9.4: Overlay Networks
- Page ID
9.4 Overlay Networks
9.4 Overlay Networks
From its inception, the Internet has adopted a clean model, in which the routers inside the network are responsible for forwarding packets from source to destination, and application programs run on the hosts connected to the edges of the network. The client/server paradigm illustrated by the applications discussed in the first two sections of this chapter certainly adhere to this model.
In the last few years, however, the distinction between packet forwarding and application processing has become less clear. New applications are being distributed across the Internet, and in many cases these applications make their own forwarding decisions. These new hybrid applications can sometimes be implemented by extending traditional routers and switches to support a modest amount of application-specific processing. For example, so-called level-7 switches sit in front of server clusters and forward HTTP requests to a specific server based on the requested URL. However, overlay networks are quickly emerging as the mechanism of choice for introducing new functionality into the Internet.
You can think of an overlay as a logical network implemented on top of some underlying network. By this definition, the Internet started out as an overlay network on top of the links provided by the old telephone network. Figure 1 depicts an overlay implemented on top of an underlying network. Each node in the overlay also exists in the underlying network; it processes and forwards packets in an application-specific way. The links that connect the overlay nodes are implemented as tunnels through the underlying network. Multiple overlay networks can exist on top of the same underlying network—each implementing its own application-specific behavior—and overlays can be nested, one on top of another. For example, all of the example overlay networks discussed in this section treat today's Internet as the underlying network.
We have already seen examples of tunneling, for example, to implement
virtual private networks (VPNs). As a brief refresher, the nodes on
either end of a tunnel treat the multi-hop path between them as a single
logical link, the nodes that are tunneled through forward packets based
on the outer header, never aware that the end nodes have attached an
inner header. Figure 2 shows three overlay nodes (A, B, and
C) connected by a pair of tunnels. In this example, overlay node B might
make a forwarding decision for packets from A to C based on the inner
IHdr), and then attach an outer header (
identifies C as the destination in the underlying network. Nodes A, B,
and C are able to interpret both the inner and outer header, whereas the
intermediate routers understand only the outer header. Similarly, A, B,
and C have addresses in both the overlay network and the underlying
network, but they are not necessarily the same; for example, their
underlying address might be a 32-bit IP address, while their overlay
address might be an experimental 128-bit address. In fact, the overlay
need not use conventional addresses at all but may route based on URLs,
domain names, an XML query, or even the content of the packet.
The simplest kind of overlay is one that exists purely to support an alternative routing strategy; no additional application-level processing is performed at the overlay nodes. You can view a virtual private network (VPN) as an example of a routing overlay, but one that doesn't so much define an alternative strategy or algorithm as it does alternative routing table entries to be processed by the standard IP forwarding algorithm. In this particular case, the overlay is said to use "IP tunnels," and the ability to utilize these VPNs is supported in many commercial routers.
Suppose, however, you wanted to use a routing algorithm that commercial router vendors were not willing to include in their products. How would you go about doing it? You could simply run your algorithm on a collection of end hosts, and tunnel through the Internet routers. These hosts would behave like routers in the overlay network: As hosts they are probably connected to the Internet by only one physical link, but as a node in the overlay they would be connected to multiple neighbors via tunnels.
Since overlays, almost by definition, are a way to introduce new technologies independent of the standardization process, there are no standard overlays we can point to as examples. Instead, we illustrate the general idea of routing overlays by describing several experimental systems that have been built by network researchers.
Experimental Versions of IP
Overlays are ideal for deploying experimental versions of IP that you hope will eventually take over the world. For example, IP multicast started off as an extension to IP and even today is not enabled in many Internet routers. The MBone (multicast backbone) was an overlay network that implemented IP multicast on top of the unicast routing provided by the Internet. A number of multimedia conference tools were developed for and deployed on the Mbone. For example, IETF meetings—which are a week long and attract thousands of participants—were for many years broadcast over the MBone.
Like VPNs, the MBone used both IP tunnels and IP addresses, but unlike VPNs, the MBone implemented a different forwarding algorithm—forwarding packets to all downstream neighbors in the shortest path multicast tree. As an overlay, multicast-aware routers tunnel through legacy routers, with the hope that one day there will be no more legacy routers.
The 6-BONE was a similar overlay that was used to incrementally deploy IPv6. Like the MBone, the 6-BONE used tunnels to forward packets through IPv4 routers. Unlike the MBone, however, 6-BONE nodes did not simply provide a new interpretation of IPv4's 32-bit addresses. Instead, they forwarded packets based on IPv6's 128-bit address space. The 6-BONE also supported IPv6 multicast.
End System Multicast
Although IP multicast is popular with researchers and certain segments of the networking community, its deployment in the global Internet has been limited at best. In response, multicast-based applications like videoconferencing have recently turned to an alternative strategy, called end system multicast. The idea of end system multicast is to accept that IP multicast will never become ubiquitous and to instead let the end hosts that are participating in a particular multicast-based application implement their own multicast trees.
Before describing how end system multicast works, it is important to first understand that, unlike VPNs and the MBone, end system multicast assumes that only Internet hosts (as opposed to Internet routers) participate in the overlay. Moreover, these hosts typically exchange messages with each other through UDP tunnels rather than IP tunnels, making it easy to implement as regular application programs. This makes it possible to view the underlying network as a fully connected graph, since every host in the Internet is able to send a message to every other host. Abstractly, then, end system multicast solves the following problem: Starting with a fully connected graph representing the Internet, the goal is to find the embedded multicast tree that spans all the group members.
