These are the protocols that make up the full WebRTC stack. The listed Sub-protocols are mentioned in the WebRTC spec.

API spec:https://w3c.github.io/webrtc-pc/

Draft overview:https://tools.ietf.org/wg/rtcweb/

WebRTC overview:https://tools.ietf.org/html/draft-ietf-rtcweb-overview

Transports overview:https://tools.ietf.org/html/draft-ietf-rtcweb-transports

Spec:RFC4566

Usage: Used to negotiate capabilities between peers. The capabilities include but are not limited to: networking details, media tracks, data channel information.

Extensions

• RFC3264 An Offer/Answer Model
• JSEP JavaScript Session Establishment Protocol
• RFC5576 Source-Specific Media Attributes (ssrc, ssrc-group, cname and fmtp)
• RFC4574 Label Attribute
• draft-ietf-mmusic-ice-sip-sdp ICE Candidates
• draft-alvestrand-mmusic-msid Cross Session Stream Identification (msid-semantic and WMS)
• draft-ietf-mmusic-msid Grouping Framework (group and mid)
• RFC8122 TCP/TLS and SDP 'fingerprint' attribute.

Spec:draft-ietf-mmusic-sdp-bundle-negotiation

API spec:http://draft.ortc.org/

Spec:draft-ietf-rtcweb-data-channel

Usage: Non-media data transport using WebRTC.

Sub-specs:

• draft-ietf-rtcweb-data-protocol

Spec:RFC8445

Usage: Used to establish a p2p connection by gathering network addresses using STUN and TURN and trying different combinations.

Sub-specs:

• IPv6 support
• RFC6544: ICE-TCP candidates
  • RFC4571: RTP Framing
• Trickle ICE
• Dual Stack Fairness

Spec:RFC5389

Usage: Used to identify your external IP address and keep a NAT binding alive until a direct connection to a peer can be established.

Sub-specs:

• RFC7064 URI Scheme for STUN
• IPv6 support (part of the main spec).
• (TLS-over-)TCP
• ALTERNATE-SERVER

Spec:RFC5766

Usage: Used to relay network traffic through the TRUN server. This is required in networks where direct p2p connection cannot be established due to highly restrictive NAT(s).

Sub-specs:

• RFC7065 URI Scheme for TURN
• RFC6156: TURN Extension for IPv6
• (TLS-over-)TCP Sending over TCP or TLS-over-TCP

Spec:RFC3711 The Secure Real-time Transport Protocol

Usage: The real-time transport protocol provides end-to-end network transport functions suitable for applications transmitting real-time data, such as audio or video.

Sub-specs:

• RFC5761 Multiplexing RTP data packets and RTP Control Protocol
• RFC7022 Canonical Names (CNAMEs)
• RFC8108 Sending Multiple RTP Streams in a Single RTP Session
• RFC3611 RTCP XR
• RFC5763 Framework for Establishing a SRTP Security Context Using DTLS

Spec:RFC6347

Usage: Datagram Transport Layer Security

Sub-specs:

• RFC5246 TLS 1.2
• DTLS-SRTP Multiplexing
• [RFC7983] Multiplexing Scheme Updates for SRTP Extension for DTLS

Spec:RFC4960

Usage: Stream Control Transmission Protocol

Sub-specs:

• RFC3758: SCTP Partial Reliability Extension
• RFC7496: Additional Policies for Partially Reliable Extension
• RFC6525: SCTP Stream Reconfiguration
• RFC8260: Stream Schedulers and User Message Interleaving for the SCTP
• RFC8261: DTLS Encapsulation of SCTP Packets
• RFC4821: Packetization Layer Path MTU Discovery
• RFC4820: Padding Chunk and Parameter for SCTP
• RFC5061: SCTP Dynamic Address Reconfiguration
• RFC4895: Authenticated Chunks

API:webrtc-quic

Usage: QUIC API for Peer-to-peer Connections

Sub-specs:

• draft-ietf-quic-transport
• QUIC MultiplexingLacking?
• RTP over QUIC