Webrtc is a peer-to-peer technology that allows for voice and video communication in real-time. It is used by many popular websites, including Google Hangouts, Facebook Messenger, and Skype. We will take a look at how webrtc internals works under the hood.
Webrtc is built on a number of different technologies, including the Session Initiation Protocol (SIP), Real-time Transport Protocol (RTP), and Secure RTP (SRTP). SIP is responsible for initiating and terminating communication sessions. RTP is responsible for transporting media-data, such as audio and video. SRTP is responsible for encrypting it. Webrtc also uses the Datagram Congestion Control Protocol (DCCP) to manage congestion control.
The Webrtc architecture consists of three main components: the signaling server, the peer connection, and the data channel. The signaling server is responsible for managing signals between peers. The peer connection is responsible for connecting peers and sending/receiving Media-Data. The data channel is responsible for sending/receiving application data, such as text messages and file transfers. Webrtc uses a variety of codecs to encode Media Data, such as H.264, VPX, and Opus. Webrtc also uses the Session Description Protocol (SDP) to describe media sessions.
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