STUN and TURN servers are two types of WebRTC signaling servers that can be used to create your peer-to-peer (P2P) connection when you are building a real-time communication application. Signaling is not part of the WebRTC standard and thus needs to be handled separately if you are building the WebRTC based application yourself rather than using a commercial platform or Video API Provider

The protocols used to realize connections over the network communication protocols are STUN, TURN, and TURNS.

Session Traversal Utilities for NAT (STUN) - Used to establish a direct UDP connection between two clients.

Traversal Using Relay around NAT (TURN) - Used to establish a relayed UDP or TCP connection between two clients. Here, the traffic must be relayed through the TURN server to bypass restrictive firewall rules, and the preference is UDP over TCP because TCP's guaranteed ordered delivery of packets implies overhead that is undesirable for real-time communications.

Secure Traversal Using Relay around NAT (TURNS) - Used to establish a relayed TCP/TLS connection between two clients. Here, the traffic must be relayed through the TURN server and through a TLS socket to bypass extremely restrictive firewall rules.

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