Exam 1 study guide
The one-hour study guide for exam 1
Latest update: Fri May 6 15:41:35 EDT 2016
Disclaimer: This study guide attempts to touch upon the most important topics that may be covered on the exam but does not claim to necessarily cover everything that one needs to know for the exam. Finally, don't take the one hour time window in the title literally.
Terminology from the history of data communication
Data communication, the process of conveying information from one place to another, existed long before computers and dates back to earliest recorded history. As various pre-computer techniques for conveying data reliably over distances were invented, so too were a number of underlying mechanisms that proved to be useful in digital networks.
The most basic form of message delivery is moving a message from one entity (person) to another. This is known as point-to-point, or unicast delivery. Transmitting a single message so that everyone gets it is known as a broadcast. It’s the difference between a messenger and a smoke signal. Two crucial categories of data are control and message data. Message data is the information that needs to be delivered. Control data comprises the information that is used to manage the delivery of the message. It includes things such as acknowledgements, retransmission requests, and rate controls. In some cases, control data is sent via a separate communication channel than message data. In other cases, it is intermixed with some syntax for identifying which data is control and which is message.
Synchronization, in the context of messaging, is the coordination of activity between the sender and receiver of messages. It includes controls such as ready to send, ready to receive, transmission complete.
Congestion is the inability of a network element to receive or transmit messages at the desired rate, leading to a buildup or possibly a loss of messages and a deterioration in the quality of service. Flow control is the aspect of synchronization that deals with the rate of transmission, controlling when the sender transmits messages to keep the receiver or network from getting overwhelmed with data traffic. Rate control is the aspect of flow control that controls just the speed of transmission. Other elements of flow control may be directives to delay for a specific time or wait until some feedback is received. By reducing the flow of traffic, congestion can be alleviated.
An acknowledgement (also known as a positive acknowledgement) is a control message that states that a message was received. A negative acknowledgement is a control message that states that a message was not delivered to the destination or that it was received with errors. It is a form of error notification.
Packets may be lost or corrupted during transmission. With best-effort message delivery the network does the best it can do in providing service but makes no guarantee that data will be delivered to its destination or the time it will take for it to be delivered. Reliable delivery ensures that data does arrive reliably. Reliable delivery may be implemented as a layer of software on top of a network that provides best-effort delivery. With reliable delivery, if a message does not arrive or arrive correctly, the transmitter will be notified and will attempt to resend the message. Delivery time of the message can vary (since there is an extra delay to detect lost data dnt to retranmit it). With flow control mechanisms to relieve congestion as well as reliable delivery, there is generally no guarantee on the resulting bit rate between two communicating parties.
A key part of any communication is the encoding of data. This ranges from how data is represented in the medium (the component that carries data, whether it is radio frequency, a wire, or a piece of paper) to the meaning of the messages themselves. To expedite messaging in the past, table-based encoding was sometimes used to have a letter or a number represent an entire message instead of transmitting each character in the message.
A repeater, also known as an extender, is a device that regenerates a signal to cover longer distances. Pre-networking, this would be a relay station where one would switch horses or runners. These days, it is a often a device such as an ethernet extender that regenerates incoming signals to full strength, allowing longer runs of ethernet cable.
Origins of the Internet
The precursor to the Internet was ARPANET, a research project funded by DARPA, the Defense Advanced Research Projects Agency. This was an experiment in using packet switched networking to share computing resources over long distances. ARPANET got its start in late 1968 and was inspired by three developments. First, Leonard Kleinrock’s concept of packet switching: breaking messages into small chunks called packets and have them compete with packets from other messages on the same shared communication link. Second, J.C.R. Licklider’s vision of an “Intergalactic Network”, a globally connected set of computers where you can access data from or run programs on remote machines. Third, the demonstration of the viability of a wide-area network accomplished by connecting a computer in Massachusetts to one in California over a telephone line via a dial-up connection.
The crucial component in the early ARPANET was the Interface Message Processor, or IMP. This was the device that served as the interface between one or more connected host computers and the external packet-based ARPANET, processed the flow of packets, and sent them to other IMPs or to a connected host . It was the predecessor to the router. The ARPANET went live in 1969 with two computers and two more by the end of the year. In the early ARPANET, all the protocols for delivering packets were implemented in the IMPs. The software that ran on the computer and interfaced with the IMP was the Network Control Program, or NCP. This allowed applications to interface with the network and provided them with the abstraction of having a dedicated communication stream. It also handled flow control to the IMP and retransmission. As the ARPANET evolved, NCP became TCP, which handled reliable application-to-application communication.
ARPANET was designed to support the interconnection of networks (inter-networking). It is a network of networks rather than a single network to which computers connect. For an organization to join the ARPANET, there was no requirement for it to have a specific internal network. This layer of networking would be a logical layer on top of any underlying physical network.
Because there was no assumption on the design of any physical network, the ARPANET assumed that communication is not guaranteed to be reliable. Instead, it would provide best effort packet delivery. Software, originally NCP and later TCP, provided the ability to detect errors or lost packets and request retransmission.
Routers connect the various individual networks and links together that make up the Internet. While routers are crucial in determining where to send packets, they were designed to not store information about the flow of packets. Any packet can be routed on its own with no a priori connection setup.
Finally, the entire network is decentralized. There is no central administration or control point. This aspect not only makes the network scalable (no single point of congestion) but aids in making it a fault tolerant network. If a router is not working, it is likely that there may be alternate paths to the destination.
The ARPANET clearly demonstrated the value of wide-area, hardware agnostic networking but access to it was restricted to organizations working on U.S. Department of Defense projects. Other networks were created to cater to non-defense communities and some of these, such as NSFNET and CSNET, also chose to use the IP platform.
NSFNET was a government funded (by the NSF, the National Science Foundation) network that was created in 1985 to connect NSF-funded supercomputing centers. It initially did permit commercial traffic on its networks.
By the late 1980s, the NSF was looking for commercial partners who would provide wide area networking services. A collection of these partners removed any need for government funding. In 1990, the ARPANET was decommisssioned and in 1995, the NFSNET backbone was transitioned to commercial networks, leading to the Internet of today.
LAN and Internet structure
The Internet comprises the network edge and the network core. The network core is the set of interconnected networks that provide wide area connectivity to the customers of the network. The network edge is the set of devices and local networks that connect to the core network. Your computers, TV sets, thermostats, and local network constitute a network edge. Your Internet service provider and its Internet service provider are components of the network core.
Local area networks
A local area network, or LAN, is a data communication network that covers a relatively small area, typically a building. It uses a the same network access protocol and usually the same transmission medium (e.g., Ethernet), allowing message delivery without the need to route messages through different networks. Devices that send and receive messages on the network are called hosts or nodes. These devices are peers, meaning that no device has more control or coordination of the network than any other device. Any host can initiate a data transfer with any other host on the LAN. LANs usually exhibit very low latency and a high data rate: typically 10s to a gigabit per second (Gbps) for wireless networks and a 1 gigabit per second (Gbps) or more for wired connections (although speeds as high as 10 and 100 Gbps are available).
Nodes connect to a local area network with an adapter. These are usually integrated onto the main circuit board but may be separate components, such as a USB ethernet adapter. Another term for these is NIC, which stands for Network Interface Controller.
The physical data communication links that the adapter uses to send and receive data are called media. Common examples are unshielded twisted pair (UTP) copper wire (e.g., ethernet cable), radio frequency (e.g., the 5 GHz frequency bands used by 802.11ac), coaxial cable (e.g., used by cable TV in the home and the MoCA standard, multimedia over coax), and optical fiber (which is not commonly used in the home).
The other end of the media terminates at a switch or hub. A hub is a device that acts as a central point for multiple LAN cables. It takes any data that comes in one port and sends it to all the other ports. A switch is similar to a hub but smarter. It looks at incoming data and determines the port or ports on which to transmit it. Switches have largely replaced hubs. They provide scalable bandwidth in that they do not introduce more network congestion as you add more hosts onto your LAN. Switches and hubs are link-layer devices (more on that later). That is, they move ethernet packets to their destination as opposed to relaying data between networks. They are responsible for creating the physical network. For wireless networks, a wireless access point serves as link-layer switch.
The connection between the LAN and the Internet (via the Internet Service Provider) is called the access network. A residential gateway (a type of router) or access router connects a home or office LAN to the Internet. A modem, which stands for modulator/demodulator converts data between various analog formats as needed by the underlying media. Think of a modem as back-to-back NICs, each converting data to their type of media. Modems are generally built into access routers. Examples of access links are:
- DSL (digital subscriber line)
- DSL uses existing copper telephone wiring between your home and the phone company’s central office. Since voice uses only the 0 - 4 kHz range of frequencies, there is a lot of untapped bandwidth in a phone wire. DSL uses the 4 kHz through 50 kHz frequency band for upstream data (data you send) and the 50 kHz through 1 MHz band for downstream data (data you receive). A DSL modem serves as an access router and modem. At the phone company’s central office, the access link terminates at a DSLAM (digital subscriber line multiplexor). From there, the data signals are sent to the data network (Internet) and the voice signals are sent to the phone network.
