Makes it boolean. It's connected, or it's not. "One of RTP media transports to one of destinations is failing to establish DTLS ciphering and I think it has to do with either RTC issue or Chrome bug" is a self inflicted pain.
UDP is unreliable transport by specification, so I guess that if a network equipment such as a router cannot cope with the general workload, it would probably sacrifice UDP first without a second thought.
This is not how congestion control works on the internet.
Indeed TCP depends on packets getting dropped as the feedback mechanism for knowing when to slow down.
It's important that packets are dropped fairly, as otherwise on a loaded network only the preferred protocol(s) would keep working and the others would get starved. You don't want DNS to stop working when a HTTP flow is running at capacity on your link for example.
Huh? It's an obvious thing to do. If you have to drop a packet because your queues are full, any engineer with an IQ over 50 will pick the victim from the UDP packets, because the sender expects it might happen, and also because it won't necessary cause a retransmission - e.g. an RTP packet.
Why is that the obvious choice? TCP can recover through retransmission, UDP can not. Sounds just as logical of a choice to prioritize UDP and allow TCP connections to have a slowdown rather than allow UDP connections to have data loss.
As I said, application programmers expect and accept that their UDP packets might be lost or duplicated. This is sort of part of the contract. Even datagram integrity is in theory not guaranteed, as the checksum field of UDP is optional.
Sometimes people don't see a point at first in UDP because you eventually have to implement sequence numbers, CRCs, time-outs, retries, etc. that are similar to what TCP does. One can finds the reasons why one wants to do this anyway in [1]. In a nutshell, reliability is often insured by the application layer anyway so you don't need the transport protocol to do extra stuff you have no control over and might even get in the way (see the numerous esoteric ioctl and sysctl settings under Linux).
It is an obvious choice because, as I said again, a router dropping a packet does not necessary triggers a resend, e.g. RTP or syslog (over UDP). In TCP, this is guaranteed. If you are overloaded, you'd rather take the action you can get away with than probably just buy time.