We used to destroy and recreate down streams whenever something changed,
which turned out to be racy. We now properly implement renegotiation,
as well as atomic replacement of a stream by another one.
Split the id field into id and source, where source indicates the sender
of the message and id the entity being sent. Remove the label request,
just use the offerer's username. Maintain the username within the
ServerConnection, this removes a parameter from some methods.
It is now possible to specify codecs other than VP8 and Opus. This
turns out not to be very useful, since VP8 is the only codec supported
by all browsers (in violation of the WebRTC spec, which mandates support
for H.264), and there is no good reason to use anything other than Opus
for audio.
We already send NACKs when a packet is missing. Under high packet loss,
however, the recovery packet might get lost two. Forward receiver NACKs
to the sender, but only after a delay and after checking that the packet
has not arrived in the meantime.
