Here are the components necessary for establishing WebRTC over UDP and their roles in securing the connection, guaranteeing delivery, and other aspects:

1. Signaling:
   - Signaling is used to exchange session descriptions (SDP) and ICE candidates between peers.
   - It is not part of the WebRTC specification and can be implemented using various protocols like WebSocket, HTTP, or a custom signaling server.
   - Signaling is required for establishing the initial connection but does not directly impact security or delivery guarantees.

2. ICE (Interactive Connectivity Establishment):
   - ICE is a framework used to establish a direct connection between peers.
   - It uses STUN (Session Traversal Utilities for NAT) and TURN (Traversal Using Relays around NAT) servers to discover and negotiate the best path for communication.
   - ICE is responsible for traversing NATs and firewalls to enable direct peer-to-peer connectivity.
   - It is required for establishing the connection but does not provide security or delivery guarantees on its own.

3. DTLS (Datagram Transport Layer Security):
   - DTLS is used to secure the WebRTC connection over UDP.
   - It provides encryption, authentication, and integrity for the data exchanged between peers.
   - DTLS is responsible for securing the connection and preventing eavesdropping and tampering.
   - It is required for establishing a secure WebRTC connection.

4. SRTP (Secure Real-time Transport Protocol):
   - SRTP is used for encrypting and authenticating media packets (audio and video) in WebRTC.
   - It ensures the confidentiality and integrity of the media streams.
   - SRTP is responsible for securing the media transmission and preventing unauthorized access.
   - It is required for secure media transmission over WebRTC.

5. UDP (User Datagram Protocol):
   - UDP is the underlying transport protocol used for WebRTC's real-time communication.
   - It provides low-latency and efficient transmission of data packets.
   - UDP does not guarantee reliable delivery or ordering of packets.
   - It is not responsible for securing the connection or providing delivery guarantees.

6. SCTP (Stream Control Transmission Protocol):
   - SCTP can be used over DTLS for reliable data transmission in WebRTC.
   - It provides features like message orientation, multi-streaming, and congestion control.
   - SCTP ensures reliable delivery and ordering of data packets.
   - It is optional and used when reliable data transmission is required.

7. RTP (Real-time Transport Protocol):
   - RTP is used for transmitting real-time media data (audio and video) over UDP.
   - It provides timestamps, sequence numbers, and payload type identification for media packets.
   - RTP itself does not provide security or delivery guarantees.
   - It is used in conjunction with SRTP for secure media transmission.

In summary:
- DTLS and SRTP are responsible for securing the WebRTC connection and media transmission.
- ICE is required for establishing the direct peer-to-peer connection but does not provide security or delivery guarantees.
- SCTP can be used for reliable data transmission when required.
- UDP is the underlying transport protocol that enables low-latency communication but does not guarantee reliable delivery or provide security on its own.
- Signaling, RTP, and ICE are necessary components for establishing and maintaining the WebRTC connection but do not directly handle security or delivery guarantees.