What are these “built in listeners” that you speak of, and which OS’s can I find this on?

Also TCP will work before UDP. For whatever reason firewalls and such do weird things with UDP traffic.

This doesn’t really explain why troubleshooting is more difficult, but rather that firewalls often cause trouble for UDP. My question was “why would diagnosing a UDP problem be harder than anything else?”, not “why is UDP more problematic?”. I’m fully aware that some firewalls are configured with absurd parameters, like blocking ICMP, QUIC, or even TLS in some bizarre situations. But my tools for debugging network troubles have always been the same: tcpdump for passive tracing, nc to coerce an active response, and icmp/icmp6 ping for basic connectivity and MTU verification.

Whether it’s UDP or anything other protocol, those tools have proven sufficient for me. Do you use something else?

I’ll add some color to this post.

The original FTP is, frankly, a monster of a protocol. Very useful, but an anomalous protocol that even without NAT breaking the end-to-end principle, it is unlikely to have survived modern corporate firewall rulesets in any case. The fact that FTP was even ported for TCP is its own historical quirk.

BTW, that is the term which this post is missing: the end-to-end principle is the design philosophy that the network itself should not have to perform work on payloads transmitted, except to carry it towards the destination. This also implies that no fields or bits should be modified in transit, once it leaves the sender; encap/decap restores the fields so that the receiver is none the wiser.

NAT breaks end-to-end in two respects: corruption of the original sender, and corruption of L4 port numbers. Note that Legacy IP also violated end-to-end, when packets are fragmented due to MTU issues. With IPv6, fragmentation by the network is disallowed outright, and the technical case for NAT is non-existent.

There’s something to be said about adopters of NAT, that they were facing a Hobson’s choice: use NAT so that they could connect early dial-up users to the emerging Internet, or reinvent the 20 year experiment of the Internet so that it could scale properly. In 1995, IPv6 did not exist so they made the only real choice available.

In the year 2026 though, that argument doesn’t hold water: it is a choice to continue to ignore the dual-stack and IPv6-only internet. There’s a quote that all sufficiently complex technical problems are in-fact political, and this is that: the modern case for Legacy IP and NAT is rooted in inertia, resource scarcity (eg hyperscale cloud companies buying up IP subnets), and recalcitrance by professionals that abdicate their responsibility to their clients to pursue the available technology.

There is no colorable technical rationale for why IPv6 best practices cannot be adopted today for most organizations, when all network hardware, all major consumer and enterprise OS’s, and all mobile phones support v6. Note that I said “best practices”, because a minority of orgs such as certain American ISPs have undertaken truly bonkers decisions that are putting us on track for the very same sins as Legacy IP.

The fact that some ISPs assign nothing but a single /128 via DHCP6 is absurd: this malpractice perpetuates the same problem as NAT44, except that there’s no good excuse for it. Even the most delusional of ISPs will never run out of /128 addresses in their assigned /32. Per best practice, even handing out /48’s to customers is not a problem either, because that’s 65536 customers and if that’s really a problem, just ask the RIR for another /32, which they can do as RIR dues-paying members. There is no practical limit, except that some people just cannot math properly to see that there’s no practical limit. When a technical solution to a technical problem fails because of innumeracy by those tasked with implementing it, then that’s so much worse than any 1990s workaround.

I’ve harped a lot about IPv6 because its strength today is that it’s technically competent, future-proofed, and most importantly, practitioners that saw the first travesty of NAT will not allow a redux to play out with IPv6. The technical reasons to deploy NAT66 are non-existent: it is always a workaround for political issues.

I can (just barely) accept NPTv6 as a like subnet-for-subnet mapping that does not harm L4, but that’s still glossing over a political issue. IPv6 gave network engineers real choices in numbers and administering their networks, and while some will squander it, I will be encouraging people to not let that happen.

Why would UDP be any more difficult than TCP (or anything else) for troubleshooting? If nothing else, it should be easier because there are no stateful connections and every packet is “fire and forget”. Now, as a tunneling technology, WG might be more difficult than a simple client/server socket-driven application, but that just means some routing tables need to be inspected and you have to understand IP subnets. In any case, you’d still want to obtain packet traces from both ends of the WG tunnel.

