Juha Saarinen dropped me a note a week or two back, asking for an update to my last post, in the wake of the IANA IP address pool finally running out and the recently announced successful bid for Nortel Networks’ IP address space by Microsoft for inclusion in NZCS Newsline.
The published article can be found here, and is different enough from the previous version to warrant re-posting.
I hate IPv4 link broadcast interface (e.g. Ethernet) addressing semantics. To recap, if I have two boxes on each end of a point-to-point link (say between a gateway and an end host), we address as follows (for example):
That’s four IP addresses, for a link to a single host. Hello? Haven’t you heard the news? IP addresses are running out!
Some folks manage to get away with using /31 masks, e.g.
which is just wrong. Better in terms of address usage (two addresses instead of four), but still just plain wrong. An you’re still wasting addresses.
The PPP folks a long time ago figured that a session, particularly in client to concentrator type configurations, only needs one IP address. A “point to point” interface has a local address, and a remote address, of which only the remote address needs to be stuffed in the routing table. The local address can be the address of the concentrator, and doesn’t even need to be in the same subnet.
So why can’t my Ethernet interfaces work the same way?
A point to point link really doesn’t have broadcast semantics. Apart from stuff like DHCP, you never really need to broadcast — after all, our PPP friends don’t see a need for a “broadcast” address.
Well, we decided we had to do something about this. The weapon of choice is NetGraph on FreeBSD. NetGraph basically provides a bunch of kernel modules that can be linked together. It’s been described as “network Lego”. I like it because it’s easy to slip new kernel modules into the network stack in a surprising number of places. This isn’t a NetGraph post, so I won’t spend more verbiage on it,but it’s way cool. Google it.
In a real point-to-point interface, both ends of the link know the semantics of the link. For Ethernet point-to-point addressing, we can still do this (and my code happily supports this configuration), but obviously both ends have to agree to do so. “Normal” clients won’t know what we’re up to, so we have to do this in such a way that we don’t upset their assumptions.
So we cheat. And we lie. And worst of all,we do proxy ARP!
What we do is tell our clients that they are on a /24 network. Their IP address is, for example, 10.1.2.5/24, and the gateway is 10.1.2.1. Any time we get a packet for 10.1.2.5, we’ll send it out that interface, doing ARP as normal to resolve the remote host’s MAC address.
Going the other way, we answer ARP requests for any IP address in 10.1.2.0/24, except 10.1.2.5, with our own MAC address. That means that if they ARP for 10.1.2.6, we’ll answer the ARP request, which directs that packet to us, where we can use our interior routes to route it correctly. In our world, two “adjacent” IP addresses could be on opposite sides of the network, or it could be on a different VLAN on the same interface.
The result is one IP address per customer. We “waste” three addresses per 256, the network (.0), gateway (.1) and broadcast (.255), and we have to be a bit careful about what we do with the .1 address — it could appear on every router that is playing with that /24. But we can give a user a single IP address, and put it anywhere in the network.
We can actually have multiple IP addresses on the same interface; we do this by having the NetGraph module have a single Ethernet interface but multiple virtual point-to-point interfaces. So if we want to give someone two IP addresses, we can do that as two, not necessarily adjacent, /32 addresses. We don’t answer ARPs for any of the assigned addresses, but do answer everything else. The module maintains a mapping of point-to-point interface to associated MAC address.
The following is a technique I’ve used over the last decade or so for distributing web traffic (or potentially any services) across multiple services, using just DNS. Being an old DNS hack, I’ve called this technique Poor Man’s Anycast, although it doesn’t really use anycasting.
But before we get into the technique, we need to make a brief diversion into the little-known but rather neat feature of the DNS, or more accurately, DNS forwarders, which makes this a cool way to do stuff. The feature is name server selection.
Most DNS clients, and by this I include your home PC, make use of a DNS forwarder. The forwarder is the thing that handles (and caches) DNS requests from end clients, while a DNS server carries authoritative information about a limited set of domains and only answers queries for them. These two functions have historically been conflated rather severely, mainly due to the use of BIND for both, and why this is a bad thing is the subject for a whole other post.
Moving right along. A DNS forwarder gets to handle lots of queries for any domain that its clients ask for. When you ask for foo.example.net, it asks one of the root servers (a.root-servers.net, b.root-servers.net et al) for that full domain name (let’s assume it’s just come up and doesn’t have anything cached). It gets back a delegation from the root servers, saying basically, “I don’t know, but the GTLD (.com, .net) servers will”, and tell you where to find the GTLD servers (a.gtld-servers.net et al). You ask the one of the GTLD servers, and get back an answer that says that they don’t know either, but ns1.example.net and ns2.example.net do.
You then ask (say) ns1.example.net, and hopefully you’ll get the answer you want (e.g. the IP address).
Now, along the way, the forwarder has been caching everything it got. Every time it asks a name server for data, it stores the time it took to reply. That means that when looking up names in example.net, the forwarder has been collecting timing and reliability data which it uses to choose which name server to ask next time, as well as the answers it received. So if ns1.example.net answers in 20 ms, but ns2.example.net answers in 10 ms, roughly two thirds of the queries for something.example.net will be sent to ns2.example.net. If the timing difference is much greater, the split of queries will be even more marked. Similarly, if a name server fails to respond at all, that fact will be reflected in the accumulated preference assigned to that server, and it will get very few queries in future; just enough so that we know we can start sending it queries again when it comes back.
