This paper was originally submitted to the InterNIC and posted on their Gopher site on 5 August 1994. This document is a continually updated version of that paper.
Contents
1. Introduction 2. What are TCP/IP and the Internet?
2.1. The Evolution of TCP/IP (and the Internet) 2.2. Internet Growth 2.3. Internet Administration 2.4. Domain Names and IP Addresses (and Politics)
3. The TCP/IP Protocol Architecture
3.1. The Network Interface Layer
3.1.1. PPP
3.2. The Internet Layer
3.2.1. IP Addressing and Subnet Masks 3.2.2. Conserving IP Addresses: CIDR, DHCP, NAT, and PAT 3.2.3. The Domain Name System 3.2.4. ARP and Address Resolution 3.2.5. IP Routing: OSPF, RIP, and BGP 3.2.6. IP version 6
3.4.1. TCP and UDP Applications 3.4.2. Protocol Analysis
3.5. Summary
4. Other Information Sources 5. Acronyms and Abbreviations 6. Author‘s Address
1. Introduction
An increasing number of people are using the Internet and, many for the first time, are using the tools and utilities that at one time were only available on a limited number of computer systems (and only for really intense users!). One sign of this growth in use has been the significant number of TCP/IP and Internet books, articles, courses, and even TV shows that have become available in the last several years; there are so many such books that publishers are reluctant to authorize more because bookstores have reached their limit of shelf space! This memo provides a broad overview of the Internet and TCP/IP, with an emphasis on history, terms, and concepts. It is meant as a brief guide and starting point, referring to many other sources for more detailed information.
2. What are TCP/IP and the Internet?
While the TCP/IP protocols and the Internet are different, their histories are most definitely intertwingled! This section will discuss some of the history. For additional information and insight, readers are urged to read two excellent histories of the Internet: Casting The Net: From ARPANET to INTERNET and beyond... by Peter Salus (Addison-Wesley, 1995) and Where Wizards Stay Up Late: The Origins of the Internet by Katie Hafner and Mark Lyon (Simon & Schuster, 1997). In addition, the Internet Society maintains a number of on-line "Internet history" papers at http://www.isoc.org/internet/history/.
2.1. The Evolution of TCP/IP (and the Internet)
While the Internet today is recognized as a network that is fundamentally changing social, political, and economic structures, and in many ways obviating geographic boundaries, this potential is merely the realization of predictions that go back nearly forty years. In a series of memos dating back to August 1962, J.C.R. Licklider of MIT discussed his "Galactic Network" and how social interactions could be enabled through networking. The Internet certainly provides such a national and global infrastructure and, in fact, interplanetary Internet communication has already been seriously discussed.
Prior to the 1960s, what little computer communication existed comprised simple text and binary data, carried by the most common telecommunications network technology of the day; namely, circuit switching, the technology of the telephone networks for nearly a hundred years. Because most data traffic is bursty in nature (i.e., most of the transmissions occur during a very short period of time), circuit switching results in highly inefficient use of network resources.
The fundamental technology that makes the Internet work is called packet switching, a data network in which all components (i.e., hosts and switches) operate independently, eliminating single point-of-failure problems. In addition, network communication resources appear to be dedicated to individual users but, in fact, statistical multiplexing and an upper limit on the size of a transmitted entity result in fast, economical networks.
In the 1960s, packet switching was ready to be discovered. In 1961, Leonard Kleinrock of MIT published the first paper on packet switching theory (and the first book on the subject in 1964). In 1962, Paul Baran of the Rand Corporation described a robust, efficient, store-and-forward data network in a report for the U.S. Air Force. At about the same time, Donald Davies and Roger Scantlebury suggested a similar idea from work at the National Physical Laboratory (NPL) in the U.K. The research at MIT (1961-1967), RAND (1962-1965), and NPL (1964-1967) occurred independently and the principal researchers did not all meet together until the Association for Computing Machinery (ACM) meeting in 1967. The term packet was adopted from the work at NPL.
