Combine `gstreamer-receive` and `gstreamer-send-offer` so people can
easily see how to transport audio/video without using their browser.
This moves the HTTP endpoint out of `examples/sfu` so it can be used in
multiple examples
Resolves#472
Let's pull off the bandaid!
* Reduces studdering: webrtc.RTCTrack -> webrtc.Track
* Makes it easier to find types by editor autocomplete
* Makes code read more fluently (less repetition)
Since we're breaking the API in 2.0, our only chance to
do this is now.
Relates to #408
This deviates from the WebRTC spec, so we need to fix it. This is a
massively breaking change, so we need to figure out the best way to help
users with this.
I also renamed our RTCPeerConnection constructor. The hope is that
people will refer to the examples/backlog and see what changed.
Resolves#309