The thing is, if you don’t wish to port-forward on Legacy IP nor open firewall ports for IPv6, then the only other option is to reach out to some sort of relay server. But whether your own VPS fulfils that role or you use Tailscale’s DERP/TURN relays, the same class of latency and bandwidth issues will still erupt. So you’d still have to debug them, although you presumably would have better visibility into diagnosing issues with your VPS relay.

Generally speaking, if Tailscale isn’t working for you, then all other solutions will increase in complexity, not decrease.

Answering the ultimate question, I operate and secure my SSH server, with port 22 open only for IPv6 inbound. There’s no NAT, so I don’t have to deal with debugging odd port mangling. The SSH server only allows key auth, and all password attempts are logged and blocked, because there can’t be any legitimate traffic like that. On this SSH server would be my mounts for backups.

IIRC, the USA Social Security Administration’s website operates somewhat like this, because they seem to still do batch processing (a la mainframes) during the evening hours. As a result, they prevent people from logging in so that they don’t get interstitial data about their public pension.

Was AI used to generate part or all of this tool?

Is there a link somewhere? Maybe my reader isn’t showing the link?

This looks amazing. That said, you may want to have a look at rules 7 and 8 for this community.

Contributors: Claude

Please kindly see Rule 8 of this community and its associated post.

Would cert-manager for Kubernetes be approximately what you’re looking for? In that particular case, it’s a service that runs in a K8s cluster, that can auto-request CSRs and send them to Lets Encrypt using one of the challenge types. It takes care of managing the returned certificates, such as renewing them before expiration and registering the private keys within the K8s key store.

I presume that you’re not already using K8s, so maybe that project will help you find the keywords to describe the sort of certificate manager that you want.

If you’re already using Debian on desktop, then Debian on a VPS makes sense from a maintenance perspective: you only have to track one set of security updates. When there’s suddenly a critical update for some vulnerability, you can update both at once because it’s the same upstream repos. Whereas with Ubuntu on your VPS, you’d have to be mindful to make sure to check back when the same vulnerability is patched there.

Do not underestimate the effort to upkeeping a server, and do try to unify your procedures so you’re spending less time doing the necessary things, and more time building up your self hosting stack.

I’ll offer a comparison not yet mentioned in the context of hosting software, but which should sound familiar in the social media context: moderation is hard at scale.

By every metric of scale, Codeberg is a competitor to the behemoth that is GitHub. And that means they have to contend with tough choices, such as who and what to host on their servers, no different than BlueSky, Facebook, and the individual instances of the Fediverse. And to be clear, Codeberg has always been limited to FOSS projects anyway, so nobody can genuinely argue that limits weren’t in the cards.

The distinction is that less alike Meta, and more alike with the Fediverse, Codeberg takes no hostages: people with obvious AI repos (or even marginal repos, given the possible ambiguity) are free to take their code and leave. Yes, they have to rebuild their CI pipelines, but that’s always required for any platform, including self hosting.

The fact is that Codeberg is a centralized entity that needs to (try to) make consistent policy. In this case, they have a concern about copyright and nothing anyone says can magically dematerialize that risk. If their new policy is not agreeable, then decentralized self hosting is probably the only way forward. But that was always an option with ForgeJo and federation.

And indeed, folks moving from GitHub or elsewhere are not exactly without fair warning that they might have to migrate again; we’ve not reached a utopia yet.

What is the absolute simplest way to accomplish this?

< gets on soap box>

The absolute simplest way is to use IPv6: on your Debian machine, assign multiple IPv6 addresses, one for each service you want to expose to your LAN. There’s no penalty with having a dozen v6 addresses.

Although you have those addresses, your machine generally uses just a single one for its own outbound and inbound traffic. For each service, you would edit their config so that they bind to a specific IPv6 address. Finally, you would configure an AAAA entry in DNS so that your chosen subdomain will point to the IPv6 address in question.