The modern Internet began as a U.S. Department of Defense (DoD) funded experiment to interconnect DoD-funded research sites in the U.S. The 1967 ACM meeting was also where the initial design for the so-called ARPANET — named for the DoD‘s Advanced Research Projects Agency (ARPA) — was first published by Larry Roberts. In December 1968, ARPA awarded a contract to Bolt Beranek and Newman (BBN) to design and deploy a packet switching network with a proposed line speed of 50 kbps. In September 1969, the first node of the ARPANET was installed at the University of California at Los Angeles (UCLA), followed monthly with nodes at Stanford Research Institute (SRI), the University of California at Santa Barbara (UCSB), and the University of Utah. With four nodes by the end of 1969, the ARPANET spanned the continental U.S. by 1971 and had connections to Europe by 1973.
The original ARPANET gave life to a number of protocols that were new to packet switching. One of the most lasting results of the ARPANET was the development of a user-network protocol that has become the standard interface between users and packet switched networks; namely, ITU-T (formerly CCITT) Recommendation X.25. This "standard" interface encouraged BBN to start Telenet, a commercial packet-switched data service, in 1974; after much renaming, Telenet became a part of Sprint‘s X.25 service.
The initial host-to-host communications protocol introduced in the ARPANET was called the Network Control Protocol (NCP). Over time, however, NCP proved to be incapable of keeping up with the growing network traffic load. In 1974, a new, more robust suite of communications protocols was proposed and implemented throughout the ARPANET, based upon the Transmission Control Protocol (TCP) for end-to-end network communication. But it seemed like overkill for the intermediate gateways (what we would today call routers) to needlessly have to deal with an end-to-end protocol so in 1978 a new design split responsibilities between a pair of protocols; the new Internet Protocol (IP) for routing packets and device-to-device communication (i.e., host-to-gateway or gateway-to-gateway) and TCP for reliable, end-to-end host communication. Since TCP and IP were originally envisioned functionally as a single protocol, the protocol suite, which actually refers to a large collection of protocols and applications, is usually referred to simply as TCP/IP.
The original versions of both TCP and IP that are in common use today were written in September 1981, although both have had several modifications applied to them (in addition, the IP version 6, or IPv6, specification was released in December 1995). In 1983, the DoD mandated that all of their computer systems would use the TCP/IP protocol suite for long-haul communications, further enhancing the scope and importance of the ARPANET.
In 1983, the ARPANET was split into two components. One component, still called ARPANET, was used to interconnect research/development and academic sites; the other, called MILNET, was used to carry military traffic and became part of the Defense Data Network. That year also saw a huge boost in the popularity of TCP/IP with its inclusion in the communications kernel for the University of California s UNIX implementation, 4.2BSD (Berkeley Software Distribution) UNIX.
In 1986, the National Science Foundation (NSF) built a backbone network to interconnect four NSF-funded regional supercomputer centers and the National Center for Atmospheric Research (NCAR). This network, dubbed the NSFNET, was originally intended as a backbone for other networks, not as an interconnection mechanism for individual systems. Furthermore, the "Appropriate Use Policy" defined by the NSF limited traffic to non-commercial use. The NSFNET continued to grow and provide connectivity between both NSF-funded and non-NSF regional networks, eventually becoming the backbone that we know today as the Internet. Although early NSFNET applications were largely multiprotocol in nature, TCP/IP was employed for interconnectivity (with the ultimate goal of migration to Open Systems Interconnection).
The NSFNET originally comprised 56-kbps links and was completely upgraded to T1 (1.544 Mbps) links in 1989. Migration to a "professionally-managed" network was supervised by a consortium comprising Merit (a Michigan state regional network headquartered at the University of Michigan), IBM, and MCI. Advanced Network & Services, Inc. (ANS), a non-profit company formed by IBM and MCI, was responsible for managing the NSFNET and supervising the transition of the NSFNET backbone to T3 (44.736 Mbps) rates by the end of 1991. During this period of time, the NSF also funded a number of regional Internet service providers (ISPs) to provide local connection points for educational institutions and NSF-funded sites.
In 1993, the NSF decided that it did not want to be in the business of running and funding networks, but wanted instead to go back to the funding of research in the areas of supercomputing and high-speed communications. In addition, there was increased pressure to commercialize the Internet; in 1989, a trial gateway connected MCI, CompuServe, and Internet mail services, and commercial users were now finding out about all of the capabilities of the Internet that once belonged exclusively to academic and hard-core users! In 1991, the Commercial Internet Exchange (CIX) Association was formed by General Atomics, Performance Systems International (PSI), and UUNET Technologies to promote and provide a commercial Internet backbone service. Nevertheless, there remained intense pressure from non-NSF ISPs to open the network to all users.
