When learning WebRTC developers often feel frustrated by the complexity. They see WebRTC features irrelevant to their current project and wish WebRTC was simpler. The issue is that everyone has a different set of use cases. Real-time communications has a rich history with lots of different people building many different things.
This chapter contains interviews with the authors of the protocols that make up WebRTC. It gives insight into the designs made when building each protocol, and finishes with an interview about WebRTC itself. If you understand the intentions and designs of the software you can build more effective systems with it.
RTP and RTCP is the protocol that handles all media transport for WebRTC. It was defined in RFC 1889 in January 1996. We are very lucky to have one of the authors Ron Frederick talk about it himself. Ron recently uploaded Network Video tool to GitHub, a project that informed RTP.
In October of 1992, I began to experiment with the Sun VideoPix frame grabber card, with the idea of writing a network videoconferencing tool based upon IP multicast. It was modeled after “vat” – an audioconferencing tool developed at LBL, in that it used a similar lightweight session protocol for users joining into conferences, where you simply sent data to a particular multicast group and watched that group for any traffic from other group members.