There is a simpler version of this problem, enabled by the ready availability of cloud-hosted VMs around the world. The multicast-aware "end systems" can be VMs running at multiple sites. As these sites are well-known and relatively fixed, it's possible to construct a static multicast tree in the cloud, and have the actual end-hosts simply connect to the nearest cloud location. But for the sake of completeness, the following describes the approach in its full glory.
Since we take the underlying Internet to be fully connected, a naive solution would be to have each source directly connected to each member of the group. In other words, end system multicast could be implemented by having each node send unicast messages to every group member. To see the problem in doing this, especially compared to implementing IP multicast in routers, consider the example topology in Figure 3. Figure 3(a) depicts an example physical topology, where R1 and R2 are routers connected by a low-bandwidth transcontinental link; A, B, C, and D are end hosts; and link delays are given as edge weights. Assuming A wants to send a multicast message to the other three hosts, Figure 3(b) shows how naive unicast transmission would work. This is clearly undesirable because the same message must traverse the link A-R1 three times, and two copies of the message traverse R1-R2. Figure 3(c) depicts the IP multicast tree constructed by the Distance Vector Multicast Routing Protocol (DVMRP). Clearly, this approach eliminates the redundant messages. Without support from the routers, however, the best one can hope for with end system multicast is a tree similar to the one shown in Figure 3(d). End system multicast defines an architecture for constructing this tree.
The general approach is to support multiple levels of overlay networks, each of which extracts a subgraph from the overlay below it, until we have selected the subgraph that the application expects. For end system multicast, in particular, this happens in two stages: First we construct a simple mesh overlay on top of the fully connected Internet, and then we select a multicast tree within this mesh. The idea is illustrated in Figure 4, again assuming the four end hosts A, B, C, and D. The first step is the critical one: Once we have selected a suitable mesh overlay, we simply run a standard multicast routing algorithm (e.g., DVMRP) on top of it to build the multicast tree. We also have the luxury of ignoring the scalability issue that Internet-wide multicast faces since the intermediate mesh can be selected to include only those nodes that want to participate in a particular multicast group.
The key to constructing the intermediate mesh overlay is to select a topology that roughly corresponds to the physical topology of the underlying Internet, but we have to do this without anyone telling us what the underlying Internet actually looks like since we are running only on end hosts and not routers. The general strategy is for the end hosts to measure the roundtrip latency to other nodes and decide to add links to the mesh only when they like what they see. This works as follows.
First, assuming a mesh already exists, each node exchanges the list of all other nodes it believes is part of the mesh with its directly connected neighbors. When a node receives such a membership list from a neighbor, it incorporates that information into its membership list and forwards the resulting list to its neighbors. This information eventually propagates through the mesh, much as in a distance vector routing protocol.
When a host wants to join the multicast overlay, it must know the IP address of at least one other node already in the overlay. It then sends a "join mesh" message to this node. This connects the new node to the mesh by an edge to the known node. In general, the new node might send a join message to multiple current nodes, thereby joining the mesh by multiple links. Once a node is connected to the mesh by a set of links, it periodically sends "keep alive" messages to its neighbors, letting them know that it still wants to be part of the group.
When a node leaves the group, it sends a "leave mesh" message to its directly connected neighbors, and this information is propagated to the other nodes in the mesh via the membership list described above. Alternatively, a node can fail or just silently decide to quit the group, in which case its neighbors detect that it is no longer sending "keep alive" messages. Some node departures have little effect on the mesh, but should a node detect that the mesh has become partitioned due to a departing node, it creates a new edge to a node in the other partition by sending it a "join mesh" message. Note that multiple neighbors can simultaneously decide that a partition has occurred in the mesh, leading to multiple cross-partition edges being added to the mesh.
As described so far, we will end up with a mesh that is a subgraph of the original fully connected Internet, but it may have suboptimal performance because (1) initial neighbor selection adds random links to the topology, (2) partition repair might add edges that are essential at the moment but not useful in the long run, (3) group membership may change due to dynamic joins and departures, and (4) underlying network conditions may change. What needs to happen is that the system must evaluate the value of each edge, resulting in new edges being added to the mesh and existing edges being removed over time.
To add new edges, each node periodically probes some random member that it is not currently connected to in the mesh, measures the round-trip latency of edge , and then evaluates the utility of adding this edge. If the utility is above a certain threshold, link is added to the mesh. Evaluating the utility of adding edge might look something like this:
EvaluateUtility(j) utility = 0 for each member m not equal to i CL = current latency to node m along route through mesh NL = new latency to node m along mesh if edge (i,j) is added} if (NL < CL) then utility += (CL - NL)/CL return utility
Deciding to remove an edge is similar, except each node computes the cost of each link to current neighbor as follows:
EvaluateCost(j) Cost[i,j] = number of members for which i uses j as next hop Cost[j,i] = number of members for which j uses i as next hop return max(Cost[i,j], Cost[j,i])
It then picks the neighbor with the lowest cost, and drops it if the cost falls below a certain threshold.
Finally, since the mesh is maintained using what is essentially a distance vector protocol, it is trivial to run DVMRP to find an appropriate multicast tree in the mesh. Note that, although it is not possible to prove that the protocol just described results in the optimum mesh network, thereby allowing DVMRP to select the best possible multicast tree, both simulation and extensive practical experience suggests that it does a good job.