FIGURE 1. NSFNET structure initiated in 1994 to merge the academic and commercial networks.
In 1994, a plan was put in place to reduce the NSF‘s role in the public Internet. The new structure comprises three parts:
Network Access Points (NAPs), where individual ISPs would interconnect, as suggested in Figure 1. The NSF originally funded four such NAPs: Chicago (operated by Ameritech), New York (really Pensauken, NJ, operated by Sprint), San Francisco (operated by Pacific Bell, now SBC), and Washington, D.C. (MAE-East, operated by MFS, now part of Worldcom).
The very High Speed Backbone Network Service, a network interconnecting the NAPs and NSF-funded centers, operated by MCI. This network was installed in 1995 and operated at OC-3 (155.52 Mbps); it was completely upgraded to OC-12 (622.08 Mbps) in 1997.
The Routing Arbiter, to ensure adequate routing protocols for the Internet.
In addition, NSF-funded ISPs were given five years of reduced funding to become commercially self-sufficient. This funding ended by 1998 and a proliferation of additional NAPs have created a "melting pot" of services. Today‘s terminology refers to three tiers of ISP:
Tier 1 refers to national ISPs, or those that have a national presence and connect to at least three of the original four NAPs. National ISPs include AT&T, Sprint, and Worldcom.
Tier 2 refers to regional ISPs, or those that have primarily a regional presence and connect to less than three of the original four NAPs. Regional ISPs include Adelphia, BellAtlantic.net, and BellSouth.net.
Tier 3 refers to local ISPs, or those that do not connect to a NAP but offer services via an upstream ISP.
It is worth saying a few words about the NAPs. The NSF provided major funding for the four NAPs mentioned above but they needed to have additional customers to remain economically viable. Some companies — such as then-Metropolitan Fiber Systems (MFS) — decided to build other NAP sites. One of MFS‘ first sites was MAE-East, where "MAE" stood for "Metropolitan Area Ethernet." MAE-East was merely a point where ISPs could interconnect which they did by buying a router and placing it at the MAE-East facility. The original MAE-East provided a 10 Mbps Ethernet LAN to interconnect the ISPs‘ routers, hence the name. The Ethernet LAN was eventually replaced with a 100 Mbps FDDI ring and the "E" then became "Exchange." Over the years, MFS/MCI Worldcom has added sites in San Jose, CA (MAE-West), Los Angeles, Dallas, and Houston.
Other companies also operate their own NAPs. Savvis, for example, operates an international Internet service and has built more than a dozen private NAPs in North America. Many large service providers go around the NAPs entirely by creating bilateral agreement whereby the directly route traffic coming from one network and going to the other. Before their merger in 1998, for example, MCI and LDDS Worldcom had more than 10 DS-3 (44.736 Mbps) lines interconnecting the two networks.
The North American Network Operators Group (NANOG) provides a forum for the exchange of technical information and the discussion of implementation issues that require coordination among network service providers. Meeting three times a year, NANOG is an essential element in maintaining stable Internet services in North America. Initially funded by the NSF, NANOG currently receives funds from conference registration fees and vendor donations.
In 1988, meanwhile, the DoD and most of the U.S. Government chose to adopt OSI protocols. TCP/IP was now viewed as an interim, proprietary solution since it ran only on limited hardware platforms and OSI products were only a couple of years away. The DoD mandated that all computer communications products would have to use OSI protocols by August 1990 and use of TCP/IP would be phased out. Subsequently, the U.S. Government OSI Profile (GOSIP) defined the set of protocols that would have to be supported by products sold to the federal government and TCP/IP was not included.
Despite this mandate, development of TCP/IP continued during the late 1980s as the Internet grew. TCP/IP development had always been carried out in an open environment (although the size of this open community was small due to the small number of ARPA/NSF sites), based upon the creed "We reject kings, presidents, and voting. We believe in rough consensus and running code" [Dave Clark, M.I.T.]. OSI products were still a couple of years away while TCP/IP became, in the minds of many, the real open systems interconnection protocol suite.
