The objective of this chapter is to make a simple WebRTC client and server module that bypasses a centralized server and, instead, makes a direct peer-to-peer connection between browsers through a Session Initiation Protocol (SIP) proxy server. The aim is to connect a WebRTC client to another WebRTC client using SIP over WebSocket as the signaling protocol. In this chapter, we will study the following three prime ways of making SIP WebRTC calls:
WebRTC to WebRTC call through a public cloud-hosted, WebRTC-capable SIP server, such as SIP2SIP
WebRTC to WebRTC call through a locally hosted, WebRTC-capable SIP server, such as OfficeSIP
WebRTC call to SIP phone through a WebSocket gateway and SIP server, such as Kamailio
We will begin the chapter by describing a simple WebRTC client-server model.