Resilient Overlay Networks
Another function that can be performed by an overlay is to find alternative routes for traditional unicast applications. Such overlays exploit the observation that the triangle inequality does not hold in the Internet. Figure 5 illustrates what we mean by this. It is not uncommon to find three sites in the Internet—call them A, B, and C—such that the latency between A and B is greater than the sum of the latencies from A to C and from C to B. That is, sometimes you would be better off indirectly sending your packets via some intermediate node than sending them directly to the destination.
How can this be? Well, the Border Gateway Protocol (BGP) never promised that it would find the shortest route between any two sites; it only tries to find some route. To make matters more complex, BGP's routes are heavily influenced by policy issues, such as who is paying whom to carry their traffic. This often happens, for example, at peering points between major backbone ISPs. In short, that the triangle inequality does not hold in the Internet should not come as a surprise.
How do we exploit this observation? The first step is to realize that there is a fundamental tradeoff between the scalability and optimality of a routing algorithm. On the one hand, BGP scales to very large networks, but often does not select the best possible route and is slow to adapt to network outages. On the other hand, if you were only worried about finding the best route among a handful of sites, you could do a much better job of monitoring the quality of every path you might use, thereby allowing you to select the best possible route at any moment in time.
An experimental overlay, called the Resilient Overlay Network (RON), did exactly this. RON scaled to only a few dozen nodes because it used an strategy of closely monitoring (via active probes) three aspects of path quality—latency, available bandwidth, and loss probability—between every pair of sites. It was then able to both select the optimal route between any pair of nodes, and rapidly change routes should network conditions change. Experience showed that RON was able to deliver modest performance improvements to applications, but more importantly, it recovered from network failures much more quickly. For example, during one 64-hour period in 2001, an instance of RON running on 12 nodes detected 32 outages lasting over 30 minutes, and it was able to recover from all of them in less than 20 seconds on average. This experiment also suggested that forwarding data through just one intermediate node is usually sufficient to recover from Internet failures.
Since RON was not designed to be a scalable approach, it is not possible to use RON to help random host A communicate with random host B; A and B have to know ahead of time that they are likely to communicate and then join the same RON. However, RON seems like a good idea in certain settings, such as when connecting a few dozen corporate sites spread across the Internet or allowing you and 50 of your friends to establish your own private overlay for the sake of running some application. (Today, this idea is put to practice with the marketing name Software-Defined WAN, or SD-WAN.) The real question, though, is what happens when everyone starts to run their own RON. Does the overhead of millions of RONs aggressively probing paths swamp the network, and does anyone see improved behavior when many RONs compete for the same paths? These questions are still unanswered.
All of these overlays illustrate a concept that is central to computer networks in general: virtualization. That is, it is possible to build a virtual network from abstract (logical) resources on top of a physical network constructed from physical resources. Moreover, it is possible to stack these virtualized networks on top of each other and for multiple virtual network to coexist at the same level. Each virtual network, in turn, provides new capabilities that are of value to some set of users, applications, or higher-level networks.
Music-sharing applications like Napster and KaZaA introduced the term "peer-to-peer" into the popular vernacular. But what exactly does it mean for a system to be "peer-to-peer"? Certainly in the context of sharing MP3 files it means not having to download music from a central site, but instead being able to access music files directly from whoever in the Internet happens to have a copy stored on their computer. More generally then, we could say that a peer-to-peer network allows a community of users to pool their resources (content, storage, network bandwidth, disk bandwidth, CPU), thereby providing access to a larger archival store, larger video/audio conferences, more complex searches and computations, and so on than any one user could afford individually.
Quite often, attributes like decentralized and self-organizing are mentioned when discussing peer-to-peer networks, meaning that individual nodes organize themselves into a network without any centralized coordination. If you think about it, terms like these could be used to describe the Internet itself. Ironically, however, Napster was not a true peer-to-peer system by this definition since it depended on a central registry of known files, and users had to search this directory to find what machine offered a particular file. It was only the last step—actually downloading the file—that took place between machines that belong to two users, but this is little more than a traditional client/server transaction. The only difference is that the server is owned by someone just like you rather than a large corporation.
So we are back to the original question: What's interesting about peer-to-peer networks? One answer is that both the process of locating an object of interest and the process of downloading that object onto your local machine happen without your having to contact a centralized authority, and at the same time the system is able to scale to millions of nodes. A peer-to-peer system that can accomplish these two tasks in a decentralized manner turns out to be an overlay network, where the nodes are those hosts that are willing to share objects of interest (e.g., music and other assorted files), and the links (tunnels) connecting these nodes represent the sequence of machines that you have to visit to track down the object you want. This description will become clearer after we look at two examples.
Gnutella is an early peer-to-peer network that attempted to distinguish between exchanging music (which likely violates somebody's copyright) and the general sharing of files (which must be good since we've been taught to share since the age of two). What's interesting about Gnutella is that it was one of the first such systems to not depend on a centralized registry of objects. Instead, Gnutella participants arrange themselves into an overlay network similar to the one shown in Figure 6. That is, each node that runs the Gnutella software (i.e., implements the Gnutella protocol) knows about some set of other machines that also run the Gnutella software. The relationship "A and B know each other" corresponds to the edges in this graph. (We'll talk about how this graph is formed in a moment.)
