Cloud Computing 15 min read

How Ultra‑Low Latency Live Streaming Cuts Delay by 90% with WebRTC

The Tencent Cloud and China Academy of Information and Communications Technology whitepaper introduces ultra‑low latency live streaming technology, detailing its WebRTC‑based architecture, miniSDP compression, AV1 support, adaptive bitrate, and real‑world applications in e‑commerce, cross‑domain events, and education, while outlining future trends and industry impact.

Tencent Architect
Tencent Architect
Tencent Architect
How Ultra‑Low Latency Live Streaming Cuts Delay by 90% with WebRTC

Whitepaper Release and Industry Background

On February 22, Tencent Cloud and the China Academy of Information and Communications Technology jointly released the first "Ultra‑Low Latency Live Streaming (Fast Live)" whitepaper, which defines technical standards for ultra‑low latency streaming and lowers development barriers. The report outlines the growth of live streaming, its application in e‑commerce, education, social, sports, gaming, tourism, and cultural dissemination, and highlights the need for sub‑second latency.

1. Limitations of Standard Live Streaming

Traditional live streaming relies on fixed buffers and fully reliable TCP‑based protocols (RTMP, HTTP‑FLV, HLS), causing high latency, increased stutter under weak networks, and inability to prioritize video frames. These constraints limit QoS/QoE improvements such as edge acceleration and adaptive bitrate.

Standard live streaming playback model

2. Ultra‑Low Latency Model (Fast Live)

Fast Live replaces the traditional transmission‑playback model with a WebRTC‑based feedback loop, allowing real‑time adjustment of buffering and transmission strategies based on network conditions, achieving sub‑500 ms latency while maintaining quality.

Ultra‑low latency playback model

3. Technical Optimizations

Signalling Compression : MiniSDP binary compression reduces SDP size to fit within a single UDP packet, enabling 0‑RTT handshakes and cutting signalling latency by up to 70%.

Audio/Video Codec Support : Full AAC support, H.265 (HEVC) integration, and B‑frame handling improve quality and reduce transcoding costs.

Flexible Tiered Transmission : Adaptive packet loss handling and tiered frame dropping lower bitrate under weak networks without affecting all viewers.

Adaptive Pacing and Bitrate : Client‑side cache information drives smooth packet emission and Simulcast/ABR strategies.

P2P Distribution : Leveraging WebRTC’s native peer‑to‑peer capability creates local sharing clusters, reducing CDN load.

Private Data Passthrough & Optional Encryption : RTP extensions allow metadata, SEI, and custom NALU transmission; encryption can be toggled to save bandwidth.

Media Processing Optimisation : AI‑assisted pre‑processing reduces I‑frame size while preserving visual quality.

4. Application Scenarios

E‑commerce Live : Sub‑second interaction enables real‑time product queries, flash sales, and higher conversion rates.

Cross‑Domain Events : Multi‑venue streaming with seamless multi‑camera switching and real‑time synchronization.

Education : Millisecond‑level latency supports large‑scale interactive classrooms with quizzes, polls, and whiteboards.

5. Future Outlook

Continued upgrades in edge computing and network architecture will further reduce costs, enabling massive adoption of ultra‑low latency streaming across industries such as e‑commerce, enterprise events, and specialized live productions.

cloud computinglow-latency streamingreal-time communicationWebRTCe-commerce liveeducation streaming
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