- Internet service provided by a TV cable company uses the same coax cable that provides TV signals. With cable TV, hundreds of channels are transmitted at once, each occupying a different frequency band (high definition channels occupy a 6 MHz band and standard definition digital channels typically occupy a 1.5 GHz band). This type of transmission is called broadband. A certain number of channels are not used for TV services but instead are used for Internet access. The customer has an access router/modem that conforms to the DOCSIS (Data Over Cable Service Interface Specification) standard. Some number of channels are devoted to downstream communication (data you receive). Each channel provides 38 Mbps service. Another set of channels are devoted to upstream service, with each channel providing 27 Mbps service. The cable terminates at the cable company’s headend on a device called the CMTS, the cable modem termination system. Here, the Internet service frequencies are filtered to separate out the data and send it to the cable company’s Internet link. A key distinction between DSL and cable service is that the phone wire between the DSL modem and the phone company’s central office is a dedicated, non-shared line. The coax cable of cable TV service is shared among a neighborhood. However, even though the channel is shared, the capacity of coax media is greater than that of a phone line and its signal attenuation over distance is far lower.
- FTTH (Fiber to the Home), FTTN (fiber to the Neighborhood)
- Fiber offers even greater bandwidth than cable and can propagate signals longer distances without degradation. Fiber to the Home (FTTH) is an optical fiber link that connects a home to a central office and delivers a combination of TV, telephone, and Internet service. Verizon’s FiOS service is an example of a service that delivers fiber to the home. Access links in this architecture often use optical splitters to allow a single fiber from the central office to fan out to several homes (typically 16–128). FTTH requires an end-to-end fiber infrastructure, which is costly to deploy (Verizon spent $23 billion through 2010 deploying FiOS at a cost of over $800 per home). An alternative architecture that is designed to be more cost effective is Fiber to the Neighborhood (FTTN). AT&T’s U-verse service is an example of this. A fiber runs from the provider’s central office to the neighborhood (within a half mile of the customers’ homes). There, it goes to a mini headend that uses copper wiring to connect to each customer’s home via VDSL (very high bitrate DSL).
The organization that provides Internet service to a customer is called an Internet Service Provider (ISP). One ISP does not have access links to every Internet user in the world so there are thousands of ISPs: approximately 12,700 worldwide; 7,000 in the U.S. (about 100 or so larger-sized ones and lots of tiny regional ones). Most smaller ISPs that serve end users purchase Internet connectivity from other ISPs. ISP networsk are categorized by tiers. There isn’t a formal definition of what constitutes a tier but there are generally accepted conventions.
At the very top of the hierarchy, Tier 1 ISPs own the infrastructure that forms the backbone of the Internet. Each Tier 1 ISP has a peering agreement with every other Tier 1 ISP. This means that they agree to forward and receive traffic from each other without charging the other ISP for it. Tier 1 ISPs have access to the Internet routing table, also known as the global routing table. What this means that they know the top-level ISP to which any IP address should be sent. They also know which of their lower-tier ISPs receive any given IP address. As such, there is no concept of a “default route” at this level. With lower-tier ISPs, a router can give up if it does not know where a certain packet should go and just send it to a higher-level ISP. Tier 1 ISPs do not pay for data transit. Examples of Tier 1 ISPs include AT&T, Verizon, CenturyLink, Level 3, Telefónica, and NTT.
Tier 2 ISPs purchase data transit from Tier 1 ISPs and other Tier 2 ISPs but may also peer with other networks for direct connectivity and cost saving. They may then resell Internet access. Examples of Tier 2 ISPs include Comcast, British Telecom, Vodafone, and Sprint Communications.
Tier 3 ISPs occupy the lowest level and solely purchase Internet transit from one or more Tier 1 and Tier 2 IPSs. A Tier 3 ISP will typically provide coverage in a limited region of a country. Examples of these are Powernet Global, and Excel.Net. They concentrate on the retail and consumer markets.
A packet will often pass through several networks en route to its destination, both within and between ISPs. Each link terminates at a router which makes a decision on where the packet should be delivered. Each transmission of a packet is called a hop.
Within ISPs, edge routers are placed at the edge of an ISP’s network and communicate with other networks. For larger organizations, an edge router may sit at the edge of a customer’s network and connect to one or more ISPs. A core router is a router that connects routes on the Internet backbone. A core router may also be used in dispersed organizations to interconnect routers from multiple locations.
A network is inherently a shared resource. Lots of devices send and receive data on it. How do we share the network without clobbering each other’s data?
The most basic approach is to establish a dedicated connection from the source to the destination. This is a physical circuit and is what was used in the early days of the phone system: your phone line went to the central office where it connected to a patch cord that was in turn connected to the wire of the person with whom you were speaking. The same stream of electrons flowed from one end to the other. This is not a viable use of network resources. Moreover, we will likely have multiple applications running on a machine and using the network. We need to find a way to share network links.
One way can share the medium is to have each party communicate on different frequencies. This ability to transmit different data simultaneously is called broadband communication. Each transmission is assigned a frequency band: a portion of the total bandwidth. Broadband communications uses Frequency Division Multiplexing (FDM). Note that bidirectional communication on one channel is not possible: each sender-receiver set needs its own frequency band. Cable TV is an example of broadband.
An alternate method is to have everyone take turns in accessing the medium. This is called baseband communication. Each device is allowed full access to the medium’s bandwidth but only for a portion of time. Each communication session is assigned a specific set of short, fixed-length time slots during which it can transmit. This is called Time Division Multiplexing (TDM). Time Division Multiplexing is an example of circuit switching.
Both FDM and TDM are examples of circuit switching. Circuit switching sets up a dedicated communication channel, similar to a physical circuit. The key difference from a physical circuit is that the effective bandwidth is lower than the capacity of the medium since it is shared. With Frequency Division Multiplexing, the available bandwidth is sliced into frequency ranges. With Time Division Multiplexing, the available bandwidth is sliced into time slots.
An althernate way of sharing a medium is to use variable-size time slots with no scheduling on the transmitter’s part. This is called packet switching.
Circuit switching requires a connection setup (or circuit setup). A control message is sent from the source to establish a path (route) from the source to the destination. Every switching element in the path agrees to the setup of the path and allocates the appropriate time slots (and other resources, such as memory buffers). The originating node is then informed that the connection is established and communication can take place. The path and all the switching resources (e.g., time slices or frequency bands) remain allocated to the communication session whether data is being sent or not. All data travels along this predetermined path from the source to the destination. When the communication is complete, the sender “releases” the circuit. This results in the sending of another control message through the path informing routers to de-allocate the resources they were using for the session.
Benefits of circuit switching are that it offers constant latency and guaranteed bandwidth. Another benefit is that data routing decisions do not have to be made for each message that is transmitted. They are made once at the start of the communication session. Data can flow through a router without having to be stored first until a decision is made where and when to transmit it. The downside of circuit switching is that each connection ties up bandwidth and switching resources whether any data is being transmitted or not. Conversely, if a connection needs to transfer a larger amount of data, it still needs to spread it over bandwidth given to it even if the rest of the network is not being used at the moment. In short, circuit switching does not use network resources efficiently. Each circuit is allocated a fixed bandwidth whether it is used or not.
With packet switching, a communication stream is broken into chunks of data called packets. Each packet must contain a destination address in its message header. The packets travel from the source node to their final destination via packet switches. Routers and ethernet switches are examples of packet switches. Routers are used to transmit packets between different networks and switches are used to transmit packets within a local area network. Each packet switch decides on the disposition of the packet (to what port it should be transmitted) based on the packet’s destination address. There is no need to retain memory of a predefined route for a stream of packets that represents a communication stream. In fact, there is no real concept of a communication stream because no routes have to be set up and no resources need to be reserved ahead of time. Because packet switching is designed for baseband networks, each packet has full use of the network link. If there are no other packets transmitted on the network, a node may see its available bandwidth approach the maximum capacity of the network link.
Packet switched traffic is known as datagram service in contrast to circuit switched virtual circuit service. Think of a datagram as a telegram or letter, where each message has to be addressed individually and may take a different path through the network. Think of a virtual circuit as a telephone call where the call is first set up but then gets an established route constant bandwidth for its duration.
Packet switching employs statistical multiplexing. Multiplexing means dividing a communication channel among multiple data streams. With TDM’s circuit switching, a communication channel was divided into fixed time slots per data stream. With packet switching, we are still sharing the network but now using variable time slots. What this means is that if a node has a lot of data to transmit and others do not then it can transmit large packets (or a lot of smaller packets) and use more of the network. If a node has little to transmit, it will use less of the network and more network capacity will be available for others. Of course, there might be times when a node may have to wait longer for the network to be free. If a lot of nodes have a lot of data to transmit, they will collectively have to wait longer to use the network, some more than others. Similarly, routers may end up queuing packets for a particular outbound port. This leads to variable latency. Packet switching is characterized by variable bandwidth and variable latency. With packet switching, an entire packet needs to be received by a router before it is transmitted on an outgoing link. This is called store and forward delivery and also contributes to network latency as we will see in our discusion on delay and throughput.
