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            When a WebRTC Agent starts it has no idea who it is going to communicate with and what they are going to communicate about. Signalling solves this issue! Signalling is used to bootstrap the call                  call so that two WebRTC agents can start communicating. Signaling uses an existing protocol SDP (Session Description Protocol). SDP is a plain-text protocol. Each SDP message is made up of                              of key/value pairs and contains a list of “media sections”.

•  Connecting:    

            The two WebRTC Agents now know enough details to attempt to connect to each other. WebRTC then uses another established technology called ICE.

            ICE (Interactive Connectivity Establishment) is a protocol that pre-dates WebRTC. ICE allows the establishment of a connection between two Agents. These Agents could be on the same network, or                on the other side of the world. ICE is the solution to establishing a direct connection without a central server.

            The real magic here is ‘NAT Traversal’ and STUN/TURN Servers. These two concepts are all you need to communicate with an ICE Agent in another subnet. Once ICE successfully connects, WebRTC                then moves on to establishing an encrypted transport. This transport is used for audio, video, and data.

• Securing:

           Now that we have bi-directional communication (via ICE) we need to establish secure communication. This is done through two protocols that pre-date WebRTC. The first protocol is DTLS (Datagram             Transport Layer Security) which is just TLS over UDP. TLS is the cryptographic protocol used to secure communication over HTTPS. The second protocol is SRTP (Secure Real-time Transport                             Protocol).

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