As an example, suppose you had three different web servers running Nginx, currently on port 80, 8081, and 42069. What you would change is the server config for each server instance, adidng the IPv6 address as part of the “listen” directive. Since each instance is now bound to a different IP address, nginx can now listen to the conventional port 80 and n’ary will the three collide. In other software, the configuration option you’re looking for is the “bind address”. By binding each app to its own IP, it will only respond if you send a request to that IP (or DNS name, which translates to an IP); this conveniently makes debugging really easy.

For a home network, you might have a single /64 IPv6 subnet. But that still means you literally have billions upon billions of addresses to use before you ever run out. And as you’ve recognized, using DNS names to identify services is a lot easier and intuitive than using port numbers. It also neatly avoids the need to memorize IPv6 addresses, because that’s never been necessary: we have DNS.

If your ISP won’t give you IPv6, you can still use locally-assigned private ranges, known as ULA, and this works because your services are contained to your LAN. Best practice is to randomly generate a subnet then use it.

< gets off soap box>

Is that something you’re looking for? As a practical matter, the effort to maintain multiple release branches and backports – compared to the norm of just maintaining a single main branch – is an outsized effort, available to only well-funded FOSS projects, usually by having an industry sponsor that makes it a priority.

I posit that the grand majority of selfhosted FOSS software, by project count, do not have back ported branches. And so production environments will want to set up an offline validation setup to evaluate update, prior to making the switch. Not convenient, sure, but not exactly insurmountable either.

At some point, some amount of responsibility for use of FOSS software must fall upon the users, or else the project is less about creating value by building up the commons for software, and more on the exploitation/abuse of the volunteers.

Downside: breaking changes are fairly common AFAIK.

By my extremely cursory reading, this stopped being a problem after the v2.0.0 release, when they adopted semantic versioning. According, breaking changes would only occur when the major version number is rolled. And version v3.0.0 was just released 5 days ago, whereas v2.0.0 was released in October 2025.

It may be down to taste, but I view 9 months as being acceptable, given the breadth of features offered and the folks willing to put effort into the project.

If a Nintendo Wii counts as retro, there’s at least one person hosting a website using one: https://infosec.exchange/@alexhaydock/114377262481451962

But that was (incredibly!) able to run a prebuilt version of NetBSD, which had drivers for the official Wii USB LAN adapter. If instead there’s a piece of hardware that can boot an OS but doesn’t have a network interface device, then SLIP could be useful. Here’s a vape hosting a website: https://bogdanthegeek.github.io/blog/projects/vapeserver/

And if still, said retro device has no concept of IP or networking, can it at least speak serial over a 300 baud modem?

I’ll start with an observation that SSIDs, LANs, and subnets are all exercises in logical grouping and isolation. An optimal setup would place similarly-situated clients in the same group, while isolating clients that have no business with each other.

To that end, I’ll start with how I allocate SSIDs, which expects VLAN-aware APs but nothing fancy like RADIUS authentication. I only have two SSIDs, with one being the “public” network where the PSK WiFi password is distributed to all visitors, and the “private” network which is only used for my own long-term devices. There would have been a third SSID if I had IoT devices in my home, but I later implemented a rule to not allow those. There are no servers that sit on the Wifi network, because of the availability of CAT6 to every space that matters.

At L2, the two SSIDs map to two VLANs, but there are also VLANs for: the DMZ, internal-only servers, and the IP phones. The rationale for the DMZ being its own VLAN is to concentrate security into L3 (discussed later) and the firewall. The rationale for the internal servers and IP phone VLANs is because those endpoints should keep working even if most everything else on the network is having a problem. There should be no instance where a broadcast storm prevents me from making a phone call or managing the APs.

Finally, at L3, my setup has one large IP subnet that is divided across the VLANs so that inter-VLAN routing (and firewalling) is necessary between most devices. You’d think this would cause the router to be a bottleneck, but it doesn’t seem too bad in practice, since traffic within a VLAN doesn’t hit the router at all, and traffic destined for the WAN already has to be routed anyway.

