Earlier quoted context omitted.
Not quite - getUserMedia is one part of WebRTC. If you use the 'full' WebRTC (i.e. RTCPeerConnection), you're tied into a certain architecture, which isn't generally useful if you already have a server architecture. getUserMedia is more useful in general for developing web conferencing solutions IMO. I think a lot of browser developers (Safari/IE) are baulking at implementing google's WebRTC. I would hope they would…
I don't see over-engineering in WebRTC; I see the same assisted-peer-to-peer routing required to support the major use-cases of traditional SIP: 1. When people are within the same firewall, they want to be able to communicate directly without routing anything but connection set-up through any servers; and 2. when people are members of the same corporation or mobile ISP, but are behind different firewalls, they want t…
In our product, our customers generally want the session recorded, which doesn't seem possible with WebRTC. Also we need to provide group video, which peer-to-peer isn't very useful for (in terms of efficiency).
That's not to say that WebRTC isn't useful in other use-cases (such as your product). However getUserMedia is more generic because it can be used in web conferencing products like ours. That is why I say that if browser developers don't want to implement the full WebRTC, they should at least implement getUserMedia.