Despite its variable bandwidth and variable latency, packet switching allows for far more efficient use of the network than circuit switching and has no limit on the number of concurrent communication sessions. Because, on average, applications do not use the network non-stop, switching and link resources are wasted whenever data is not flowing on an established connection. With packet switching, there is no such reservation of these resources and more streams can be accommodated while providing the same bandwidth to applications. Packet switching is the dominant means of data communication. The Internet is built around packet switching.
Throughout our discussions on networking, we will bring up units of measure. The three crucial ones for us are the size of data, the speed at which it moves, and the time that it takes for it to get somewhere.
- The fundamental unit of size is a bit (b). Eight bits make a byte (B). Packet sizes are generally measured in bytes (watch out for the factor of eight when working with bits per second!). Networks tend to use base–10 units, so a kilobyte (KB) is 1,000 bytes rather than the 1,024 bytes we are used to in programming. A kilobit (Kb) is 1,000 bits. A megabit (Mb) is 106 bits and a gigabit (Gb) is 109 bits. A megabyte (1 MB) is 1,000 KB or 106 bytes or 8×106 bits.
- Time is measured in seconds (s). One second (1 s) = 1,000 ms (milliseconds) = 106 μs (microseconds) = 109 ns (nanoseconds).
- Rate is measured in bits per second (b/s or bps). Moving a megabit (1 Mb) of data over a 10 Mbps (megabit per second) network will take (1×106 b ÷ 1×106; bps) = 0.1 s, or 100 ms. Transmitting 1 kilobit on a 1 Gbps link will take (kilo ÷ giga) = (103 ÷ 109) = 10–6 = 1 μs, or one millionth of a second.
Delay and throughput in networks
As a packet flows from its source to its ultimate destination, it goes through multiple routers. Each router introduces a delay as does the transit of the packet over the communication link.
With packet switching, a packet must be fully received by a router before it can be sent out. This is store and forward packet delivery. To see how this contributes to overall delay, let us consider each link. If data is transmitted at R bits per second and a packet is L bits long, it takes L/R seconds to transmit a packet from one link to the next. Since transmission on the next link will not start until the packet is received, each link adds a delay of L/R seconds. With N links (there are N–1 routers or transmitters but we also count the delay of the initial transmission), we have a total delay of N(L/R) seconds.
Network delay is due to four factors:
Processing delay. The processing delay is the computation that a router has to do to examine the header, check for packet errors, figure out the outbound port (route), and move data around. It is usually not a significant contributor to the overall delay and consumes a few microseconds.
Transmission delay. The transmission delay is the time that it takes to get a complete packet out onto the network. This is a function of the speed of the link (e.g., 1 Gbps) and the number of bits in the packet: (packet size ÷ transmission speed). If the packet size is L and the transmission rate is R, the transmission delay is L/R.
Propagation delay. The propagation delay is the time it actually takes the signal to move from one end of the medium to the other. While we might transmit the bits onto the network at, say, 100 megabits per second, there is a delay between the time that the signal is sent and the signal is received. This is the speed of signal propagation in the medium. For electrical signals in unshielded twisted pair or for light pulses in fiber optics, this value is approximately 2×108 m/s (about 67% of the speed of light in a vacuum). An electrical signal propagates in air on a wireless network at approximately 3×108 m/s. Depending on the distance the packet needs to travel, the delay may be from a few nanoseconds to a few tens of milliseconds. It might be considerably longer for satellite transmission due to the longer distance covered.
Queuing delay. With packet based networks, we can only transmit one packet onto a link at a time. Any other packets that need to go out on that link will need to wait in a queue. The queuing delay is a function of the amount of bits that are ahead of the packet (number of packets × the size of each packet) and the transmission rate of the outbound link. Queuing delay can vary a lot depending on how much data traffic is flowing over any particular link. It is dependent on how much traffic arrives at a router at approximately the same time that needs to go out on the same link and on how quickly the router can transmit the data out (see transmission delay).
One useful measure for estimating the likelihood of queuing delays is traffic intensity. Traffic intensity is the average packet transmission delay (L/R; see transmission delay) multiplied by the average rate of packet arrival. If the average rate of packet arrival is a, traffic intensity is La/R. It is technically a unitless quantity since packets/second × bits/packet ÷ bits/second cancel out. [Trivia - not on the exam: the unit of this measure is called an erlang and refers to the load on a network.]
If the traffic intensity is greater than one, that means that, on average, packets arrive faster than they can be transmitted and the queue will keep growing without bound. This assures us that the queue will eventually overflow and packets will have to be dropped, leading to packet loss.
If the traffic intensity is less than or equal to one, packets are arriving slower or at the same speed that they are being transmitted. This does not mean that packets will never get queued up. A number of packets may occasionally arrive in rapid succession — a burst — and will have to be queued. As traffic intensity approaches one, the probability that there will be bursts of packets that need to be enqueued increases drastically. Hence, as traffic intensity approaches 1, queuing delay starts to increase dramatically. In some cases, this will lead to lost packets due to limited queue sizes (routers have only a fixed amount of memory to devote to queues).
The total delay for a node is the sum of the four delays we just mentioned: processing + queue + transmission + propagation. The total delay for N links in a store-and-forward network is simply N times that amount.
Data networking is generally implemented as a stack of several protocols – each responsible for a specific aspect of networking. The OSI reference model defines seven layers of network protocols.
- 1. Physical
- Deals with hardware, connectors, voltage levels, frequencies, etc. This layer does not care about contents but defines what constitutes a 1 or a 0. Examples of this layer are USB and Bluetooth.
- 2. Data link
- Sends and receives packets on the physical network. It may detect and possibly correct errors but only on this link (for example, queue overflow at a router may still cause packet loss). Ethernet packet transmission is an example of this layer.
- 3. Network
- Relays and routes data to its destination. This is where networking gets interesting because we are no longer confined to a single physical network but can route traffic between networks. IP, the Internet Protocol, is an example of this layer.
- 4. Transport
- Provides a software endpoint for networking. Now we can communicate application-to-application instead of machine-to-machine. Each application can create one or more distinct streams of data. TCP/IP and UDP/IP are examples of this layer.
- 5. Session
- Manages multiple logical connections over a single communication link. Examples are SSL (Secure Sockets Layer) tunnels and remote procedure call connection management.
- 6. Presentation
- Converts data between machine-specific data representations. Examples are data representation formats such as MIME (for media encoding on email), XML, XDR (for ONC remote procedure calls), NDR (for Microsoft COM+ remote procedure calls), and ASN.1 (used for encoding cryptographic keys and digital certificates).
- 7. Application
- This is a catch-all layer that includes every application-specific communication protocol. For example, SMTP (sending email), IMAP (receiving email), FTP (file transfer), HTTP (getting web pages).
The OSI reference model gives us a terminology to discuss and compare different networks. Any specific network may not necessarily implement all these layers. The Internet protocol stack relies on layers 1 through 4 (physical through transport) but it is up to applications to implement and use session, presentation, and, of course, application layers.
A key aspect of this layering approach is that each layer only has to interact with the corresponding layer on the other side. For example, an application talks to another application. TCP on one system deals with issues of retransmission and message acknowledgement by talking to the TCP layer on the remote system. A layer also does not need to be aware of the implementation of layers above it or below it: they are just data sources and data sinks.
if we want to send an IP packet (layer 3) out on an Ethernet network (layers 1 and 2), we need to send out an Ethernet packet (an Ethernet NIC or transceiver knows nothing about IP). The entire IP packet becomes the payload (data) of an Ethernet packet. Similarly, TCP and UDP, layers above IP, have their own headers, distinct from IP headers (they need a port number, for example). A TCP or UDP packet is likewise treated simply as data by the IP layer. This wrapping process is known as protocol encapsulation. Each layer of the networking stack can ignore the headers outside of its layer and treat anything from higher layers simply as part of the payload that needs to be sent.
The Application Layer
There are two ways that network applications are structured: client-server and peer-to-peer.
- This is the dominant model of interaction. One application, called the client (and usually run by the end user), requests something from another application, called a server. The server provides a service. Examples of this are a web browser (client) requesting a web page from a web server, a mail application (client) accessing a mail server to get mailbox contents, or a print server being given content to print. In this model, clients communicate with the server and not with other clients.
- A peer-to-peer architecture employs a collection of applications, any of which can talk to any other. These applications are peers and are generally run by a collection of end users rather than some service provider. The name peer implies that there is no leader: applications all have equal capabilities. An appealing aspect of a peer to peer design is self-scalability. As more and more nodes join the collection of peers, the system has more peers to do the work and can hence handle a large workload. Examples of peer-to-peer architectures are BitTorrent and Skype.
- A difficulty with peer-to-peer architectures is that one often needs to do things such as keep track of peers, identify which system can take on work or has specific content, and handle user lookup and authentication. This led to a variation of the peer-to-peer model where a coordinator, a central server, is in place to deal with these centralized needs. However, the peers still handle all the bandwidth-intensive or compute-intensive work.
No matter what architecture is used, there is still a fundamental client-server relationship. One system (a client) will send some request to another (a server).
