WebRTC Signaling in FS: SIP and Verto

As we've already seen in this chapter, WebRTC needs a session protocol to complement its own streaming capabilities. That protocol will use signaling to find the peer location, establish and tear down sessions, manage presence and messaging, etc. FreeSWITCH supports two such protocols: SIP and Verto.

These two different protocols only affect how is written the client application (client application is the web page and Javascript the browser load in order to access FreeSWITCH) and how it "talks" to FreeSWITCH. Both protocols give full access to FreeSWITCH features and services. Also, both signaling protocols can be used at the same time by different clients accessing the same service, and even interacting with each other (eg, we can have a videoconference in FreeSWITCH that's concurrently accessed by one WebRTC client "speaking" SIP and another "talking" Verto, in addition to other clients coming from TDM and "normal" (non-WebRTC) SIP: