replace RTMP in professional workflows but requires compatible encoders and infrastructure.
WebRTC delivers sub-second latency, usually around 200 to 500 milliseconds, and works natively in modern browsers. It’s perfect for interactive formats like live chats during esports streams, auctions, or Q&A sessions where responsiveness is key. However, it’s harder to scale to millions of viewers due to its peer-to-peer foundation and infrastructure demands.
A note on latency
Latency varies widely based on the protocol, encoding settings, and network conditions. RTMP may introduce a few seconds of delay, while WebRTC aims for sub-second interaction. HLS and DASH prioritize scalability over speed. Choosing the right protocol comes down to what matters most for your stream reach, quality, or real-time responsiveness.
For live concerts, festivals, and large-audience streams, stability matters just as much as speed. That’s why we recommend SRT (Secure Reliable Transport) as the go-to protocol for contribution in high-traffic environments.
SRT delivers the low latency of WebRTC but adds something critical: resilience. It recovers from packet loss, adjusts to changing bandwidth, and includes built-in encryption of your stream stays stable, even on unpredictable networks. Whether you’re streaming from a crowded venue, a mobile unit, or a remote stage, SRT holds up where other protocols drop off.
That’s why FastPix supports SRT across all ingest endpoints. It helps you push high-quality video into our platform with fewer interruptions and without needing complex workarounds. It just works, even when the network doesn’t.
Next, let’s walk through how to live stream using SRT with FastPix.
FastPix supports live streaming over SRT, making it ideal for high-traffic events like concerts, esports, and religious broadcasts. With built-in AES-128 encryption, multi-CDN delivery, and API support for generating VOD clips from live streams, SRT is both reliable and production-ready.
In your encoder (OBS, vMix, or Wirecast), set your stream output to the following SRT URL:
srt://live.fastpix.io:778?streamid=<Stream Key>&passphrase=<Secret Key> Replace <Stream Key> and <Secret Key> with the values from your FastPix dashboard. This endpoint securely pushes your live feed into the FastPix platform.
To test SRT playback, construct the playback URL like this:
srt://live.fastpix.io:778?streamid=<Playback Stream ID>&passphrase=<Playback Secret>
You can test playback using VLC or FFmpeg. For FFmpeg, use:
ffplay -analyzeduration 1 -fflags -nobuffer -probesize 32 -sync ext "<SRT_PLAYBACK_URL>"Make sure port 778 (UDP) is open in your firewall to allow SRT traffic.
If you want to go live with RTMPS, we’ve got that covered too. FastPix supports both SRT and RTMPS out of the box, giving you flexibility based on your encoder setup and latency requirements. You can also enable simulcasting to push your live stream to multiple platforms like YouTube, Twitch, and Facebook directly from a single input.
And if you’re building a custom live video experience, FastPix offers APIs for ingest, real-time playback, live to VOD generation, and analytics all designed to scale from one-off events to full streaming platforms.
To learn more, check out our Live Streaming Docs and Guides.
Streaming to thousands or millions shouldn’t feel risky. Whether you need low-latency ingest, multi-CDN delivery, simulcasting or live clipping. FastPix helps you go live with confidence. Reach out to see how we can support your high-traffic streams from start to scale.
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