Network Application API
When we write network-aware applications, they need to use the network to communicate with each other. These applications are no different than any other processes on the computer. Any process can access the network and it is up to the operating system to coordinate this access. When writing applications, the programmer will use a set of interfaces referred to as a Network API (Application Programming Interface) to interact with the network and not worry about the lower layers of the network. For example, a programmer does not need to know about ethernet or IP to communicate with another program but needs to have available the abstraction of being able to send data from one logical port on an application to one on another application.
The communication session is the conducted by the application layer protocol. This is a definition of the valid sequence of requests, responses, and their respective message formats for a particular network service. The protocol needs to be well-defined for applications to be able to communicate with each other.
Given a well-defined protocol, any application should be able to follow the rules of the protocol and create messages that the other side can understand, regardless of the implementation language or operating system. For instance, an iPad running an Swift should be able to talk to a mail server written in Java running on a Windows platform.
The Network API will have core services, such as those related to sending and receiving data, that are provided by the operating system. These are augmented with libraries to handle other functions, such as looking up names, converting data, and simplifying certain operating-system interfaces.
As programmers writing network-aware applications, we obviously need functions for sending and receiving data but we may want to be able to specify something about the behavior of that data over the network. For example:
- Do we need reliable data transfer?
- Is it important to the application that data arrives reliably and in order at the destination? It seems like the answer should always be yes, but we have to realize that ensuring reliability entails the detection, request for, and retransmission of lost packets. This adds a considerable delay to the delivery of that lost packet. For streaming media applications, such as telephony, that packet may arrive too late. In this case, it is useless to the application and the application could just as easily unsed best-effort service. Moreover, some applications may choose to handle retransmission requests themselves in a different manner. Applications that can handle unreliable media streams are called loss-tolerant applications.
- An application may have specific bandwidth requirements. For example, video and voice telephony applications may have minimum bandwidth needs. Applications with such needs are bandwidth sensitive applications. Applications that can adapt to whatever bandwidth is available (for example, switch to a lower bandwidth codec) are known as elastic applications.
- Delay and Jitter
- Interactive applications, such as voice and video telephony may want to ensure minimal network delay to reduce the delay to the receiver. Jitter is the variation in delay and these applications would also like to see low jitter.
- Applications may need to ensure that they are truly communicating with the proper computer and a legitimate application on that computer. They may be concerned about the integrity of the data that is being transmitted and want to ensure that it cannot be modified or read by outside parties.
These are all legitimate desires. Unfortunately, IP gives us no control over throughput, delay, jitter, and security. We can handle security at the application layer and we will later examine mechanisms that were added to IP to support some degree of control over packet delivery.
IP Transport Layers
Applications interact with IP’s transport layer. There are two dominant transport-layer protocols on top of IP (IP is the network layer): TCP and UDP (there are a few others, such as the SCTP, but these two dominate).
TCP, the Transmission Control Protocol, provides connection-oriented service. This does not imply that an actual network connection is being set up as one would for a circuit-switched network. We are strictly talking about transport-layer services here. For layer 2 and layer 3 protocols, a connection refers to setting up a pre-defined route (circuit) and providing that connection with guaranteed bandwidth. At the transport layer (4), we still strive strive to provide the illusion of a reliable bidirectional communication channel but it is all done in software on top of unreliable datagrams (IP). At the transport layer, the software does not have any control of the route that packets take or the bandwidth that is available to the connection.
The TCP layer of software ensures that packets are delivered in order to the application (buffering them in memory in the operating system if any arrive out of order) and that lost or corrupt packets are retransmitted. TCP keeps track of the destination so that the application can have the illusion of a connected data stream (just keep feeding data to the stream and don’t worry about addressing it). TCP provides a full-duplex connection, meaning that both sides can send and receive messages over the same link. TCP is stream oriented, meaning that data is received as a continuous stream of bytes and there is no preservation of message boundaries.
UDP, the User Datagram Protocol is designed as a very thin transport layer over IP. It provides connectionless service, also known as datagram service. While UDP drops packets with corrupt data, it does not ensure in-order delivery or reliable delivery. UDP’s datagram service preserves message boundaries. If you send n messages, you will receive n messages; they will not be combined into one message.
Port numbers in both TCP and UDP are used to allow the operating system to direct the data to the appropriate application or, more precisely, to the socket that is associated with the communication stream on the application. A port number is just a 16-bit number that is present in both TCP and UDP headers to identify a specific endpoint on a node.
Sockets are an interface to the network provided to applications by the operating system. They were created at the University of California at Berkeley for 4.2BSD (a derivative of UNIX) in 1983 and most operating systems now support this interface. The purpose of sockets is to provide a protocol-independent interface for applications to communicate with each other. The underlying network does not have to by IP. Once set up, the socket looks like a file descriptor for an open file. With connection-oriented (e.g., TCP) sockets, you can use the regular file system read and write system calls to receive and send data.
Sockets are the mechanism that the operating system exposes to the user for accessing the network. A socket is created with the socket system call and assigned a local address and port number with the bind system call. The OS can fill in defaults if you do not want to specify a specific address and port. (Note that you specify an address because your system might have multiple IP addresses; one for each of its network interfaces). The socket also requires that the programmer identify the address family for the socket (e.g., the protocol stack: IP, IP version 6, Bluetooth, local) as well as the mode of communication (e.g., connection-oriented or datagrams).
Sockets for connection-oriented protocols (streams)
For connection-oriented protocols (TCP), a socket on a server can be set to listen for connections with the listen system call. This turns it into a listening socket. Its only purpose will now be to receive incoming connections.
The accept call waits for a connection on a listening socket. It blocks until a connection is received, at which point the server receives a new socket that is dedicated to that connection.
A client establishes a connection with the connect system call. After the connection is accepted by the server, both sides now have a socket on which they can communicate.
Sending and receiving data is compatible with file operations: the same read/write system calls can be used. Data communication is stream-oriented. A sender can transmit an arbitrary number of bytes and there is no preservation of message boundaries.
When communication is complete, the socket can be closed with the shutdown or close system calls.
Sockets for connectionless protocols (datagrams)
With connectionless protocols, there is no need to establish a connection or to close one. Hence, there is no need for the connect, listen, or shutdown system calls.
Unlike connection oriented sockets, data communication is message-oriented. A sender transmits a message. The size of the message is limited to the maximum size allowable by the underlying network (the MTU, maximum transfer unit).
Because you need to specify the destination as the operating system does not keep state of a “connection”, new system calls were created for sending and receiving messages. The sendto and recvfrom system calls are used to send and receive datagrams. sendto allows you to send a datagram and specify its destination. recvfrom allows you to receive a datagram and identify who sent it.
The Java interface to sockets
Java provides many methods to deal with sockets and some commonly-used ones consolidate the sequence of steps that need to be take place on the operating system. The constructor for the ServerSocket class creates a socket for the TCP/IP protocol, binds it to a specified port and host (using defaults if desired), and sets that socket to the listening state. A client’s Socket class constructor creates a socket for the TCP/IP protocol, binds it to any available local port and host, and connects to a specified host and port. It returns when the server accepts the connection. The connected socket object allows you to acquire an InputStream and OutputStream for communication.
Threads, concurrency, and synchronization
A process normally has one thread of execution, or flow of control. A process may be multithreaded, where the same program has multiple concurrent threads of execution.
In a multi-threaded process, all of the process’ threads share the same memory and open files. Within this shared memory, each thread gets its own stack, which is where return addresses from functions are placed and where local variables get allocated. Each thread also has its own instruction pointer and registers. Since memory is shared, it is important to note that there is no memory protection among the threads in a process. Global variables are freely accessible by all threads. In particular, the heap, the pool of memory that is used for dynamic memory allocation is shared and freely accessible to all threads in the process.
Advantages of threads
There are several benefits in using threads. Threads are more efficient than processes. The operating system does not need to create and manage a new memory map for a new thread (as it does for a process). It also does not need to allocate new structures to keep track of the state of open files and increment reference counts on open file descriptors. Threading maps nicely to multicore architectures and allows for the effective use of multiple process cores.
Threading also makes certain types of programming easy. While it’s true that there is a potential for bugs because memory is shared among threads, shared memory makes it trivial to share data among threads. The same global and static variables can be read and written among all threads in a process. For a network server process, threading is appealing because it becomes easy to write code that handles multiple client requests at the same time. A common programming model is to have one master thread that waits for client connections and then dispatches a worker thread to handle the request.
While thread usage differs slightly among languages, there always needs to be a mechanism to create a thread and to wait for threads to exit. When a thread is created, a specific method (function) is called in that new execution flow. The original execution flow (the one thread that started when the process began) continues normally. When that thread eventually returns from that method, the thread terminates. If a thread needs to wait for another thread, it can choose to block until the other thread terminates. This is called a join.