It is not the purpose of this memo to take a position in the OSI vs. TCP/IP debate (although it is absolutely clear that TCP/IP offers the primary goals of OSI; namely, a universal, non-proprietary data communications protocol. In fact, TCP/IP does far more than was ever envisioned for OSI — or for TCP/IP itself, for that matter). But before TCP/IP prevailed and OSI sort of dwindled into nothingness, many efforts were made to bring the two communities together. The ISO Development Environment (ISODE) was developed in 1990, for example, to provide an approach for OSI migration for the DoD. ISODE software allows OSI applications to operate over TCP/IP. During this same period, the Internet and OSI communities started to work together to bring about the best of both worlds as many TCP and IP features started to migrate into OSI protocols, particularly the OSI Transport Protocol class 4 (TP4) and the Connectionless Network Layer Protocol (CLNP), respectively. Finally, a report from the National Institute for Standards and Technology (NIST) in 1994 suggested that GOSIP should incorporate TCP/IP and drop the "OSI-only" requirement. [NOTE: Some industry observers have pointed out that OSI represents the ultimate example of a sliding window; OSI protocols have been "two years away" since about 1986.]
None of this is meant to suggest that the NSF isn‘t funding Internet-class research networks anymore. That is just the function of http://www.internet2.edu/, a consortium of nearly 200 universities working in partnership with industry and government to develop and deploy advanced network applications and technologies for the next generation Internet. Goals of Internet2 are to create a leading edge network capability for the national research community, enable the development of new Internet-based applications, and to quickly move these new network services and applications to the commercial sector.
2.2. Internet Growth
In Douglas Adams‘ The Hitchhiker‘s Guide to the Galaxy (Pocket Books, 1979), the hitchhiker describes outer space as being "...big. Really big. ...vastly hugely mind-bogglingly big..." A similar description can be applied to the Internet. To paraphrase the hitchhiker, you may think that your 750 node LAN is big, but that‘s just peanuts compared to the Internet.
The ARPANET started with four nodes in 1969 and grew to just under 600 nodes before it was split in 1983. The NSFNET also started with a modest number of sites in 1986. After that, the network experienced literally exponential growth. Internet growth between 1981 and 1991 is documented in "Internet Growth (1981-1991)" (RFC 1296).
The Internet Software Consortium hosts the Internet Domain Survey (with technical support from Network Wizards, who originated the survey). According to their chart, the Internet had nearly 30 million reachable hosts by January 1998 and over 56 million by July 1999. Dedicated residential access methods, such as cable modem and asymmetrical digital subscriber line (ADSL) technologies, are undoubtedly the reason that this nunber has shot up to over 171 million by January 2003. During the boom-1990s, the Internet was growing at a rate of about a new network attachment every half-hour, interconnecting hundreds of thousands of networks. It was estimated that the Internet was doubling in size every ten to twelve months and traffic was doubling every 100 days (for 1000% annual growth). For the last several year, the number of nodes has been growing at a rate of about 50% annually and taffic continues to keep pace with that growth.
And what of the original ARPANET? It grew smaller and smaller during the late 1980s as sites and traffic moved to the Internet, and was decommissioned in July 1990. Cerf & Kahn ("Selected ARPANET Maps," Computer Communications Review, October 1990) re-printed a number of network maps documenting the growth (and demise) of the ARPANET.
2.3. Internet Administration
The Internet has no single owner, yet everyone owns (a portion of) the Internet. The Internet has no central operator, yet everyone operates (a portion of) the Internet. The Internet has been compared to anarchy, but some claim that it is not nearly that well organized!
Some central authority is required for the Internet, however, to manage those things that can only be managed centrally, such as addressing, naming, protocol development, standardization, etc. Among the significant Internet authorities are:
The Internet Society (ISOC), chartered in 1992, is a non-governmental international organization providing coordination for the Internet, and its internetworking technologies and applications. ISOC also provides oversight and communications for the Internet Activities Board.
The Internet Activities Board (IAB) governs administrative and technical activities on the Internet.
