WebRTC Data Channels vs WebSockets: P2P, SCTP & Configurable Reliability
When real-time delivery goes peer-to-peer: how WebRTC data channels differ from WebSockets — SCTP-over-DTLS-over-UDP transport, configurable reliability and ordering, signaling (often carried over a WebSocket), NAT traversal with STUN/TURN, and mandatory encryption. The reliability modes are verified in Node.
Questions
- Not answered. How does a WebRTC data channel's topology fundamentally differ from a WebSocket's?
- Not answered. WebRTC peers must first exchange SDP offers/answers and ICE candidates. What does the spec say about transporting this "signaling"?
- Not answered. WebRTC data channels carry application data over which message-oriented transport protocol (itself layered on DTLS/UDP)?
- Not answered. Which statements about
RTCDataChannelreliability and ordering are correct? - Not answered. A fast multiplayer game sends 60 position updates per second. Why might you choose an unreliable, unordered data channel?
- Not answered. Two browsers behind home routers want a direct peer-to-peer data channel. What do STUN and TURN each do?
- Not answered. WebRTC data channels are protected with DTLS. Is encryption optional or mandatory in WebRTC? (one word)
- Not answered. Like WebSocket, a data channel hands you whole messages, not a raw byte stream. Which SCTP property provides that?
- Not answered. You need server-authoritative real-time messaging — the server must validate and broadcast every message. WebSocket or WebRTC data channel, and why?
- Not answered. Which statements correctly contrast WebRTC data channels with WebSockets?
- Not answered. What is the correct protocol stack beneath a WebRTC data channel message?
- Not answered. What value of
maxRetransmitsconfigures a fully unreliable (fire-and-forget) data channel that never retransmits a lost message?