Mutual exclusion: avoiding stepping on each other
Because threads within a process share the same memory and hence share all global data (static variables, global variables, and memory that is dynamically-allocated via malloc or new), there is an opportunity for bugs to arise where multiple threads are reading and writing the same data at the same time. A race condition is a bug where the outcome of concurrent threads is unexpectedly dependent on a specific sequence of thread scheduling. Thread synchronization provides a way to ensure mutual exclusion, where we can have regions of code that only one thread can execute at a time. Any other thread that tries to run in that region of code will go to sleep (be blocked) until the lock is released when the current thread in that region leaves it.
Java allows a synchronized keyword to be added to a method to ensure
that no more than one thread will be allowed to run in that method. If that degree of
control is too coarse, Java also allows the programmer to use the
keyword to define a region of code that will be locked by a variable called a
monitor object. Any other thread that tries to enter any region
of code that is synchronized by the same monitor object will be blocked. This region
of code that provides mutual exclusion is called a synchronized block.
Domain Name System
A node on the Internet is identified by its IP address. For IP version 4, the most common version deployed today, an IP address is a 32-bit value that is expressed as a set of four bytes, each as a decimal number and separated by dots. For instance, the IP address of the Rutgers web server is 188.8.131.52. [IP version 6, which is rapidly expanding as we are out of IPv4 addresses in some areas, is a 128-bit value and is expressed as a set of 8 groups of four hexadecimal digits.] As humans, however, we prefer to identify endpoints by name rather than by a number. For example, we think of the Rutgers web server by its name, www.rutgers.edu. We will now explore the management of IP domain names and IP addresses and converting between them.
How are IP addresses assigned
IP addresses are distributed hierarchically. At the very top level, an organization called the IANA (Internet Assigned Numbers Authority) is responsible for the entire set of IP addresses. It allocates blocks of addresses to Regional Internet Registries (RIR). There are five RIRs, each responsible for a part of the world’s geography. For instance, the U.S. and Canada get addresses from ARIN, the American Registry for Internet Numbers. Countries in Europe and the mid-East get addresses from the RIPE Network Coordination Centre. These RIRs in turn allocate blocks of IP addresses to ISPs within their region. Since ISPs are tiered, an ISP may allocate a smaller block of addresses to a lower-tier ISP as well as to a company that subscribes to its services.
How are names assigned
In the early days of the ARPANET, each machine had to have a globally unique name. The Network Information Center (NIC) at the Stanford Research Institute (SRI) kept the master list of machine names and their corresponding IP addresses. This solution does not scale. As the number of hosts on the Internet grew larger, a domain hierarchy was imposed on the name space. This created a a tree-structured name space with name management delegated to the various nodes of the tree, where each node is responsible for the names underneath it. Rutgers, for example, can name a new machine on the Internet anything it wants as long as the name is unique within Rutgers and is suffixed with rutgers.edu. The textual representation of Internet domain names is a set of strings delimited by periods with each set representing a level in the naming hierarchy. The rightmost string is the highest level in the hierarchy.
The hierarchy of Internet domain names has a single root under which are top-level domains (TLDs). These are the .com, .edu, .org suffixes that you are familiar with. Currently, there are 1,239 top-level domains. They are divided into two categories: generic top-level domains and country-code top-level domains.
Generic TLDs (gTLD) include the .com, .edu, .gov, .net, etc. domains. Many of them date back to the first proposal of creating a domain hierarchy RFC 920. Each of these domain names is three or more characters long. As the Internet became international, country-specific domains were created. These Country-code TLDs (ccTLDs) are two-letter ISO 3166 country codes (e.g., .ad for Andorra, .dk for Denmark, .es for Spain, .us for the U.S.A.). The root of the domain hierarchy initially allowed only US-ASCII (Latin) characters. This rule changed in 2009 and a new set of Internationalized Domain Names for country code top-level domains (IDN ccTLD) became available. Examples of these domains are السعودية. for Saudi Arabia, .рф for Russia, and .中國 for mainland China. In 2011, internationalized domain names were approved for generic top-level domains (IDN gTLD), giving us domains such as .みんな (“everyone” in Japanese), .移动 (“mobile” in Chinese), and .дети (“kids” in Russian).
Each top-level domain has one administrator assigned to it. The IANA keeps track of the organizations that manage the various top-level domains. Until 1999, for example, a company called Network Solutions Inc. operated the .com, .org, and .net registries. Until that time, Network Solutions maintained the registry of names and processed registration requests from customers. Since then, the process has been decentralized to support shared registration. This allows multiple companies to provide domain registration services. One company is still assigned by the IANA to be the keeper of the master list for a specific top-level domain. This list of registered domain names for a particular TLD is called the domain name registry. The company that maintains this registry is called the domain name registry operator, also known as the network information center (NIC). The IANA keeps track of all these organizations. A domain name registrar is a company that provides domain registration services to customers, allowing them to register domain names for a fee. There are approximately 2,124 of these companies. Examples of these are GoDaddy (with over 60 million domains), Namecheap, eNom, and Tucows.
When you pay GoDaddy, the registrar, $11.99 to register poobybrain.com, it consults the .com domain name registry at Verisign, which is the registry operator for the .com domain. If the domain name is available, GoDaddy becomes the designated registrar for that domain. This means that Verisign knows that Go Daddy has information on the owner of poopybrain.com and that changes and requests to transfer ownership or the registrar of your domain will have to come from that registrar. Of the $11.99 that you paid GoDaddy, $7.85 went to Verisign as a registry fee (different TLDs have different fees; .net registration costs $7.46). A $0.18 yearly fee that goes to ICANN to manage the registry.
Associating names with addresses
We now saw how IP addresses are allocated and how domain names are registered. There is no correlation between the two of them. A domain name does not imply a specific address and adjacent IP address numbers may belong to completely different domain names. We need a way to look up www.rutgers.edu and find out that its address is 184.108.40.206 since the IP layer knows absolutely nothing about domain names. Since the network core has no interest in domain names, name-to-address resolution is handled at the network edge, in the application before it establishes a socket connection. The process of looking up a name is an application-layer protocol.
In the past, Stanford Research Institute’s Network Information Center maintained the entire list of hosts on the internet (in a hosts.txt file; /etc/hosts on Unix systems). This file would be periodically downloaded by every system on the Internet. Clearly, this solution was not sustainable. We already saw that it made managing unique names problematic. Moreover, with millions of hosts on the Internet, there was a lot of churn in this database. Downloading a new copy of every host on the Internet constantly just doesn’t make sense.
The system that was put in place was a database of DNS servers (Domain Name System servers). Like domain names themselves, DNS is a distributed, hierarchical database.
A DNS server is responsible for a managing a sub-tree in the domain name hierarchy. For example, a server might be responsible for everything under rutgers.edu or even just the machines under under cs.rutgers.edu. This sub-tree of a group of managed nodes is called a zone. Each authoritative name server is responsible for answering queries about its zone. The authoritative name server for rutgers.edu is therefore responsible for the rutgers.edu zone. The question now is, how do you find it?
A DNS server accepts queries from clients (called questions) and provides responses (called answers). By default, interactions with DNS servers use UDP for improved performance, although TCP is almost always supported as well.
Any DNS server can be found by starting at the top of the name hierarchy. There are 13 root name servers that can provide a list of authoritative name servers for all the top-level domains (you can download the list of root name servers here. By contacting any one of these servers, you can find out the address of a name server responsible for a specific top-level domain (such as .edu). Then, by querying a name server for that domain (e.g., the .edu name server), you can find a name server responsible for a name within that domain (such as rutgers.edu). The process can continue until you find a name server that is responsible for the zone that contains the host you need.
There are two basic approaches for name resolution when dealing with a hierarchy of name servers: iterative or recursive queries. An iterative query is a single query to a name server. That server will return the best answer it can with the knowledge it has. This can be the the information configured for its zone (e.g., the domain names for which it is responsible) or cached results. If it does not have an answer to the query, it may return a referral. A referral is the name server for the next lower layer, taking you closer to your destination. For example, the root server can return a referral to tell you how to get to the .edu name server. A query to the .edu name server can return a referral to tell you how to get to the rutgers.edu name server. The advantage of this approach is that each name server can be completely stateless. It either knows the answer or it does not.
With recursive resolution, the DNS server takes on the responsibility of performing the set of iterative queries to other DNS servers on behalf of the requestor and sends back a single answer. With recursive resolution, a DNS server may first send a query for the full domain name to the root name server. The root name server will return a referral to, for example, the .edu name server. A query to that server will then return a referral to the rutgers.edu name server. In reality, the recursive server will cache past lookups so it will likely know the addressses of recently-used top-level domains.
The advantage of recursive resultion is that it incurs less communication at the client, simplifies the client’s protocol, and allows for caching of results at all the intermediate servers. The disadvantage is that a recursive server has to keep state about the client’s request until it has completed all processing and is ready to send a response back to the client.
A DNS server is not obligated to support recursion. Most top-level DNS servers, such as root servers, do not support recursive queries.
How does a DNS query work?