The Internet Engineering Task Force (IETF) is one of the two primary bodies of the IAB. The IETF‘s working groups have primary responsibility for the technical activities of the Internet, including writing specifications and protocols. The impact of these specifications is significant enough that ISO accredited the IETF as an international standards body at the end of 1994. RFCs 2028 and 2031 describe the organizations involved in the IETF standards process and the relationship between the IETF and ISOC, respectively, while RFC 2418 describes the IETF working group guidelines and procedures. The background and history of the IETF and the Internet standards process can be found in "IETF—History, Background, and Role in Today‘s Internet."
The Internet Engineering Steering Group (IESG) is the other body of the IAB. The IESG provides direction to the IETF.
The Internet Research Task Force (IRTF) comprises a number of long-term reassert groups, promoting research of importance to the evolution of the future Internet.
The Internet Engineering Planning Group (IEPG) coordinates worldwide Internet operations. This group also assists Internet Service Providers (ISPs) to interoperate within the global Internet.
The Forum of Incident Response and Security Teams is the coordinator of a number of Computer Emergency Response Teams (CERTs) representing many countries, governmental agencies, and ISPs throughout the world. Internet network security is greatly enhanced and facilitated by the FIRST member organizations.
The World Wide Web Consortium (W3C) is not an Internet administrative body, per se, but since October 1994 has taken a lead role in developing common protocols for the World Wide Web to promote its evolution and ensure its interoperability. W3C has more than 400 Member organizations internationally. The W3C, then, is leading the technical evolution of the Web, having already developed more than 20 technical specifications for the Web‘s infrastructure.
2.4. Domain Names and IP Addresses (and Politics)
Although not directly related to the administration of the Internet for operational purposes, the assignment of Internet domain names (and IP addresses) is the subject of some controversy and a lot of current activity. Internet hosts use a hierarchical naming structure comprising a top-level domain (TLD), domain and subdomain (optional), and host name. The IP address space, and all TCP/IP-related numbers, have historically been managed by the Internet Assigned Numbers Authority (IANA). Domain names are assigned by the TLD naming authority; until April 1998, the Internet Network Information Center (InterNIC) had overall authority of these names, with NICs around the world handling non-U.S. domains. The InterNIC was also responsible for the overall coordination and management of the Domain Name System (DNS), the distributed database that reconciles host names and IP addresses on the Internet.
The InterNIC is an interesting example of the recent changes in the Internet. Since early 1993, Network Solutions, Inc. (NSI) operated the registry tasks of the InterNIC on behalf of the NSF and had exclusive registration authority for the .com, .org, .net, and .edu domains. NSI‘s contract ran out in April 1998 and was extended several times because no other agency was in place to continue the registration for those domains. In October 1998, it was decided that NSI would remain the sole administrator for those domains but that a plan needed to be put into place so that users could register names in those domains with other firms. In addition, NSI‘s contract was extended to September 2000, although the registration business was opened to competition in June 1999. Nevertheless, when NSI‘s original InterNIC contract expired, IP address assignments moved to a new entity called the American Registry for Internet Numbers (ARIN). (And NSI itself was purchased by VeriSign in March 2000.)
The newest body to handle governance of global Top Level Domain (gTLD) registrations is the Internet Corporation for Assigned Names and Numbers (ICANN). Formed in October 1998, ICANN is the organization designated by the U.S. National Telecommunications and Information Administration (NTIA) to administer the DNS. Although surrounded in some early controversy (which is well beyond the scope of this paper!), ICANN has received wide industry support. ICANN has created several Support Organizations (SOs) to create policy for the administration of its areas of responsibility, including domain names (DNSO), IP addresses (ASO), and protocol parameter assignments (PSO).
On April 21, 1999, ICANN announced that five companies had been selected to be part of this new competitive Shared Registry System for the .com, .net, and .org domains:
America Online, Inc. (U.S.)
CORE (Internet Council of Registrars) (International)
France Telecom/Oléane (France)
Melbourne IT (Australia)
register.com (U.S.)
Phase I of the competitive registrar testbed program was scheduled to run until June 1999, although that date was subsequently extended to August; at the end of Phase I, the Shared Registry System for the .com, .net, and .org domains was to be opened to all ICANN-accredited registrars. By the end of 1999, ICANN had added an additional 29 registrars and more have been added so that there are about 100 different registrars today. Definitive ICANN registrar accreditation information can be found at ICANN‘s Web site.
