UDP yes. Repeated frames introduce latency. Broadcasters are used to latency measured in lines -- that's microseconds. In the analog days an extra foot on your cable would mean the difference between out of sync or not in the chroma domain.

We use SDI rather than HDMI, but fundamentally it's a container, with an active picture frame in 4K video of 2160 lines, with 3840 samples per line. Each sample is a 10 bit sample of the luma, and every other sample gets 2 extra 10 bit samples of chroma (The eye being less sensitive to chroma). This is 4:2:2 sampling.

This means that a single 4K frame is 2160x3840x20 bits, and in the americas there are about 60 lots of 165mbit, every second. SDI adds various things like blanking thanks to it's legacy of a digitised version of PAL/NTSC. Audio is tiny, <15mbit/sec.

Normal HD, with interlacing, is far more reasonable, at 1080x1920x20x30ish, 1.2gbit (less in 25fps countries), but SDI adds another 300mbit a second on top of that (UDP/RTP also adds overhead too).

The cheapest SDI routers you can get tend to be blackmagic ones, which are fine. There's a lot of people in the industry that remember the days of analog routers, where they were legitimatly expensive. Since SDI came out thought the electronics is little more than a switch chip. A 40x40 matrix will cost about $1300, but prices increase exponentially.

On wan links we tend to use compressed video in the 5-30mbit range (5mbit from somewhere skanky with poor internet). We have forward error correction on that, but the standards aren't great at high bit rates (only coping with 25 lost packets). A 20ms outage on a line with that type of error correction causes on air disruption.

Using something like ARQ which allows missing frames to be repeated adds latency, on a link from Europe to Singapore for example it could add 600ms. That's on top of the c. 100ms propagation latency and 400ms encode/decode, pushing the latency from half a second to over a second, and people start getting grumpy. There's still no guarentees either - as if you were to lose 100ms of traffic at 30mbit, you end up peaking your traffic far higher than the 30kbit/ms that you normally send -- to retransmit that 100ms in 100ms, you need to peak at 60mbit.

As an example of how important latency is for two-way communication, I'm currently investigating replacing our 12 frame (480ms) latency encoders from a couple of locations in London with ones that run at about 1-2 frames (40-80ms).

With wan links sending the same packet on two links doesn't even work. I've seen outages of 20ms on both a private wire and two separate internet paths in the past -- presumably some common router in the subsea cable system does something. Currently using temporal shifting (and some receive buffering) as well.

The expectations that the broadcast industry place is somewhat different to the IT industry. 3 second end-to-end latency is nearer 'store and forward' than 'low latency'. Compression in a production facility is unacceptable, and increasingly we're moving away from things like 5:1 compression on wan links and bandwidth becomes cheaper. One broadcaster in the UK has just replaced the main in-country network with an uncompressed network based on multi-path 100gbit wavelengths running over dark fibre. Another has replaced it's core switching network with multi-terabit backplanes. Just getting 10gbit/second in/out through the kernel is a challange - doing it with nanosecond packet timing that the next SMPTE standard currently requires is proving even harder.