The client interaction with DNS is via a DNS resolver. This is a a DNS server that is not necessarily part of the DNS hierarchy (that is, it does not have to be a server responsible for a zone). However, it is capable of taking a recursive request from a client and performing a set of iterative queries, going to the root servers if necessary, to get the result. Resolvers could be hosted on the client, within the organization, or by third parties such as Google Public DNS, or OpenDNS. Most ISPs provide a DNS resolver service. Many systems (such as Windows and Linux platforms) support extremely limited local DNS resolvers that are incapable of iterative queries and simply talk to another DNS server (e.g., a resolver hosted by the customer’s ISP). These limited DNS resolvers are called stub resolvers.
DNS resolvers maintain a local cache of frequently used lookups to avoid the overhead of repeated lookups for the same name and to avoid the overhead of iterative queries. For example, it does not make sense to look up the name server responsible for .com over and over for each query (it’s 220.127.116.11, by the way).
Let us look at the sequence of operations that a query for www.cs.rutgers.edu might take from an application. We assume that the client machine is configured to use OpenDNS as a DNS resolver service.
Rutgers registers its domain with educause.edu, the domain registrar for names in the edu TLD. It provides Educause with a list of DNS servers that can answer queries for names underneath rutgers.edu. Educause.edu, in turn, registers its DNS servers with ICANN, who is responsible for the data in the root name servers.
The client application contacts a local DNS stub resolver. This checks its cache to see if it already has the answer. It also checks a local hosts file to see if the answer is hard-coded in the configuration file. Giving up, it contacts a real DNS resolver (e.g., OpenDNS at 18.104.22.168) and sends it a query for “www.rutgers.edu”.
The OpenDNS resolver checks its cache and doesn’t know either, so it contacts one of the root name servers (let’s assume the resolver’s cache is completely empty). It sends a query of “www.cs.rutgers.edu” to the root server 22.214.171.124 (a.root_servers.net).
The root server doesn’t have the answer but it knows the DNS server responsible for the edu domain, so it sends a referral back to the OpenDNS resolver giving it a list of name servers responsible for edu.
The resolver now sends a query of “www.cs.rutgers.edu” to 126.96.36.199, one of the edu name servers (a.edu-servers.net at 188.8.131.52). It does not know the answer either but it does know the name servers for rutgers.edu, so it sends back a referral with a list of those servers.
The resolver now sends a query of “www.cs.rutgers.edu” to 184.108.40.206, one of the rutgers.edu name servers (ns8.a1.incapsecuredns.net). This happens to be an authoritative name server for the rutgers.edu zone and it returns back the address, 220.127.116.11. If cs.rutgers.edu was defined as a separate zone, the rutgers.edu DNS server would send a referral to yet another name server.
The query is now complete and the OpenDNS resolver sends the result back to the stub resolver that requested the query, which sends it back to the client application.
Inside a DNS server
DNS servers store various information about domain names. Each datum is called a resource record. A resource record contains a name, value, type of record, and a time to live value. Common records include:
Address (A record): identifies the IP address for a given host name.
Canonical name (CNAME record): identifies the real host name for an alias. For example, www.cs.rutgers.edu is really a CNAME (alias) to www3.srv.lcsr.rutgers.edu.
Name server (NS record): identifies the authoritative name servers for the domain.
Mail exchanger (MX record): identifies the mail server for a given host name.
DNS uses a simple request-response protocol. Each query message from a client has a corresponding response message from the server. The exact same binary message structure is used for all DNS messages. A flag field identifies whether the message is a query or a response and whether recursion is desired. A variable-length set of fields after the fixed-length message header contains questions (e.g., that you are looking for the A record of www.rutgers.edu) and answers (the responses to the questions).
As we mentioned earlier, DNS resolvers rely on caching to avoid performing the same queries over and over. Every DNS zone contains a time to live value, which is an estimate of how long it is safe for a resolver to keep the results for that zone cached. For example, systems under rutgers.edu have a TTL of 3600 seconds (1 hour) while systems under google.com have a TTL of 900 seconds (15 minutes).
DNS servers are also able to take an IP address as a query and resolve a domain name for the address. Doing this requires a different query path: the edu server has no idea what range of IP addresses were allocated to Rutgers; it just knows the name servers for Rutgers.
A special domain,
in-addr.arpa is created for reverse lookups
(arpa stands for Address & Routing Parameter Area).
The IP address to be queried is written in reverse order, with the first byte last, to construct a name
that looks like 18.104.22.168.in-addr.arpa for the address 22.214.171.124.
An organization has a range, or several ranges, of IP addresses assigned to it. It sets up a local DNS server with PTR (pointer) records that map IP addresses to names. It then tells its ISP what DNS servers are responsible for reverse DNS lookups. The ISP knows what range of addresses belong to the organization. If it gets a query for an address in that range, it now knows which name servers to send on a referral reply. A reverse query that starts at the root will contact the root name servers. These servers, in addition to knowing the name servers of TLDs, also know the name servers for the five RIRs (ARIN, RIPE NCC, etc.) - the entities that hand out IP addresses. The root server may return a referral for the ARIN server (responsible for IP addresses in North America). The ARIN server knows the blocks of IP addresses that were allocated to various ISPs and will send a referral to the name server for the appropriate ISP. That ISP, when queried, will then respond with a referral to the name server for the organization that owns that address.
DNS Terms Glossary
- IANA: Internet Assigned Numbers Authority, the organization in charge of keeping track of IP addresses, port numbers, and other number-related aspects of the Internet
- ICANN: Internet Corporation for Assigned Names and Numbers, the non-profit company that currently runs the IANA.
- RIR: Regional Internet Registry, assigns IP addresses to ISPs within a geographic region.
- TLDs: top-level domains.
- gTLD: generic top-level domain (.com, .edu, .net, …).
- ccTLD: country code top-level domain (.ac, .ae, .ie, .nl, .us).
- IDN: Internationalized Domain Names.
- Domain name registry: the database of registered domain names for a top-level domain.
- Domain name registry operator: the company that keeps the database of domain names under a TLD.
- NIC: network information center, another name for a domain name registry operator.
- Domain name registrar: a company that lets you register a domain name.
- DNS: Domain Name System.
- Canonical name: a name that is an alias for a another domain name.
- Authoritative name server: a name server that stores, and is responsible for, specific DNS records (as opposed to storing a cached copy)
- Zone: a portion of the domain name space (a sub-tree) that is managed by a specific entity. E.g., rutgers.edu is a zone that manages all domains within rutgers.edu.
- A (address) record: a DNS record that stores the IP address corresponding to a specific host name.
- MX (mail exchanger) record: a DNS record that stores the name of the host that handles email for the domain.
- DNS resolver: the client side program that is responsible for contacting DNS servers to complete a DNS query.
- Reverse DNS: querying IP addresses to find the corresponding domain names.
HTTP stands for Hypertext Transfer Protocol and is the web’s application-layer protocol for interacting between web browsers and web servers. It is a TCP, line-oriented, text-based protocol that consists of requests to the server followed by responses from the servers. The protocol is stateless. This means that the server does not store any state from previous requests. This simplifies the design of the protocol, simplifies recovery from crashes, and makes load balancing easier. Note that web application that use HTTP may impose their own state but it is not a part of the HTTP protocol.
Persistent vs. non-persistent connections
HTTP was originally designed to support non-persistent connections. This meant that the connection was alive for only a single request-response interaction. For each new request, the client had to re-establish a connection. That may have been fine in the earliest days of the web but a request for a page is now typically accompanied by multiple successive requests to download supporting files (stylesheet files and images). The overhead of the round-trip time in setting up a connection for each piece of content adds up. HTTP was enhanced to support persistent connections, where a client and server can exchange multiple request-response interactions on the same connection.
Requests and responses
The main function of HTTP is to request objects (content). These are identified in a browser by a URL (Uniform Resource Locator). A URL takes the format:
Browsers support various protocols, not just HTTP. Common ones include HTTP, HTTP (HTTP that is made secure via SSL), FTP (file transfer protocol), and “file” (local files). If the protocol is “http” or “https”, the browser process it via its HTTP protocol module.
The HTTP protocol comprises requests and responses. Each of these messages is structured as a set of text headers, one per line, followed by a blank line, and optionally followed by content. The first line of a request contains a command. The three main HTTP requests are:
GET: request an object
HEAD: like GET but download only the headers for the object.
POST: upload a sequence of name/value pairs to the server. The data is present in the body of the message and is often the response to a form. An alternate way of uploading user data as a set of name/value pairs is to use the GET command and place the data as a set of parameters at the end of the URL. For example,
Each HTTP response contains multiple headers, the first of which contains a status code and corresponding message.
While the HTTP protocol itself does not require keeping state, HTTP provides
a way for web servers to store state about past sessions from the browser.
It does this through cookies.
A cookie is a small amount of data that is associated with the web site.
The data is created by the server when it gets an HTTP request from the client.
It then sends that data back in a
Set-Cookie line in the header of the HTTP response.
Future HTTP requests to the same server will contain a
Cookie line in the header
and contain the data that is associated with the cookie.
This simple mechanism allows a web server to create a database entry indexed by a the cookie data to keep track of a user’s session. That database entry can include things such as shopping cart contents, authentication state, pages visited, time spent on a page, etc. The actual cookie itself does not need to store any of this; it just serves as a unique key into the database table.
