Latency is the mind killer, but if available bandwidth is low, you save a ton of overhead by bundling 2-5 of your 20ms samples. Enough that the codec savings start to make sense, even though 100ms packets adds a ton of latency. Fancier systems can adapt codecs and samples per packet based on current conditions. The one I work on is a static codec and 60 ms of audio per packet, which isn't ideal, but allows us to run in low bandwidth much better than 20 ms per packet.

Edit to add: Meta can also afford to add a bit more sampling delay, because they've got very wide distribution of forwarding servers (they can do forwarding in their content appliances embedded in many ISPs), which reduces network delay vs competing services that have limited ability to host forwarding around the globe. Peer to peer doesn't always work and isn't always lower delay than going through a nearby forwarding server.