WebSockets and WebRTC are two very different protocols and have very different use cases.
WebSockets is an extension to a HTTP request that permits a bi-directional, persistent link between a client and server.
You can exchange chunked messages, reliably and in-order using TCP. Once the WebSocket is established, you have free reign over what datastream is sent.
WebRTC is an extension of RTP (a UDP based protocol) that adds a layer of safety (to both users and to network infrastructure).
The key requirement for "real-time" communications mandates that reliable and in-order delivery must be sacrificed in favour of getting low-latency data transfers.
Another benefit of WebRTC is that browser-side javascript has no control over the exact datagram being sent over the wire (called RTCWeb, all WebRTC communication is encrypted between peers using SRTP). This eliminates a whole class of UDP based attacks. The client (or intermediate proxies) must not be able to send a specially crafted UDP packet to arbitrary hosts and trigger a distributed reflection attack.
Having said that, WebRTC can be used to transport arbitrary data, you just can't send an arbitrary packet.
The terms IaaS, PaaS and SaaS are defined under the "Service Models" section.