Because a web page may contain content from other web sites (hence, other servers), it is possible that requests to those sites will result in the generation of cookies. A first-party cookie is one that comes from the web server that is serving your page request. A third-party cookie is one that comes from another web server that serves some content that is present on the page you originally requested. There has been concern over third party cookies in that they allow these parties, usually advertisers, to track your visits to specific web sites. Most web browsers block third-party cookies by default.
Caching avoids the need to request the same content over and over from the server. However, the challenge is to find out whether the content that is in the cache is still valid. HTTP provides a conditional GET mechanism that is triggered by two lines in the GET header.
When a browser requests content from a server via an HTTP GET message, the
response headers include two lines. One is a
header that contains the timestamp of the last modification time of that
content. The second is a
ETag header that contains a hash
of the content. The client stores both of these values along with the
copy of the content in its cache.
When the content is requested again, the client issues an HTTP GET
request to the server but includes two lines in the headers. One
If-Modified-Since line that contains the last modification
time from the cache and the other is an
line that contains the value from the ETag. This allows the
server to check whether the content has changed since the
version that the client cached. If it did not, the server responds
Not Modified message and no content. If it did,
the server responds just as it would with a regular GET request.
One way to avoid head of line blocking is to have the browser open a separate TCP connection for each HTTP request. There are several downsides to this. Many web pages have many dozens or even hundreds of objects (think of a photo thumbnails gallery, for instance). Opening a large number of connections can take substantial time. Moreover, it can consumer substantial resources at the server since each connection requires kernel and application memory resources as well as CPU time. Because of this, browsers support parallel connections but usually limit them to a small number (typically four). Once you limit the number of connections, you again have the risk of head-of-line blocking. Another problem with pipelining is that there’s no assurance that it will work if a proxy is present. Just because your browser establishes several TCP connections to the proxy does not mean that the proxy will, in turn, establish those connections to the server.
Another performance optimization was HTTP pipelining. With pipelining, instead of waiting for each response, multiple HTTP requests can be dispatched one after another over one connection. However, the server is still obligated to issue responses in the order that the requests were received and head-of-line blocking is still an issue since one delayed or long response can hold up the responses behind it. Most browsers as well as proxies have disabled pipelining or do not implement it.
HTTP/2, the next major update to the HTTP protocol, which came out in 2015, supports the same commands as its predecessor, HTTP/1.1. However, it adds a number of optimizations.
HTTP/2 supports multiplexing. This allows multiple messages to be interleaved on one connection. It is a form of a session layer (implemented in the application, of course). A large response may be broken up into multiple chunks with other responses interleaved among it. The browser keeps track of the pieces and reassembles all the objects.
HTTP/2 Server Push
The HTTP/2 protocol adds a server push capability that allows the server to send objects to the client proactively. The client, upon receiving them, can add them to its cache. This is useful for objects such as stylesheets that are used by an HTML page. Normally, the browser would have to first receive the HTML page so it can parse it before issuing requests for objects that the page needs. If this information is given to the server, it can start sending these objects before the server requests them.
HTTP/2 Header compression
HTTP request and response headers tend to be verbose and are text-based. Their size often requires several round trips just to get the headers for a page out to the server. Compressing headers can make requests and responses shorter and speed up page loads.
FTP, the file transport protocol, is one of the earliest Internet protocols and was designed to transfer files between computers. The protocol uses TCP and is based on commands and responses. A command is a single line of ASCII text. A response is also a single line of text and contains a status code along with a message.
To communicate, a client establishes a TCP connection from some available port N to port 21 on the server. Commands and responses are sent over this communication channel. Some basic commands are USER to identify a user name, PASS to specify the password, GET to download a file, PUT to upload a file, and DIR to get a directory listing.
If the command is a request for data transfer (such as putting a file, getting a file, or getting a directory listing), the server initiates a TCP connection back to the client on port N+1. Data is then transferred over this channel (either from client to server or server to client, depending on the request) and the connection is then closed. FTP is unique compared with most other protocols in that it separates control and data channels. Control information is sent out of band, on a different channel than the data.
Because having a server connect to a client proved problematic in some environments, FTP supports an alternate mechanism, called passive mode, where the client connects to the server to set up the data channel. This is now the more popular mode of operation and some FTP clients, such as web browsers, only support this mode.
The Simple Mail Transfer Protocol (SMTP) is designed for delivering mail to a server that hosts the recipient’s mailbox. It is a TCP-based protocol that is line-based and uses ASCII text for all interactions. An SMTP server acts as a client and a server. Typically a mail application uses SMTP to send a message to a user’s SMTP server (e.g., smtp.gmail.com). This server is often hosted by the organization that provide’s the sender’s email service (e.g., Google, Comcast, Rutgers). This SMTP server then queues the message for delivery. To deliver the message, it acts like a client. The SMTP server looks up the DNS MX (mail exchanger) record for the destination domain, connects to that SMTP server, and delivers the message. The receiving server places the message in the user’s mailbox. If the user has an account on that machine and runs the mail client locally, the mail client can access the mailbox and read the message. More often, the user is on a different system and needs to fetch messages. For that, mail retrieval protocols, such as POP or IMAP, must be used.
The SMTP protocol consists of server identification (
specifying who the mail is from (
MAIL FROM:), and then specifying
one or more recipients, one per line (
RCPT TO:). Finally, the
message is send with the
DATA command. The message is multiple lines
of ASCII text and typically starts with the mail headers that you see in your email.
It is useful to note that all those mail headers are of no value to SMTP; it just
treats them as the message data. You can have a completely different list of names
To: header than you specified in the SMTP
commands and the mail will only be delivered to the recipients you listed
RCPT TO commands.
SMTP is an example of a push protocol. The client takes content and sends it to the server. HTTP, on the other hand, is a pull protocol. The client connects to it and asks it for content.
Because SMTP was designed to handle only text-based interaction, sending mail containing binary data, such as a jpeg file, was problematic. To remedy this, an encoding format called MIME (Multipurpose Internet Mail Extensions) was created. This defines formats for encoding content in a suitable format for message delivery. A MIME header in the body of the email identifies the content type and encoding used. To support mail attachments and the encoding of multiple objects, multipart MIME headers in the message body allow one to identify multiple chunks of content. MIME has nothing to do with SMTP but is designed to cope with the restrictions that SMTP placed on the structure of a message (7-bit ASCII text with line breaks). It is up to mail clients to create and parse MIME encodings.
SMTP dealt only with mail delivery.
POP3 is a TCP-based protocol to allow a user to connect to a remote mailbox, download, and delete messages.
The entire protocol is text-based. A user authenticates with
commands and then sends commands to list messages, retrieve a specific message, or delete a message.
POP3 supports two interaction models. The download-and-delete model has a client connect to a mail server, download messages to the client’s local mailbox, and then delete them from the server. With this model, the server is just a temporary repository for mail until the client gets around to downloading it. The problem with this model is that it does not work if you access mail from multiple devices. Once a message is deleted from the server, other devices cannot get it.
The download-and-keep model has the client connect to a mail server, download messages to the client’s local mailbox, but does not delete them from the server. They only get deleted when a user deletes them locally and the mail client connects back to the server and issues a POP3 delete command for those messages. With this behavior, a user can access messages from multiple devices.
The downside of POP3 is that it does not keep state across sessions. It does not know, for example, if a user marked several messages for deletion during a previous connection session.
IMAP, the Internet Message Access Protocol was designed to operate on a mailbox remotely rather than POP’s approach of retrieving the contents of a mailbox onto a client. It can handle the case where multiple clients are accessing the same mailbox and can keep operations synchronized since state is maintained on the server.
IMAP also supports the ability to move messages into folders, search for specific messages on the server, mark messages for deletion prior to actually deleting them, and fetch headers or full messages. It allows the same offline convenience that POP does, where all content can be downloaded onto a client, but also offers full state tracking on the server.
Like POP, SMTP, and HTTP, IMAP commands are also sent as lines of ASCII text. Unlike those protocols, requests and responses can be handled asynchronously; a client can send multiple requests without first waiting for responses.
Peer to Peer Protocols
Traditional, and still the most common, network-based applications are those that follow a client-server model. A client needs a service (access to a file’s contents, for example) and contacts a server that can provide that service. A peer-to-peer model is an alternative application architecture that removes the need for dedicated servers and enables each host to participate in providing the service. Because all machines can both access as well as provide the service, they are called peers.
A true peer-to-peer architecture has no reliance on a central server. In practice, some peer-to-peer architectures are really hybrid architectures, where a central server may provide key authentication or location services. Desirable (but not necessary) characteristics of peer-to-peer application architectures are robustness and self-scalability. Robustness refers to the ability of the overall service to run even if some systems may be down. Self-scalability refers to the ability of the system to handle greater workloads as more peers are introduced into the system.
In our discussions, we focused on just one application domain: peer-to-peer file distribution.
For file distribution, there are four key operations (primitives): (1) how a peer joins and leaves a peer-to-peer system; (2) how peers register files and their metadata (names, attributes); (3) how search is handled; and (4) how files are downloaded.
