The Ingate WebRTC solution is extensive thus any attempt to visualise it gets complex.
The wrtc_kit is a software module that allows apps and webpages to get webRTC capabilities. In cooperation with the Ingate unit webRTC clients can call each other and SIP phones.
The Ingate unit has many components used by the webRTC solution. Below block diagram tries to show the major ones:
Firewall The Ingate unit has a powerful, SIP-aware firewall that protects it and all units behind it from the Internet.
QTURN A TURN server with call Quality control ensures media streams have priority over data streams.
SMS Bridge client Allows SIP and webRTC clients send and receive SMS text messages using telephony provider's SIP-to-SMS bridge.
SIP Trunks Enables SIP PBX-es to connect to telephony providers.
SIP PBX A complete switchboard with extensions etc.
SIP Presence A presence server to inform about availability, busy, in a meeting, away from phone etc.
SIP UA SIP User Agents that act as a bridge between webrtc and SIP clients.
SIP Registrar Keeping record of all logged in SIP users.
Web socket server Communicates with webRTC clients.
Web server Can be used to deliver webRTC client pages during development.
Voice bridge Used to connect less capable SIP clients that do not support media encryption or only supports one media stream.
SIP Voice services Advanced services like voice mail, auto attendant, telephone conference meeting rooms, etc.
There are much more features in the Ingate unit, these are just the major ones we would like to explain more about.