DRM Digital Rights Management: Principles and Implementation
Deep dive into the working principles, browser support, and integration methods of three major DRM systems: Widevine, FairPlay, and PlayReady.
Read More →Deep dive into the technical principles and selection strategies of mainstream low-latency live streaming solutions
Live streaming latency refers to the time difference from signal capture to playback on the viewer's side. It is typically categorized as glass-to-glass latency and end-to-end latency. Based on latency magnitude, live streaming solutions can be classified as follows:
Different use cases have vastly different latency requirements. E-commerce live streaming needs 2–3 seconds of latency to accommodate bullet chat interaction, while sports event streaming can accept 5–10 seconds of latency. Choosing the right low-latency technology requires finding a balance between latency, cost, compatibility, and video quality.
WebRTC (Web Real-Time Communication) is a real-time communication technology initiated by Google. Originally designed for video calls between browsers, it is now widely used in low-latency live streaming scenarios.
WebRTC live streaming typically employs an SFU (Selective Forwarding Unit) architecture. The publisher pushes audio and video streams to an SFU server, which is responsible for forwarding the stream to each viewer. Compared to an MCU architecture, the SFU only needs to forward streams without re-encoding, resulting in lower latency and better scalability.
// WebRTC playback side sample code
const peerConnection = new RTCPeerConnection({
iceServers: [
{ urls: 'stun:stun.l.google.com:19302' }
]
});
peerConnection.ontrack = (event) => {
const video = document.getElementById('video');
video.srcObject = event.streams[0];
};
// Receive remote SDP and answer
async function receiveOffer(offerSdp) {
await peerConnection.setRemoteDescription(
new RTCSessionDescription(offerSdp)
);
const answer = await peerConnection.createAnswer();
await peerConnection.setLocalDescription(answer);
return answer;
}LL-HLS (Low-Latency HLS) is a low-latency HLS extension introduced by Apple in 2019. By introducing partial segments and incremental playlist updates, it reduces HLS latency from the traditional 10–30 seconds down to 2–5 seconds.
#EXTM3U
#EXT-X-VERSION:9
#EXT-X-TARGETDURATION:2
#EXT-X-MAP:URI="init.mp4"
#EXT-X-SERVER-CONTROL:CAN-BLOCK-RELOAD=YES,PART-HOLD-BACK=1.0,CAN-SKIP-UNTIL=12.0
#EXT-X-PART-INF:PART-TARGET=0.5
#EXT-X-MEDIA-SEQUENCE:100
#EXT-X-GAP
#EXTINF:2.0,
gap_100.mp4
#EXT-X-PROGRAM-DATE-TIME:2026-05-05T12:00:00.000Z
#EXTINF:2.0,
file101.mp4
#EXT-X-PART:URI="part102_1.mp4",DURATION=0.5,INDEPENDENT=YES
#EXT-X-PART:URI="part102_2.mp4",DURATION=0.5
#EXT-X-PART:URI="part102_3.mp4",DURATION=0.5
#EXT-X-PRELOAD-HINT:TYPE=PART,URI="part102_4.mp4",BYTERANGE-START=0SRT (Secure Reliable Transport) is an open-source transport protocol developed by Haivision. Implemented over UDP, it combines the reliability of TCP with the low-latency characteristics of UDP, and is widely used in broadcasting and professional live streaming fields.
Below is a comparison of the latency performance of these three technologies under typical configurations:
When choosing a low-latency live streaming technology, consider the following dimensions comprehensively:
If you need ultra-low latency under 1 second, WebRTC is the top choice. If you can accept 2–5 seconds of latency, LL-HLS offers advantages in compatibility and distribution cost.
WebRTC's SFU architecture has limited capacity per server (typically 1,000–5,000 concurrent users), and large-scale distribution requires cascading or CDN integration. LL-HLS, based on HTTP, can fully leverage existing CDN infrastructure and easily support millions of concurrent viewers.
WebRTC has a complex technology stack, requiring handling of signaling, NAT traversal, packet loss recovery, and other issues — resulting in higher development and maintenance costs. LL-HLS, as an extension of standard HLS, is relatively simpler to develop and has a more mature ecosystem.
In many scenarios, the optimal approach combines multiple technologies, for example:
Each low-latency live streaming technology has its strengths and weaknesses — there is no silver bullet. WebRTC performs best in terms of latency but has high distribution costs; LL-HLS offers good compatibility and easy distribution but relatively higher latency; SRT has advantages in professional transmission but limited end-device support.
In practice, your choice should be based on business scenarios, latency requirements, audience scale, and your technical team's capabilities. For most consumer-facing live streaming applications, LL-HLS is a balanced choice; for strongly interactive scenarios, WebRTC is essential; and in professional live streaming and signal transmission, SRT is the best companion.
If you want to quickly experience HLS playback, you can use our M3U8 Online Player to play HLS video streams directly in your browser without writing any code.
More great articles and useful tools
Deep dive into the working principles, browser support, and integration methods of three major DRM systems: Widevine, FairPlay, and PlayReady.
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Read More →Use our free M3U8 Online Player — no coding required, play HLS video streams directly in your browser
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