Ironically, connections that are sending bulk data (eg. downloads) don't run into this, because a lot of packets get sent at once; if one of them in the middle gets dropped, the recipient knows it right away (because the next one has the wrong sequence number) so it can immediately send an ACK that re-requests the missing one. So on a busy connection, packet loss isn't as big a problem. On an interactive connection like ssh, it's completely deadly and makes it almost unusable.
All that to say: your problem with Wimax is fixable. It's not a latency problem, but nobody has fixed it yet.
Why is this bad? Because it screws with TCP window adjustments. Remember that your OS will send packets until it starts to see drops, then back off of the transmit rate. You can adjust the parameters (startup rate, backoff rate), but that's the basic principle. However, when there is an overly large buffer somewhere in the middle of the network, this delays drops. This has the effect of making it nearly impossible for your TCP stack to determine the true bandwidth.
For some reason, most mobile device manufacturers are the worst offenders when it comes to bufferbloat.
Anyways, this was a brief summary. For a much better explanation (including pretty graphs), see http://www.bufferbloat.net/