The hierarchical structure of domain names is best understood if the domain name is read from right-to-left. Internet hosts names end with a top-level domain name. World-wide generic top-level domains (TLDs) include:
.com: Commercial organizations (administered by VeriSign Global Registry Services through the Shared Registry System)
.edu: Educational institutions; largely limited to 4-year colleges and universities from about 1994 to 2001, but also includes some community colleges (administered by EDUCAUSE)
.net: Network providers; laregely limited to hosts actually part of an operational network from about 1994 to 2001 but now open to anyone, including the author of this paper! (administered by VeriSign Global Registry Services through the Shared Registry System)
.org: Non-profit organizations (administered by VeriSign; after January 2003, will be administered by the Public Interest Registry (PIR), an organization formed by ISOC with operational control subcontracted to Afilias, the operator of the .info domain)
.int: Organizations established by international treaty
.gov: U.S. Federal government agencies (managed by the U.S. General Services Administration, including the fed.us domain)
.mil: U.S. military (managed by the U.S. Department of Defense Network Information Center)
The host name poodle.champlain.edu, for example, is assigned to a computer named poodle (don‘t ask why...) in the Accounting & Computing Systems Division at Champlain College (champlain), within the educational TLD (edu). The host name mail.sover.net refers to a host (mail) in the SoverNet domain (sover) within the network provider TLD (net). Guidelines for selecting host names is the subject of RFC 1178.
Other top-level domain names use the two-letter country codes defined in ISO standard 3166; munnari.oz.au, for example, is the address of the Internet gateway to Australia and myo.inst.keio.ac.jp is a host at the Science and Technology Department of Keio University in Yokohama, Japan. Other ISO 3166-based domain country codes are ca (Canada), de (Germany), es (Spain), fr (France), gb (Great Britain) [NOTE: For some historical reasons, the TLD .gb is rarely used; the TLD .uk (United Kingdom) seems to be preferred although UK is not an official ISO 3166 country code.], ie (Ireland), il (Israel), mx (Mexico), and us (United States). It is important to note that there is not necessarily any correlation between a country code and where a host is actually physically located.
There are several registries responsible for blocks of IP addresses and domain naming policies around the globe. The American Registry for Internet Numbers (ARIN), was originally responsible for the Americas (western hemisphere) and parts of Africa. In 2002, the Latin American and Caribbean Internet Addresses Registry (LACNIC) was officially recognized and now covers Central and South America, as well as some Caribbean nations. The African Regional Internet Registry (AfriNIC), still on a provisional status, will be assuming responsibility for sub-Sahara Africa. Eventually, ARIN will only cover North America and parts of the Caribbean. The European and Asia-Pacific naming registries are managed by R閟eaux IP Europ閑n (RIPE) and the Asia-Pacific NIC (APNIC), respectively.
These authorities, in turn, delegate most of the country TLDs to national registries (such as RNP in Brazil and NIC-Mexico), which have ultimate authority to assign local domain names. An excellent overview of the recent history and anticipated future of the registry system can be found in "Development of the Regional Internet Registry System" (D. Karrenberg et al.) in the IP Journal, Vol. 4, No. 4.
Different countries may organize the country-based subdomains in any way that they want. Many countries use a subdomain similar to the TLDs, so that .com.mx and .edu.mx are the suffixes for commercial and educational institutions in Mexico, and .co.uk and .ac.uk are the suffixes for commercial and educational institutions in the United Kingdom.
The us domain is largely organized on the basis of geography or function. Geographical names in the us name space use names of the form entity-name.city-telegraph-code.state-postal-code.us. The domain name cnri.reston.va.us, for example, refers to the Corporation for National Research Initiatives in Reston, Virginia. Functional branches are also reserved within the name space for schools (K12), community colleges (CC), technical schools (TEC), state government agencies (STATE), councils of governments (COG), libraries (LIB), museums (MUS), and several other generic types of entities. Domain names in the state government name space usually take the form department.state.state-postal-code.us (e.g., the domain name dps.state.vt.us points to the Vermont Department of Public Safety). The K12 name space can vary widely, usually using the form school.school-district.k12.state-postal-code.us (e.g., the domain ccs.cssd.k12.vt.us refers to the Charlotte Central School in the Chittenden South School District which happens to be in Charlotte, Vermont.) More information about the us domain may be found in RFC 1480.