The systems that we examine may or may not tackle all of these areas.
Napster is the earliest of peer-to-peer systems and is the system that put peer-to-peer file sharing on the map. It was built for sharing MP3 files. Napster is not a pure peer-to-peer architecture since it relies on a single server to keep track of which peer has which content.
A peer contacts the central server and publishes a list of files that it wants to share. Anyone who wants to find a file contacts the central server to get a list of peers that have the file. The peer then connects to any of the peers in that list and downloads the file. The download is either via a direct TCP connection to the server or, if the system is inaccessible because it is behind a firewall, it contacts the central server to send a message to the desired peer requesting that it connect and upload to the requestor.
The advantage of Napster is that it is a simple design. The use of a central server, while deviating from a true peer-to-peer model, establishes a single point of control and maintains all the information on the locations of content.
The downside is that the server can become a bottleneck with high query volumes. The failure of the central server causes the entire system to cease to operate.
After Napster was shut down by shutting down its central server, Gnutella set out to create an architecture that offers truly distributed file sharing. Unlike Napster, Gnutella can not be shut down since there is no central server.
Gnutella’s approach to finding content is based on query flooding. When a peer joins the system, it needs to contact at least one other Gnutella node and ask it for a list of nodes it knows about (its “friends”). This list of peers becomes its list of connected nodes. This builds an overlay network. An overlay network is a logical network that is formed by peer connections. Each peer knows of a limited set of other peers. These become its neighbors, and do not need to be physical neighbors. A peer is capable of communicating with any other peer; it is just the lack of knowing that the other peer exists that stops it.
To search for content, a peer sends a query message to its connected nodes. Each node that receives a query will respond if it has the content. Otherwise, it forwards the content to its connected nodes. This is the process of flooding. Once the content is found, the requesting peer downloads the content from the peer hosting the content via HTTP.
A facet of the original design of Gnutella was anonymity. Replies were sent replies through the same path that the queries took. A peer receiving a query would not know if it came from the requestor or from a peer just forwarding the request.
Gnutella has a significant architectural advantage over Napster. Its design is fully decentralized. There is no central directory and hence the service cannot be shut down. On the other hand, flooding-based search is inefficient compared to maintaining a single database. Search may require contacting a large number of systems and going through multiple hops. Well-known nodes (e.g., those that may be configured in default installations) may become overly congested.
A few optimizations were later added to Gnutella.
The process of routing replies through the query path was changed to sending responses directly to the requester to reduce response times.
If connecting to a peer that serves the content is not possible because of firewall restrictions at the peer, the requesting node can send a push request, asking the serving peer to send it the file.
Much of the Gnutella network was composed of end user’s personal machines and these had varying levels of uptime and connectivity. As such, not all peers are equal. With this in mind, Gnutella divided its peers into two categories: leaf nodes and ultrapeers. Leaf nodes are normal peers. They know of a small number of ultrapeers and may not have fast connections. Ultrapeers are peers that have a high degree of connectivity (32 or more connections to other ultrapeers) and can hence flood queries with more hops.
Kazaa was created a year after Gnutella with the core premise that not all nodes have equivalent capabilities as far as network connectivity and uptime are concerned. They introduced the concept of supernodes. These nodes have high uptime, fast connectivity, faster processors, and potentially more storage than regular nodes. They also know other supernodes. This is the same concept as Gnutella’s later enhancement with its addition of ultrapeers. A client (peer) needs to know of one supernode to join the system. It sends that supernode a list of all the files that it is hosting. Only supernodes are involved in the search process. Search is a flood over the overlay network as in Gnutella. Once a query reaches a supernode that has the requested content in its list, it sends a reply directly to the peer that initiated the query. The querying peer will then download the content from the peer that hosts the content.
The design of BitTorrent was motivated by the flash crowd problem. How do you design a file sharing service that will scale as a huge number of users want to download a specific file? Systems such as Napster, Gnutella, and Kazaa all serve their content from the peer that hosts it. If a large number of users try to download a popular file, all of them will have to share the bandwidth that is available to the peer hosting that content.
The idea behind BitTorrent is to turn a peer that is downloading content into a server of that content. The more peers are downloading content, the more servers there will be for it. BitTorrent only focuses on the download problem and does not handle the mechanism for locating the content.
To offer content, the content owner creates a .torrent file. This file contains metadata, or information, about the file, such as the name, creation time, and size of the file. It also contains a list of hashes of blocks of the content. The content is logically divided into fixed-size blocks and the list of hashes in the .torrent file allows a downloading peer to validate that any downloaded blocks has been downloaded correctly. Finally, the .torrent file contains a list of trackers.
The tracker is a server running a process that manages downloads for a set of .torrent files. When a downloading peer opens a .torrent file, it contacts a tracker that is specified in that file. The tracker is responsible for keeping track of which peers have which have the content. There could be many trackers, each responsible for different torrents.
A seeder is a peer that has the entire file available for download by other peers. Seeders register themselves with trackers so that trackers can direct downloading peers to them. An initial seeder is the initial version of the file.
A leecher is a peer that is downloading files. To start the download, the leecher must have a .torrent file. That identifies the tracker for the contents. It contacts the tracker, which keeps track of the seed nodes for that file as well as other leechers, some of whom may have already downloaded some blocks of the file. A leecher contacts seeders and other leechers to download random blocks of the file. As it gets these blocks, it can make them available to other leechers. This is what allows download bandwidth to scale: every downloader increases overall download capacity. Once a file is fully downloaded, the leecher has the option of turning itself into a seeder and continue to offer serving the file.
BitTorrent scales very well. The more participants there are, the greater the aggregate bandwidth is. Peers may be given an incentive to share since BitTorrent software may choose to block downloads if you don’t offer uploads. The downside of BitTorrent is that unpopular files will not have leechers and will not offer this benefit of scale. Block sizes tend to be large (the default is often 256 KB with a maximum size of 4 MB). This makes the architecture not suitable for small files as the distributed download aspect won’t come into play unless a large number of leechers choose to act as future seeders. Finally, search is not a part of the protocol. A user needs to turn to some other mechanism to actually get the .torrent file.
Distributed Hash Tables
The systems we covered use one of three approaches for locating content:
- Central server (Napster)
- Flood (Gnutella, Kazaa)
- Nothing (BitTorrent). Search is out of scope for BitTorrent and it relies on separate solutions to allow users to locate a .torrent file for the desired content.
Flooding can be an inefficient and indeterminate procedure for finding content. Some nodes may be slower than others and some may have fewer connections than others, resulting in more hops to query the same number of machines. Gnutella and Kazaa tried to ameliorate this somewhat by creating ultrapeers (supernodes) but the mechanism of the flood still exists.
In standalone systems, hash tables are attractive solutions for high-speed lookup tables. A hash function is applied to a search key. That result becomes an index into a table. Hash tables result in O(1) lookup performance versus the O(log N) time for a binary tree or search through a sorted table. Since there is a chance that multiple keys hash to the same value (known as a collision), each table entry, called a slot (or bucket), may contain a linked list or additional hash table.
A distributed hash table, or DHT, is a peer-to-peer version of a hash table: a distributed key, value database. The interface we want for a DHT is that a client will query a DHT server with a key to get the corresponding value. This DHT server may be a separate collection of peer-to-peer systems, all acting as one server from the client’s point of view or the querying client may also be a peer. The DHT software finds the host that holds the key, value pair and returns the corresponding value to the querying host. This should be done without the inefficiency of a flood. The specific implementation of a DHT that we examine is called Chord and it creates an overlay network that is a logical ring of peers.
Chord takes a large hash of a key (e.g., a 160-bit SHA–1 hash). Each node in the system is assigned a position in the ring by hashing its IP address. Because the vast majority of bucket positions will be empty, key, value data is stored either at the node to which the key hashes (if, by some chance, the key hashes to the same value that the node’s IP address hashed) or on a successor node, the next node that would be encountered as the ring is traversed clockwise. For a simple example, let us suppose that we have a 4-bit hash (0..15) and nodes occupying positions 2 and 7. If a key hashes to 4, the successor node is 7 and hence the machine at node 7 will be responsible for storing all data for keys that hash to 4. It is also responsible for storing all data to keys that hash to 3, 5, 6, and 7.
If a node only knows of its clockwise neighbor node, then any query that a node cannot handle will be forwarded to a neighboring node. This results in an unremarkable O(n) lookup time for a system with n nodes. An alternate, faster, approach is to have each node keep a list of all the other nodes in the group. This way, any node will be able to find out out which node is responsible for the data on a key simply by hashing the key and traversing the list to find the first node ≥ the hash of the key. This gives us an impressive O(1) performance at the cost of having to maintain a full table of all the nodes in the system on each node. A compromise approach to have a bounded table size is to use finger tables. A finger table is a partial list of nodes with each node in the table being a factor of two away from the current node. Element 0 of the table is the next node (20 = 1 away), element 1 of the table is the node after that (21 = 2 away), element 2 of the table four nodes removed (22), element 3 of the table eight nodes removed (23), and so on. With finger tables, O(log n) nodes need to be contacted to find the owner of a key.