Some of it might be nostalgia, but some of it is definitely latency and jitter. Even into the 90s there was only so much latency on a phone call and most phone

Some of it might be nostalgia, but some of it is definitely latency and jitter. Even into the 90s there was only so much latency on a phone call and most phone lines tried their hardest to _act_ circuit-switched. The average video caller is on WiFi, so there's packet buffering at the AP (which can be quite large since most people are running meh quality consumer routers at home), again at the router, then however many hops until the RTP endpoint. Jitter can be quite large. That doesn't even get into the scourge that is modern NAT and the possibility of TURN relays along the way. One quick "hack" is to do your conferencing on a box connected to your router via Ethernet (or through switches). It can help bad connection paths but it's not a panacea. Another is if you have a slow internet connection, try to get a router with QoS and apply that to prioritize your VoIP packets. Unfortunately other than cellular standards, there's no good way way to easily get a low latency, low jitter connection as most packet switching is optimized around throughput. Personally we use Zoom at work and I know the Zoom edge has low jitter to me so if anyone needs to VC me at work, they should get "as good as it gets". For my personal VC needs I ended up using Matrix and hosting a TURN server in a DC that peers directly with my ISP. Folks in my area can call me on it through WebRTC and get latency on the 3-6ms range.