Whenever the user on a given node wants to find an object, Gnutella sends a QUERY message for the object—for example, specifying the file's name—to its neighbors in the graph. If one of the neighbors has the object, it responds to the node that sent it the query with a QUERY RESPONSE message, specifying where the object can be downloaded (e.g., an IP address and TCP port number). That node can subsequently use GET or PUT messages to access the object. If the node cannot resolve the query, it forwards the QUERY message to each of its neighbors (except the one that sent it the query), and the process repeats. In other words, Gnutella floods the overlay to locate the desired object. Gnutella sets a TTL on each query so this flood does not continue indefinitely.
In addition to the TTL and query string, each QUERY message contains a unique query identifier (QID), but it does not contain the identity of the original message source. Instead, each node maintains a record of the QUERY messages it has seen recently: both the QID and the neighbor that sent it the QUERY. It uses this history in two ways. First, if it ever receives a QUERY with a QID that matches one it has seen recently, the node does not forward the QUERY message. This serves to cut off forwarding loops more quickly than the TTL might have done. Second, whenever the node receives a QUERY RESPONSE from a downstream neighbor, it knows to forward the response to the upstream neighbor that originally sent it the QUERY message. In this way, the response works its way back to the original node without any of the intermediate nodes knowing who wanted to locate this particular object in the first place.
Returning to the question of how the graph evolves, a node certainly has to know about at least one other node when it joins a Gnutella overlay. The new node is attached to the overlay by at least this one link. After that, a given node learns about other nodes as the result of QUERY RESPONSE messages, both for objects it requested and for responses that just happen to pass through it. A node is free to decide which of the nodes it discovers in this way that it wants to keep as a neighbor. The Gnutella protocol provides PING and PONG messages by which a node probes whether or not a given neighbor still exists and that neighbor's response, respectively.
It should be clear that Gnutella as described here is not a particularly clever protocol, and subsequent systems have tried to improve upon it. One dimension along which improvements are possible is in how queries are propagated. Flooding has the nice property that it is guaranteed to find the desired object in the fewest possible hops, but it does not scale well. It is possible to forward queries randomly, or according to the probability of success based on past results. A second dimension is to proactively replicate the objects, since the more copies of a given object there are, the easier it should be to find a copy. Alternatively, one could develop a completely different strategy, which is the topic we consider next.
At the same time file sharing systems started fighting to fill the void left by Napster, the research community began to explore an alternative design for peer-to-peer networks. We refer to these networks as structured, to contrast them with the essentially random (unstructured) way in which a Gnutella network evolves. Unstructured overlays like Gnutella employ trivial overlay construction and maintenance algorithms, but the best they can offer is unreliable, random search. In contrast, structured overlays are designed to conform to a particular graph structure that allows reliable and efficient (probabilistically bounded delay) object location, in return for additional complexity during overlay construction and maintenance.
If you think about what we are trying to do at a high level, there are two questions to consider: (1) How do we map objects onto nodes, and (2) How do we route a request to the node that is responsible for a given object? We start with the first question, which has a simple statement: How do we map an object with name into the address of some node that is able to serve that object? While traditional peer-to-peer networks have no control over which node hosts object , if we could control how objects get distributed over the network, we might be able to do a better job of finding those objects at a later time.
A well-known technique for mapping names into an address is to use a hash table, so that
implies object is first placed on node , and at a later time a client trying to locate would only have to perform the hash of to determine that it is on node . A hash-based approach has the nice property that it tends to spread the objects evenly across the set of nodes, but straightforward hashing algorithms suffer from a fatal flaw: How many possible values of should we allow? (In hashing terminology, how many buckets should there be?) Naively, we could decide that there are, say, 101 possible hash values, and we use a modulo hash function; that is,
hash(x) return x % 101
Unfortunately, if there are more than 101 nodes willing to host objects, then we can't take advantage of all of them. On the other hand, if we select a number larger than the largest possible number of nodes, then there will be some values of that will hash into an address for a node that does not exist. There is also the not-so-small issue of translating the value returned by the hash function into an actual IP address.
To address these issues, structured peer-to-peer networks use an algorithm known as consistent hashing, which hashes a set of objects uniformly across a large ID space. Figure 7 visualizes a 128-bit ID space as a circle, where we use the algorithm to place both objects
onto this circle. Since a 128-bit ID space is enormous, it is unlikely that an object will hash to exactly the same ID as a machine's IP address hashes to. To account for this unlikelihood, each object is maintained on the node whose ID is closest, in this 128-bit space, to the object ID. In other words, the idea is to use a high-quality hash function to map both nodes and objects into the same large, sparse ID space; you then map objects to nodes by numerical proximity of their respective identifiers. Like ordinary hashing, this distributes objects fairly evenly across nodes, but, unlike ordinary hashing, only a small number of objects have to move when a node (hash bucket) joins or leaves.
We now turn to the second question—how does a user that wants to access object know which node is closest in 's ID in this space? One possible answer is that each node keeps a complete table of node IDs and their associated IP addresses, but this would not be practical for a large network. The alternative, which is the approach used by structured peer-to-peer networks, is to route a message to this node! In other words, if we construct the overlay in a clever way—which is the same as saying that we need to choose entries for a node's routing table in a clever way—then we find a node simply by routing toward it. Collectively, this approach is sometimes called a distributed hash table (DHT), since conceptually, the hash table is distributed over all the nodes in the network.
Figure 8 illustrates what happens for a simple 28-bit ID space. To keep the discussion as concrete as possible, we consider the approach used by a particular peer-to-peer network called Pastry. Other systems work in a similar manner.
