pathping-carrier-nat-followup.md

📝 pathping-carrier-nat-followup.mdTheme:

C:\Windows\System32>pathping 128.199.200.202

Tracing route to forum.drugs-and-users.org [128.199.200.202]
over a maximum of 30 hops:
  0  Andrews-NUC [10.209.30.172]
  1  10.209.30.118
  2  192.0.0.1
  3     *        *        *
Computing statistics for 50 seconds...
            Source to Here   This Node/Link
Hop  RTT    Lost/Sent = Pct  Lost/Sent = Pct  Address
  0                                           Andrews-NUC [10.209.30.172]
                                0/ 100 =  0%   |
  1    3ms     0/ 100 =  0%     0/ 100 =  0%  10.209.30.118
                              100/ 100 =100%   |
  2  ---     100/ 100 =100%     0/ 100 =  0%  192.0.0.1

Trace complete.

Claude: [14/09/2026--19:02:29:838 AEST]

This pathping shows the same signature you've already nailed down for this link: hop 1 (your own router, 10.209.30.118) is clean — 3ms, 0% loss — but the trace dies at hop 2, the ISP's carrier-NAT gateway (192.0.0.1), which stops responding to further probes. That's exactly the behavior you confirmed before: the gateway answers as a hop but won't relay/respond past itself for diagnostic purposes, so pathping (and traceroute generally) can't see anything beyond it — including the actual loss/latency happening further out toward the Live server.

A couple of notes on reading this particular run:

The "100/100 = 100%" at hop 2 in the "Source to Here" column is a pathping artifact of the next hop not replying, not evidence hop 2 itself is dropping — its own "This Node/Link" line shows 0% loss for the 100 pings it received.
"Trace complete" after only reaching hop 2 confirms nothing beyond the carrier-NAT gateway responded at all this run — consistent with your earlier finding that the loss/latency isn't diagnosable or fixable from your end, since it's happening somewhere in the ISP's or upstream path you have no visibility or control over.

So this run doesn't add new information — it reproduces the known unfixable-from-here signature rather than pointing at anything new (e.g. it's not your router, and it's not something a config change on the Dev VM or Live server would touch). If you want to actually characterize the loss/latency beyond that gateway, the options are limited from a Windows box behind carrier-NAT: an ICMP-based path like this won't get past it. An MTR run from a box that isn't behind carrier-grade NAT, or asking the ISP directly, are really the only ways to see further — otherwise this confirms you're back to the mitigations already in place (split rsync streams, longer keepalive, SSH keepalive tuning) rather than a fixable root cause.

anything i can do about it ?

Claude: [14/09/2026--19:03:15:025 AEST]

Genuinely fixing the underlying RTT/loss itself isn't something you can do from your side — that's carrier-NAT/upstream routing you don't control. But there's a real split between "make the connection itself less bad" and "make your workflows tolerate it," and you've already done a lot of the second. Here's what's actually left on the table:

Things that might reduce the loss/latency itself (long shots, but worth ruling out):

Call your ISP and describe it plainly: high RTT (~370-670ms) and ~10-12% packet loss to a specific external IP, consistently reproducible. Ask if there's congestion or a fault on their carrier-NAT/upstream path. Most consumer ISPs won't do much with this, but it costs you a phone call and rules out something being actively wrong (vs. just being the nature of your connection).
If you're on 4G/5G fixed wireless or similar (carrier-NAT is a strong hint you're not on a plain wired connection), signal quality/congestion at your tower can directly cause this pattern. Worth checking if there's a different antenna placement, or whether your ISP has a "network status" page for your area.
Ask David (the friend hosting Live) whether his upstream has any packet loss reports from other directions — if it's one-sided (only bad toward you) that points more at your ISP; if it's bad from multiple sources it points at his host/network.

Things that make the workflows tolerate the loss (this is where the real leverage is, and you're already doing most of it):

You've already got the split-rsync-by-subdirectory mitigation and SSH keepalives tuned — that's the main lever for large transfers.
For rsync specifically, --partial (resume interrupted transfers) if not already on, and considering a lower --bwlimit sometimes paradoxically helps on lossy links by avoiding the buffer-bloat-induced retransmit spiral — worth a quick A/B test.
WireGuard or another UDP-based tunnel between Dev and Live can sometimes outperform raw TCP/SSH on lossy high-RTT links, since it doesn't suffer the same TCP slow-start/backoff penalty per retransmit. This is a bigger lift (new tunnel setup) but is the one architectural change that could meaningfully help rather than just work around the symptom.

Given you've called this "unfixable from his end" already and diagnosable, I'd frame it as: not worth more diagnostic time, but the WireGuard idea is the one thing that could change the actual transfer behavior rather than just paper over it — worth flagging as a wishlist item rather than doing right now, unless the recurrence rate is bad enough to justify the setup work.

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