WebRTC is pretty tricky to work with right now. As of June 2013, support in Chrome and Firefox Nightly is fragmented and incomplete. The best explanation of WebRTC that I have found is the WebRTC Spec Draft from the W3C. This pedantic document is filled with jargon specific to Internet Connectivity Establishment. To elucidate the spec, I put together a list of descriptions for the most fundamental concepts.

Proceed with Caution! Both Chrome and Firefox are still a long way from compliance.

ICE - Protocol for Network Address Translation - Interactive Connectivity Establishment (ICE): A Protocol for Network Address Translator (NAT) Traversal for Offer/Answer Protocols. ICE agents include STUN and TURN servers.

STUN - Session Traversal Utilities for NAT (STUN). Peers connect to a stun server to find their public facing addresses, AND other info on how that peer may be reached fron the internet.

TURN - Traversal Using Relays around NAT (TURN). Like a stun server, but can relay data from one peer to another if the clients are unable to connect directly. Every TURN server is also a STUN server.

NAT - Network Address Translation - When a signal passes through a router, the router may alter the port number – effectively creating a “public facing address”. The public address is effectively mapped to a different address behing the router.

SDP - Session Description Protocol - (Wikipedia) - a format for describing streaming media initialization and connectivity parameters.

An SDP example from someRTCPeerConnection.localDescription.sdp looks like this:

v=0
o=Mozilla-SIPUA-24.0a1 15165 0 IN IP4 0.0.0.0
s=SIP Call
t=0 0
a=ice-ufrag:1f8009b2
a=ice-pwd:04f106d867cbba054729ae220b5c9618
a=fingerprint:sha-256 62:5F:18:CD:23:D3:BD:D9:C9:B4:42:BD:3B:07:2A:6A:EE:B0:06:59:6D:A3:B3:C8:E4:FA:99:B3:37:5E:56:FA
m=audio 41438 RTP/SAVPF 109 0 8 101
c=IN IP4 199.87.82.66
a=rtpmap:109 opus/48000/2
a=ptime:20
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=candidate:0 1 UDP 2113667327 10.0.1.46 65468 typ host
a=candidate:1 1 UDP 1694302207 199.87.82.66 41438 typ srflx raddr 10.0.1.46 rport 65468
a=candidate:0 2 UDP 2113667326 10.0.1.46 57425 typ host
a=candidate:1 2 UDP 1694302206 199.87.82.66 38208 typ srflx raddr 10.0.1.46 rport 57425
m=application 37652 SCTP/DTLS 5000
c=IN IP4 199.87.82.66
a=fmtp:5000 protocol=webrtc-datachannel;streams=16
a=sendrecv
a=candidate:0 1 UDP 2113667327 10.0.1.46 55401 typ host
a=candidate:1 1 UDP 1694302207 199.87.82.66 37652 typ srflx raddr 10.0.1.46 rport 55401
a=candidate:0 2 UDP 2113667326 10.0.1.46 61947 typ host
a=candidate:1 2 UDP 1694302206 199.87.82.66 46592 typ srflx raddr 10.0.1.46 rport 61947

ICE Agent - A peer trying to connect to another peer via ICE. I have also heard this term used to describe an arbitrary node in the Internet Connectivity Establishment Handshake.

The following definitions borrow from the IETF ICE Spec

ICE Peer - From the perspective of one of the agents in a session, its peer is the other agent. Specifically, from the perspective of the offerer, the peer is the answerer. From the perspective of the answerer, the peer is the offerer.

ICE Transport Address - The combination of an IP address, port and transport protocol (such as UDP or TCP) port.

ICE Candidate - A transport address that is a potential point of contact for receipt of media. An agent may have many candidate addresses – each consisting of an IP address and port number. Candidates might include:

• A transport address on a directly attached network interface (a “host” address)
• A translated transport address on the public side of a NAT (a “server reflexive” address)
• A transport address allocated from a TURN server (a “relayed” address)

Candidates have properties – their type (server reflexive, relayed or host), priority, foundation, and base.

ICE Component - A component is a piece of a media stream requiring a single transport address; a media stream may require multiple components, each of which has to work for the media stream as a whole to work. For media streams based on RTP, there are two components per media stream – one for RTP, and one for RTCP.

Helpful WebRTC Resources

Mozilla Hacks blog post mentions some of the discrepancies between chrome and Firefox
Webrtc on Chrome for Beginners
General Description on IETF
WebRTC Spec Draft from the W3C
Peer.js – This WebRTC abstraction library worked really well for me