1. Selective Forwarding Units (SFUs)
In a basic WebRTC setup, peers connect directly to each other. If one broadcaster wants to stream to 1000 viewers, a direct mesh network crashes their upload bandwidth immediately. The best WebRTC tools utilize SFUs—servers that receive one high-quality upload stream from the broadcaster and independently forward it to the 1000 viewers. Open-source SFUs like Mediasoup and Pion are heavily favored by developers for their performance and customizability.
2. Managed WebRTC Networks (Agora & LiveKit)
Deploying and scaling STUN/TURN servers across the globe to handle NAT traversal is incredibly difficult DevOps work. The best approach for many teams is utilizing managed WebRTC networks like LiveKit or Agora. LiveKit, in particular, has emerged as an incredible open-source core with a managed cloud tier, offering React SDKs that make integrating real-time video into a Next.js app as simple as importing a component.
3. WebRTC vs Low-Latency HLS
Developers often debate WebRTC vs LL-HLS. LL-HLS can achieve 2-4 second latency over massive HTTP CDNs, which scales cheaper than WebRTC. However, if true real-time interactivity (like voice chat or remote control) is required, WebRTC is the only viable option. The best streaming platforms often use WebRTC for the interactive elements and fallback to HLS for larger, passive audience broadcast feeds.