Suppose you are at the node with id
65a1fc (hex) and you are trying to
locate the object with ID
d46a1c. You realize that your ID shares
nothing with the object's, but you know of a node that shares at least
d. That node is closer than you in the 128-bit ID space,
so you forward the message to it. (We do not give the format of the
message being forwarded, but you can think of it as saying "locate
d46a1c.") Assuming node
d13da3 knows of another node that
shares an even longer prefix with the object, it forwards the message
on. This process of moving closer in ID-space continues until you reach
a node that knows of no closer node. This node is, by definition, the
one that hosts the object. Keep in mind that as we logically move
through "ID space" the message is actually being forwarded, node to
node, through the underlying Internet.
Each node maintains a both routing table (more below) and the IP addresses of a small set of numerically larger and smaller node IDs. This is called the node's leaf set. The relevance of the leaf set is that, once a message is routed to any node in the same leaf set as the node that hosts the object, that node can directly forward the message to the ultimate destination. Said another way, the leaf set facilitates correct and efficient delivery of a message to the numerically closest node, even though multiple nodes may exist that share a maximal length prefix with the object ID. Moreover, the leaf set makes routing more robust because any of the nodes in a leaf set can route a message just as well as any other node in the same set. Thus, if one node is unable to make progress routing a message, one of its neighbors in the leaf set may be able to. In summary, the routing procedure is defined as follows:
Route(D) if D is within range of my leaf set forward to numerically closest member in leaf set else let l = length of shared prefix let d = value of l-th digit in D's address if RouteTab[l,d] exists forward to RouteTab[l,d] else forward to known node with at least as long a shared prefix and numerically closer than this node
The routing table, denoted
RouteTab, is a two-dimensional array. It
has a row for every hex digit in an ID (there such 32 digits in a
128-bit ID) and a column for every hex value (there are obviously 16
such values). Every entry in row shares a prefix of length with
this node, and within this row the entry in column has the hex value
in the th position. Figure 9 shows the first three rows
of an example routing table for node
65a1fcx, where denotes an
unspecified suffix. This figure shows the ID prefix matched by every
entry in the table. It does not show the actual value contained in this
entry—the IP address of the next node to route to.
Adding a node to the overlay works much like routing a "locate object message" to an object. The new node must know of at least one current member. It asks this member to route an "add node message" to the node numerically closest to the ID of the joining node, as shown in Figure 10. It is through this routing process that the new node learns about other nodes with a shared prefix and is able to begin filling out its routing table. Over time, as additional nodes join the overlay, existing nodes also have the option of including information about the newly joined node in their routing tables. They do this when the new node adds a longer prefix than they currently have in their table. Neighbors in the leaf sets also exchange routing tables with each other, which means that over time routing information propagates through the overlay.
The reader may have noticed that although structured overlays provide a probabilistic bound on the number of routing hops required to locate a given object—the number of hops in Pastry is bounded by N, where N is the number of nodes in the overlay—each hop may contribute substantial delay. This is because each intermediate node may be at a random location in the Internet. (In the worst case, each node is on a different continent!) In fact, in a world-wide overlay network using the algorithm as described above, the expected delay of each hop is the average delay among all pairs of nodes in the Internet! Fortunately, one can do much better in practice. The idea is to choose each routing table entry such that it refers to a nearby node in the underlying physical network, among all nodes with an ID prefix that is appropriate for the entry. It turns out that doing so achieves end-to-end routing delays that are within a small factor of the delay between source and destination node.
Finally, the discussion up to this point has focused on the general problem of locating objects in a peer-to-peer network. Given such a routing infrastructure, it is possible to build different services. For example, a file sharing service would use file names as object names. To locate a file, you first hash its name into a corresponding object ID and then route a "locate object message" to this ID. The system might also replicate each file across multiple nodes to improve availability. Storing multiple copies on the leaf set of the node to which a given file normally routes would be one way of doing this. Keep in mind that even though these nodes are neighbors in the ID space, they are likely to be physically distributed across the Internet. Thus, while a power outage in an entire city might take down physically close replicas of a file in a traditional file system, one or more replicas would likely survive such a failure in a peer-to-peer network.
Services other than file sharing can also be built on top of distributed hash tables. Consider multicast applications, for example. Instead of constructing a multicast tree from a mesh, one could construct the tree from edges in the structured overlay, thereby amortizing the cost of overlay construction and maintenance across several applications and multicast groups.
BitTorrent is a peer-to-peer file sharing protocol devised by Bram Cohen. It is based on replicating the file or, rather, replicating segments of the file, which are called pieces. Any particular piece can usually be downloaded from multiple peers, even if only one peer has the entire file. The primary benefit of BitTorrent's replication is avoiding the bottleneck of having only one source for a file. This is particularly useful when you consider that any given computer has a limited speed at which it can serve files over its uplink to the Internet, often quite a low limit due to the asymmetric nature of most broadband networks. The beauty of BitTorrent is that replication is a natural side effect of the downloading process: As soon as a peer downloads a particular piece, it becomes another source for that piece. The more peers downloading pieces of the file, the more piece replication occurs, distributing the load proportionately, and the more total bandwidth is available to share the file with others. Pieces are downloaded in random order to avoid a situation where peers find themselves lacking the same set of pieces.
