Chapter 3. WebRTC with SIP and IMS

IP Multimedia Subsystem (IMS) is an architectural framework for IP Multimedia communications and IP telephony based on Convergent applications. It specifies three layers in a telecom network:

• Transport or Access layer: This is the bottom-most segment responsible for interacting with end systems such as phones.
• IMS layer: This is the middleware responsible for authenticating and routing the traffic and facilitating call control through the Service layer.
• Service or Application layer: This is the top-most layer where all of the call control applications and Value Added Services (VAS) are hosted.

IMS standards are defined by Third Generation Partnership Project (3GPP) which adopt and promote Internet Engineering Task Force (IETF) Request for Comments (RFCs). Refer to http://www.3gpp.org/technologies/keywords-acronyms/109-ims to learn more about 3GPP IMS specification releases.

This chapter will walk us through the interaction of WebRTC client with important IMS nodes and modules. The WebRTC gateway is the first point of contact for the SIP requests from the WebRTC client to enter into the IMS network. The WebRTC gateway converts SIP over WebSocket implementation to legacy/plain SIP, that is, a WebRTC to SIP gateway that connects to the IMS world and is able to communicate with a legacy SIP environment. It also can translate other REST- or JSON-based signaling protocols into SIP. The gateway also handles the media operation that involves DTLS, SRTP, RTP, transcoding, demuxing, and so on.

In the previous chapter, we saw how to create the WebRTC environment using the SIP server that has WebSocket capabilities. In this chapter, we will study a case where there exists a simple IMS core environment, and the WebRTC clients are meant to interact after the signals are traversed through core IMS nodes such as Call Session Control Function (CSCF), Home Subscriber Server (HSS), and Telecom Application Server (TAS).