Summary

In this chapter, we learned how to make a dynamic web application for the WebRTC client using primitive building blocks such as CSS, JavaScript, SIP library, and HTML form elements. We also saw the setup of various kinds of SIP servers and their applicability in establishing an end-to-end call. In this process, we studied the implementation of WebSocket-supported SIP servers. We also studied the integration of the SIP WebRTC client with non-WebSocket supported SIP servers, through WebSocket gateways.

In essence, we learned about how client development and essential servers help to support the WebRTC SIP infrastructure. This includes the Tomcat web server, which caters to the loading of a web page and the HTTP handshake; the Kamailio SIP server, which acts as a registrar; and the SIP proxy node. The WebRTC client programs used open source libraries such as jsSIP and sipML5. The interaction and challenges inherent in communication between non-WebRTC sip endpoints, such as SIP phones and softphone, were also discussed.