This is a powerful effect, and is of particular use when distributing servers over a wide geographical area. DNS specialists know about it, because poor DNS performance affects everything, and DNS people don’t like adversely affecting everything. (They’re really quite paranoid about it. Trust me, I’m one.) But it can also be used to pick the closest server for other things as well.
After all, closeness (in terms of round-trip time) is very important in network performance (see my post on bandwidth delay products).
The technique is as follows. Let’s say we have three web servers, carrying static content. Call them, say, auckland.example.net, chicago.example.net and london.example.net. Let’s say that they’re widely disparate. All three servers carry content for http://www.example.com/.
So, we start by configuring, on the example.com name servers:
$ORIGIN example.com.
$TTL 86400
www IN NS auckland.example.net.
IN NS chicago.example.net.
IN NS london.example.net.
We then run a DNS server on all three web servers. We configure the servers with a zone for www.example.com along the lines of:
$ORIGIN www.example.com
$TTL 86400 ; Long (24 hour) TTL on NS records etc
@ IN SOA auckland.example.net. webmaster.example.com (
2009112900 3600 900 3600000 300 )
IN NS auckland.example.net.
IN NS chicago.example.net.
IN NS london.example.net.
$TTL 300 ; Short (five minute) TTL on A record
@ IN A 10.0.0.1 ; Set this to host IP address
Now the key is that each web server serves up its own IP address. When a DNS forwarder makes a query for a www.example.com, it will be directed to one of auckland.example.net, chicago.example.net or london.example.net. But as more and more queries get made, one of those three will start handling the bulk of the queries, at least if that one is significantly closer than the other two. And if auckland.example.net gets the query, it answers with its own IP address, meaning that it also gets the subsequent HTTP request or other services directed to it. The short DNS TTL (5 minutes in the example) mean that the address gets queried moderately often, allowing the name server selection to “get up to speed”. Much longer TTLs on the name servers mean the data doesn’t get forgotten too quickly.
The result is that in many cases, the best server gets the request.
The technique works best if there are lots of domains being handled by the same set of servers, and there are lots of requests coming through. That way the preferences get set quickly in the major ISPs’ DNS forwarders. The down side of the technique is that far away servers will still get some queries. This non-determinism may be a reason for not deploying this technique. If you want determinism, you’ll need to look at more industrial grade techniques.
Now, this isn’t what players like Akamai do, and it isn’t what anycasting is about. Akamai and (some) other content distribution networks work by maintaining a map of the Internet, and returning DNS answers based on the requester’s IP address. But this is a fairly heavyweight answer to the problem. It’s not something you can implement with just BIND alone.
Anycasting on the other hand relies on advertising the same IP address in multiple places, and letting BGP sort out the nearest path. This has three disadvantages:
The other nice thing about poor man’s anycast is that it’s dynamic; if a node falls off the world, as long as its DNS goes away too, it’ll just disappear from the cloud as soon as the TTLs time out. If a path to it gets congested, name server selection will notice the increased TTL and de-prefer that server.
And of course you don’t need to be a DNS or BGP guru, or buy/build expensive, complex software systems to set it up.
I found myself explaining this one at Curry tonight, in the context of discussing fast broadband.
Basically, if you have a reliable stream protocol like, to take a random example, TCP, and you’re not doing anything imaginative with it, you run into the following problem:
Every byte you send might need to be resent if it gets lost along the way. So, you buffer whatever you send up until you get an acknowledgement from the other end. Let’s say, for argument’s sake you use a 64k buffer. We call this buffer the window, and the size is the window size.
Now, let’s say you have a looooonnnngggg path between you and your remote endpoint. Let’s say it’s 200 milliseconds, or 1/5th of a second. This is pretty reasonable for an NZ-US connection — the speed of light is not our friend.
And finally, for simplicity sake, let’s say that the actual bandwidth over that path is Very High, so serialisation delays (the time taken to put one bit after the next) are negligible.
So, if I send 64k bytes (or 512 k bits) worth of data, it takes 200 ms before I get an acknowledgement. It doesn’t matter how fast I send my 64k; I still have to stop when I’ve sent it. 200 ms later, I get a bunch of acknowledgements back, for the whole 64k (assuming nothing got dropped), and I can now send my next 64k.
So the actual throughput, through my SuperDuperBroadband connection, is 64k bytes per 200 ms, or 2.5 Mbps.
To turn this around, if I want 2.5 Mbps at 200 ms, I need a 64k byte window; if I want 5 Mbps on a 200 ms path, I’m going to need to up the window size to 5 Mbps times 200 ms = 128 k bytes.
That window size calculation is the bandwidth delay product.
There’s ‘s the theory. Pick a big window size and go fast. Except:
What to do? It depends what you’re trying to achieve. For file transfers, run lots of TCP sessions side by side – can anyone say BitTorrent? Local caching helps for web traffic; move the content closer, and the bandwidth delay product is less. Use a different protocol. I have to say I’ve seen quite a few UDP-based file transfer protocols come and go, because tweaking TCP parameters at both ends is usually a darn sight easier than getting a new protocol right. (see don’s law).
What it comes down to, is that if all you’re going to use your UltraSuperDuperFast broadband connection for is downloading videos from US servers, you’re going to be disappointed. The real key to making this useful is local, or at least, locally hosted, content. Preferably located right by the fibre head-ends. It’s a parallel stream to the effort to get the fibre in the ground and get it lit, and it needs to be attended to PDQ.