WebTransport is like a Swiss Army knife for real-time web comms, building on HTTP/2 and HTTP/3 to deliver secure, multiplexed streams with both reliable and unreliable channels. That means you can mix low-latency data (think live gaming or VoIP) with guaranteed-delivery packets (file transfers, chat messages) all over one TLS-encrypted connection.
In the video, Hussein Nasser walks you through why WebSockets, gRPC and even HTTP/2/3 alone fall short, then unpacks WebTransport’s game-changing features—unreliable streaming, the Extended CONNECT upgrade flow, cross-protocol support and proxying considerations—that make it the next big leap for backend engineers.
Watch on YouTube
Top comments (0)