On some VLANs, I will run dual-stack Legacy+IPv6, but I’m in the process of decommissioning Legacy IP where it’s not needed. Remarkably, even the IP phones don’t need Legacy IP anymore and run just fine on IPv6 with SLAAC address assignment.

IMO, abandonware means software that is a dead-end upon its very release, with no hopes or plans for anyone to every build upon it. Abandonware is generally not extensible, follows no good design philosophy that would let someone else build it up, and embodies essentially nothing.

Even a 100-line throwaway Python script has more utility to someone when it is published on PasteBin or whatever. But something like a binary executable released with no source code, with no support, and with no intent by the developer to ever make anything more of it, that’s abandonware.

some ominous comments stating that it is practically unmaintained (which is not true)

Objectively, I can see that the last commit to the default branch was in March 2026, and that the 10th newest commit was back in September 2025. Of these 10, 3 are new features and 6 are fixes and 1 is documentation. I also see in the issue tracker that no project developer replied to the two newest reports, which were reported 2 weeks and 2 months ago.

As a subjective opinion, the explanation that Conduit is essentially rock-solid and this doesn’t need much upkeep or commits, that is just not credible. The Git history shows fixes and new features, but at a rate that averages just one commit per month. And some of those commits are literally one-line changes.

But let’s suppose that the maintainers are uninterested in small UI or quality-of-life features, and only make changes when it crosses their threshold for what is “important” enough. That’s a choice, sure, but let’s see if that holds water. Here is the project’s response to an issue opened in January, with the response being in February that confirms a logic bug and schedules it for the next release.

That was three months ago. No updates. No mentioned branches or PRs or merges. All while this bug remains in place. And that’s understandable for FOSS project developers, for whom the project is not their day job.

But in any circumstances, the totality of the evidence does not inspire confidence, let alone a determination that Conduit is “rock solid”. And that’s even before looking at the code.

TL;DR: the premise of the question is wrong. Conduit is not maintained.

Given that your original problem was related to WAN upload performance, why did your investigation lead you to Ethernet flow-control? An ISP connection generally deals in packets at Layer 3 (“network”, eg IP) of the OSI model, whereas Ethernet is a Layer 2 (“data-link”) layer technology.

If there is a bottleneck at your WAN modem, then that will cause congestion at layer 3, but Ethernet flow-control can only deal with congestion that exists at layer 2. What has likely happened is that you have configured your gateway so that congestion at layer 3 is mirrored onto your layer 2 LAN. And if flow-control is enabled, then that would result in back-pressure propagating back to your VMs. Your VMs will then slow down their layer 2 rate, which conveniently forces the layer 3 traffic to also slow down.

This is an incredibly round-about and inefficient way to do traffic shaping. You should not configure a network so that L3 and L2 issues bleed into each other. A major consequence of using flow-control in this way is that it reduces the capacity of your LAN, even for traffic that isn’t going out to the WAN.

The customary approach for keeping L2 and L3 separate is to perform traffic shaping solely at the threshold where your LAN meets the bottleneck. This would be OpenWRT, since after OpenWRT would be the WAN (50 Mbps upload). OpenWRT would be configured with some sort of QoS feature so that certain L3 packets are selectively dropped.

You cannot do effective L3 traffic shaping without dropping packets. In fact, all competent L3 protocols expect dropped packets in order to slow down their data rate: SCTP and TCP have their own exponential congestion control mechanism, UDP simply accepts that some packets won’t make it through, and QUIC has its own mechanism as well. Simply put, all L3 protocols only understand one signal that tells them to slow down, and it is to drop a few packets. They will adjust accordingly, finding the stable equilibrium where traffic flows at the very cusp of congestion.

The Main reason for this problem seems to be the down-stepping of 10Gbit traffic to 1Gbit devices

This is a red-herring, for the reasons I’ve outlined above. With 1+ Gbps connections on your LAN, your L2 network is an order of magnitude faster than your WAN upload. It cannot be the case that a fast LAN makes a slow WAN slower. This is not RF impedance where step-transitions cause reflections; we are dealing in packet-switched networks, where queuing theory controls.

TL;DR: please try OpenWRT QoS instead