Each file is shared via its own independent BitTorrent network, called a swarm. (A swarm could potentially share a set of files, but we describe the single file case for simplicity.) The lifecycle of a typical swarm is as follows. The swarm starts as a singleton peer with a complete copy of the file. A node that wants to download the file joins the swarm, becoming its second member, and begins downloading pieces of the file from the original peer. In doing so, it becomes another source for the pieces it has downloaded, even if it has not yet downloaded the entire file. (In fact, it is common for peers to leave the swarm once they have completed their downloads, although they are encouraged to stay longer.) Other nodes join the swarm and begin downloading pieces from multiple peers, not just the original peer. See Figure 11.
If the file remains in high demand, with a stream of new peers replacing those who leave the swarm, the swarm could remain active indefinitely; if not, it could shrink back to include only the original peer until new peers join the swarm.
Now that we have an overview of BitTorrent, we can ask how requests are routed to the peers that have a given piece. To make requests, a would-be downloader must first join the swarm. It starts by downloading a file containing meta-information about the file and swarm. The file, which may be easily replicated, is typically downloaded from a web server and discovered by following links from Web pages. It contains:
The target file's size
The piece size
SHA-1 hash values precomputed from each piece
The URL of the swarm's tracker
A tracker is a server that tracks a swarm's current membership. We'll see later that BitTorrent can be extended to eliminate this point of centralization, with its attendant potential for bottleneck or failure.
The would-be downloader then joins the swarm, becoming a peer, by sending a message to the tracker giving its network address and a peer ID that it has generated randomly for itself. The message also carries a SHA-1 hash of the main part of the file, which is used as a swarm ID.
Let's call the new peer P. The tracker replies to P with a partial list of peers giving their IDs and network addresses, and P establishes connections, over TCP, with some of these peers. Note that P is directly connected to just a subset of the swarm, although it may decide to contact additional peers or even request more peers from the tracker. To establish a BitTorrent connection with a particular peer after their TCP connection has been established, P sends P's own peer ID and swarm ID, and the peer replies with its peer ID and swarm ID. If the swarm IDs don't match, or the reply peer ID is not what P expects, the connection is aborted.
The resulting BitTorrent connection is symmetric: Each end can download from the other. Each end begins by sending the other a bitmap reporting which pieces it has, so each peer knows the other's initial state. Whenever a downloader (D) finishes downloading another piece, it sends a message identifying that piece to each of its directly connected peers, so those peers can update their internal representation of D's state. This, finally, is the answer to the question of how a download request for a piece is routed to a peer that has the piece, because it means that each peer knows which directly connected peers have the piece. If D needs a piece that none of its connections has, it could connect to more or different peers (it can get more from the tracker) or occupy itself with other pieces in hopes that some of its connections will obtain the piece from their connections.
How are objects—in this case, pieces—mapped onto peer nodes? Of course each peer eventually obtains all the pieces, so the question is really about which pieces a peer has at a given time before it has all the pieces or, equivalently, about the order in which a peer downloads pieces. The answer is that they download pieces in random order, to keep them from having a strict subset or superset of the pieces of any of their peers.
The BitTorrent described so far utilizes a central tracker that constitutes a single point of failure for the swarm and could potentially be a performance bottleneck. Also, providing a tracker can be a nuisance for someone who would like to make a file available via BitTorrent. Newer versions of BitTorrent additionally support "trackerless" swarms that use a DHT-based implementation. BitTorrent client software that is trackerless capable implements not just a BitTorrent peer but also what we'll call a peer finder (the BitTorrent terminology is simply node), which the peer uses to find peers.
Peer finders form their own overlay network, using their own protocol over UDP to implement a DHT. Furthermore, a peer finder network includes peer finders whose associated peers belong to different swarms. In other words, while each swarm forms a distinct network of BitTorrent peers, a peer finder network instead spans swarms.
Peer finders randomly generate their own finder IDs, which are the same size (160 bits) as swarm IDs. Each finder maintains a modest table containing primarily finders (and their associated peers) whose IDs are close to its own, plus some finders whose IDs are more distant. The following algorithm ensures that finders whose IDs are close to a given swarm ID are likely to know of peers from that swarm; the algorithm simultaneously provides a way to look them up. When a finder F needs to find peers from a particular swarm, it sends a request to the finders in its table whose IDs are close to that swarm's ID. If a contacted finder knows of any peers for that swarm, it replies with their contact information. Otherwise, it replies with the contact information of the finders in its table that are close to the swarm, so that F can iteratively query those finders.
After the search is exhausted, because there are no finders closer to the swarm, F inserts the contact information for itself and its associated peer into the finders closest to the swarm. The net effect is that peers for a particular swarm get entered in the tables of the finders that are close to that swarm.
The above scheme assumes that F is already part of the finder network, that it already knows how to contact some other finders. This assumption is true for finder installations that have run previously, because they are supposed to save information about other finders, even across executions. If a swarm uses a tracker, its peers are able to tell their finders about other finders (in a reversal of the peer and finder roles) because the BitTorrent peer protocol has been extended to exchange finder contact information. But, how can a newly installed finder discover other finders? The files for trackerless swarms include contact information for one or a few finders, instead of a tracker URL, for just that situation.
An unusual aspect of BitTorrent is that it deals head-on with the issue of fairness, or good "network citizenship." Protocols often depend on the good behavior of individual peers without being able to enforce it. For example, an unscrupulous Ethernet peer could get better performance by using a backoff algorithm that is more aggressive than exponential backoff, or an unscrupulous TCP peer could get better performance by not cooperating in congestion control.
