Summary

In this chapter, we discussed the following:

• What is WebRTC and why it is very much relevant
• Mandatory security in WebRTC
• WebRTC implementation in FreeSWITCH
• The role of signaling protocols (SIP and Verto) in WebRTC
• Complete installation and configuration instruction for both FreeSWITCH and supporting WebRTC components (certificates, web servers)
• A super advanced and fully featured browser client for WebRTC videoconferencing and telepresence: Verto Communicator
• How to write a client web application using VERTO or SIP.js

In this chapter, we looked at both the theory and the nitty gritty details, and we learned how to setup real world WebRTC services for our user base.

In the next chapter we'll have a first look at the XML dialplan, the call routing engine at the core of FreeSWITCH. We'll look at how to implement the most diverse services and leverage the wealth of examples in the demo default configuration