Microsoft did propose some extensions to the existing W3C WebRTC
Microsofts full proposal can be found at: http://html5labs.com/cu-rtc-web/cu-rtc-web.htm
According to their announce they’ve added significant changes to the existing proposal to be able to adopt bandwidth while streaming. But before any application would change bandwidth settings in a Real-Time communication you would need to find out if there are actual bandwidth issues. From what I understood their proposal does not say how to measure bandwidth issues. I guess from their point of view measuring bandwidth issues are not part of the API itself.
What makes me wonder is that they did not propose to set Buffer sizes for incoming and outgoing media traffic. From my point of view having the possibility to set buffer sizes is an essential functionality in streaming. Eventually it could be also used to measure bandwidth issues.
I also wonder when Adobe will start to add their proposal for Real-Time Communication. As a company with that history in Real-Time Communication it should be a quite valuable contribution.