The good behavior that BitTorrent depends on is peers uploading pieces to other peers. Since the typical BitTorrent user just wants to download the file as quickly as possible, there is a temptation to implement a peer that tries to download all the pieces while doing as little uploading as possible—this is a bad peer. To discourage bad behavior, the BitTorrent protocol includes mechanisms that allow peers to reward or punish each other. If a peer is misbehaving by not nicely uploading to another peer, the second peer can choke the bad peer: It can decide to stop uploading to the bad peer, at least temporarily, and send it a message saying so. There is also a message type for telling a peer that it has been unchoked. The choking mechanism is also used by a peer to limit the number of its active BitTorrent connections, to maintain good TCP performance. There are many possible choking algorithms, and devising a good one is an art.
Content Distribution Networks
We have already seen how HTTP running over TCP allows web browsers to retrieve pages from web servers. However, anyone who has waited an eternity for a Web page to return knows that the system is far from perfect. Considering that the backbone of the Internet is now constructed from 40-Gbps links, it's not obvious why this should happen. It is generally agreed that when it comes to downloading Web pages there are four potential bottlenecks in the system:
The first mile. The Internet may have high-capacity links in it, but that doesn't help you download a Web page any faster when you're connected by a 1.5Mbps DSL line or a poorly performing wireless link.
The last mile. The link that connects the server to the Internet can be overloaded by too many requests, even if the aggregate bandwidth of that link is quite high.
The server itself. A server has a finite amount of resources (CPU, memory, disk bandwidth, etc.) and can be overloaded by too many concurrent requests.
Peering points. The handful of ISPs that collectively implement the backbone of the Internet may internally have high-bandwidth pipes, but they have little motivation to provide high-capacity connectivity to their peers. If you are connected to ISP A and the server is connected to ISP B, then the page you request may get dropped at the point where A and B peer with each other.
There's not a lot anyone except you can do about the first problem, but it is possible to use replication to address the remaining problems. Systems that do this are often called Content Distribution Networks (CDNs). Akamai operates what is probably the best-known CDN.
The idea of a CDN is to geographically distribute a collection of server surrogates that cache pages normally maintained in some set of backend servers. Thus, rather than having millions of users wait forever to contact when a big news story breaks—such a situation is known as a flash crowd—it is possible to spread this load across many servers. Moreover, rather than having to traverse multiple ISPs to reach , if these surrogate servers happen to be spread across all the backbone ISPs, then it should be possible to reach one without having to cross a peering point. Clearly, maintaining thousands of surrogate servers all over the Internet is too expensive for any one site that wants to provide better access to its Web pages. Commercial CDNs provide this service for many sites, thereby amortizing the cost across many customers.
Although we call them surrogate servers, in fact, they can just as correctly be viewed as caches. If they don't have a page that has been requested by a client, they ask the backend server for it. In practice, however, the backend servers proactively replicate their data across the surrogates rather than wait for surrogates to request it on demand. It's also the case that only static pages, as opposed to dynamic content, are distributed across the surrogates. Clients have to go to the backend server for any content that either changes frequently (e.g., sports scores and stock quotes) or is produced as the result of some computation (e.g., a database query).
Having a large set of geographically distributed servers does not fully solve the problem. To complete the picture, CDNs also need to provide a set of redirectors that forward client requests to the most appropriate server, as shown in Figure 12. The primary objective of the redirectors is to select the server for each request that results in the best response time for the client. A secondary objective is for the system as a whole to process as many requests per second as the underlying hardware (network links and web servers) is able to support. The average number of requests that can be satisfied in a given time period—known as the system throughput—is primarily an issue when the system is under heavy load, such as when a flash crowd is accessing a small set of pages or a Distributed Denial of Service (DDoS) attacker is targeting a particular site, as happened to CNN, Yahoo, and several other high-profile sites in February 2000.
CDNs use several factors to decide how to distribute client requests. For example, to minimize response time, a redirector might select a server based on its network proximity. In contrast, to improve the overall system throughput, it is desirable to evenly balance the load across a set of servers. Both throughput and response time are improved if the distribution mechanism takes locality into consideration; that is, it selects a server that is likely to already have the page being requested in its cache. The exact combination of factors that should be employed by a CDN is open to debate. This section considers some of the possibilities.
As described so far, a redirector is just an abstract function, although it sounds like what something a router might be asked to do since it logically forwards a request message much like a router forwards packets. In fact, there are several mechanisms that can be used to implement redirection. Note that for the purpose of this discussion we assume that each redirector knows the address of every available server. (From here on, we drop the "surrogate" qualifier and talk simply in terms of a set of servers.) In practice, some form of out-of-band communication takes place to keep this information up-to-date as servers come and go.
First, redirection could be implemented by augmenting DNS to return different server addresses to clients. For example, when a client asks to resolve the name , the DNS server could return the IP address of a server hosting CNN's Web pages that is known to have the lightest load. Alternatively, for a given set of servers, it might just return addresses in a round-robin fashion. Note that the granularity of DNS-based redirection is usually at the level of a site (e.g., ) rather than a specific URL (e.g., ). However, when returning an embedded link, the server can rewrite the URL, thereby effectively pointing the client at the most appropriate server for that specific object.
Commercial CDNs essentially use a combination of URL rewriting and DNS-based redirection. For scalability reasons, the high-level DNS server first points to a regional-level DNS server, which replies with the actual server address. In order to respond to changes quickly, the DNS servers tweak the TTL of the resource records they return to a very short period, such as 20 seconds. This is necessary so clients don't cache results and thus fail to go back to the DNS server for the most recent URL-to-server mapping.
Another possibility is to use the HTTP redirect feature: The client sends a request message to a server, which responds with a new (better) server that the client should contact for the page. Unfortunately, server-based redirection incurs an additional round-trip time across the Internet, and, even worse, servers can be vulnerable to being overloaded by the redirection task itself. Instead, if there is a node close to the client (e.g., a local Web proxy) that is aware of the available servers, then it can intercept the request message and instruct the client to instead request the page from an appropriate server. In this case, either the redirector would need to be on a choke point so that all requests leaving the site pass through it, or the client would have to cooperate by explicitly addressing the proxy (as with a classical, rather than transparent, proxy).
At this point you may be wondering what CDNs have to do with overlay networks, and while viewing a CDN as an overlay is a bit of a stretch, they do share one very important trait in common. Like an overlay node, a proxy-based redirector makes an application-level routing decision. Rather than forward a packet based on an address and its knowledge of the network topology, it forwards HTTP requests based on a URL and its knowledge of the location and load of a set of servers. Today's Internet architecture does not support redirection directly—where by "directly" we mean the client sends the HTTP request to the redirector, which forwards to the destination—so instead redirection is typically implemented indirectly by having the redirector return the appropriate destination address and the client contacts the server itself.
We now consider some example policies that redirectors might use to forward requests. Actually, we have already suggested one simple policy—round-robin. A similar scheme would be to simply select one of the available servers at random. Both of these approaches do a good job of spreading the load evenly across the CDN, but they do not do a particularly good job of lowering the client-perceived response time.
It's obvious that neither of these two schemes takes network proximity into consideration, but, just as importantly, they also ignore locality. That is, requests for the same URL are forwarded to different servers, making it less likely that the page will be served from the selected server's in-memory cache. This forces the server to retrieve the page from its disk, or possibly even from the backend server. How can a distributed set of redirectors cause requests for the same page to go to the same server (or small set of servers) without global coordination? The answer is surprisingly simple: All redirectors use some form of hashing to deterministically map URLs into a small range of values. The primary benefit of this approach is that no inter-redirector communication is required to achieve coordinated operation; no matter which redirector receives a URL, the hashing process produces the same output.
So what makes for a good hashing scheme? The classic modulo hashing scheme—which hashes each URL modulo the number of servers—is not suitable for this environment. This is because should the number of servers change, the modulo calculation will result in a diminishing fraction of the pages keeping their same server assignments. While we do not expect frequent changes in the set of servers, the fact that the addition of new servers into the set will cause massive reassignment is undesirable.
An alternative is to use the same consistent hashing algorithm discussed in the previous section. Specifically, each redirector first hashes every server into the unit circle. Then, for each URL that arrives, the redirector also hashes the URL to a value on the unit circle, and the URL is assigned to the server that lies closest on the circle to its hash value. If a node fails in this scheme, its load shifts to its neighbors (on the unit circle), so the addition or removal of a server only causes local changes in request assignments. Note that unlike the peer-to-peer case, where a message is routed from one node to another in order to find the server whose ID is closest to the objects, each redirector knows how the set of servers map onto the unit circle, so they can each, independently, select the "nearest" one.
This strategy can easily be extended to take server load into account. Assume the redirector knows the current load of each of the available servers. This information may not be perfectly up-to-date, but we can imagine the redirector simply counting how many times it has forwarded a request to each server in the last few seconds and using this count as an estimate of that server's current load. Upon receiving a URL, the redirector hashes the URL plus each of the available servers and sorts the resulting values. This sorted list effectively defines the order in which the redirector will consider the available servers. The redirector then walks down this list until it finds a server whose load is below some threshold. The benefit of this approach compared to plain consistent hashing is that server order is different for each URL, so if one server fails its load is distributed evenly among the other machines. This approach is the basis for the Cache Array Routing Protocol (CARP) and is shown in pseudocode below.
SelectServer(URL, S) for each server s in server set S weight[s] = hash(URL, address[s]) sort weight for each server s in decreasing order of weight if Load(s) < threshold then return s return server with highest weight
As the load increases, this scheme changes from using only the first server on the sorted list to spreading requests across several servers. Some pages normally handled by busy servers will also start being handled by less busy servers. Since this process is based on aggregate server load rather than the popularity of individual pages, servers hosting some popular pages may find more servers sharing their load than servers hosting collectively unpopular pages. In the process, some unpopular pages will be replicated in the system simply because they happen to be primarily hosted on busy servers. At the same time, if some pages become extremely popular, it is conceivable that all of the servers in the system could be responsible for serving them.
Finally, it is possible to introduce network proximity into the equation in at least two different ways. The first is to blur the distinction between server load and network proximity by monitoring how long a server takes to respond to requests and using this measurement as the "server load" parameter in the preceding algorithm. This strategy tends to prefer loaded servers over distant/heavily loaded servers. A second approach is to factor proximity into the decision at an earlier stage by limiting the candidate set of servers considered by the above algorithms () to only those that are nearby. The harder problem is deciding which of the potentially many servers are suitably close. One approach would be to select only those servers that are available on the same ISP as the client. A slightly more sophisticated approach would be to look at the map of autonomous systems produced by BGP and select only those servers within some number of hops from the client as candidate servers. Finding the right balance between network proximity and server cache locality is a subject of ongoing research.