The scheme of TLD assignment and management has worked well for many years, but the pressures of increased commercial activity, network size, and international use have caused controversy about how names can be fairly assigned without violating trademarks and conflicting claims to names. In November 1996, an Internet International Ad Hoc Committee (IAHC) was formed to resolve some of these naming issues and to act as a focal point for the international debate over a proposal to establish additional global naming registries and global gTLDs. The IAHC was dissolved in May 1997 with the publication of the Generic Top Level Domain Memorandum of Understanding framework. The Council of Registrars (CORE) an operational body made up of all of the Registrars established under the gTLD-MoU framework.
In November 2000, the first new set of TLDs were approved by ICANN, the first one of which went online in October 2001. The seven new TLDs, their purpose, and applicants are:
.aero - aviation industry, application by Societe Internationale de Telecommunications Aeronautiques SC (SITA)
.biz - businesses, application by JVTeam, LLC (administered by NeuLevel)
.coop - business cooperatives, application by National Cooperative Business Association (NCBA)
.info - general use, application by Afilias, LLC (administered by Afilias)
.museum - museums, application by Museum Domain Management Association (MDMA)
.name - individuals, application by Global Name Registry, LTD
.pro - professionals, application by RegistryPro, LTD
More information about these TLDs, the registration process, and new TLDs can be found at the ICANN New TLD Program Web page.
Last but not least, there is the never-ending issue of who owns domain names and IP addresses. I will make no claim to provide an authoritative answer but... domain names are owned by whoever registers them. This alone is a potential problem. Some ISPs are obtaining names on behalf of their customers and paying the annual fee. The issue has already arisen, "Who owns the name? The registrar or the customer?" Most ISPs have stated that they believe that the customer owns the name, even if the ISP registers the name, because there would be no reason for them to keep the name. Consider, however, that if an ISP insisted that it owned a name, it essentially ties a customer to an ISP forever, destroying the concept of domain name portability.
There is also an issue of violation of trade mark, service mark, or copyright in the choice and ownership of domain names. Consider this example from the 2001 era. A common Microsoft tag line is Where Would You Like to go Today? It so happens that the domain name wherewouldyouliketogotoday.com was registered to The Eagles Nest in Corfu, NY. I don‘t know anything about The Eagles Nest of Corfu, NY but it should not be mistaken for either Eagles Nest Enterprises of Grapevine, TX (the owner of eaglesnest.com) nor The Eagles Nest Internet Services of Newark, OH (owner of theeaglesnest.com).
In any case, suppose that Microsoft decided that someone using their service mark was not in their best interest and they pursued the issue; could they wrestle that domain name away from another registrant? Today‘s general rule of thumb is that if an organization believes that it‘s name or mark is being used in someone else‘s domain name in an unfair or misleading way, then they can take legal action against the name holder and the assignment of the name will be held up pending the outcome of the legal action. More information about this issue can be found at ICANN‘s Uniform Domain-Name Dispute-Resolution Policy Web page. By the way, this is, of course, the question behind the new industry of cybersquatting; someone registers a domain name hoping that someone else with buy it from them later on!
And what about IP addresses? Prior to the widespread use of CIDR (see Section 3.2.1), individual organizations were assigned an address (usually a Class C!) and domain name at the same time. In general, the holder of the domain name owned the IP address and if they changed ISP, routing tables throughout the Internet were updated.
Today, ISPs are assigned addresses in blocks called CIDR blocks. A customer today, whether they already own a domain name or are obtaining a new one, will be assigned an IP address from the ISP‘s CIDR block. If the customer changes ISP, they have to relinquish the IP address.
A good overview of the naming and addressing procedures can be found in RFC 2901, titled "Guide to Administrative Procedures of the Internet Infrastructure."
3. The TCP/IP Protocol Architecture
TCP/IP is most commonly associated with the Unix operating system. While developed separately, they have been historically tied, as mentioned above, since 4.2BSD Unix started bundling TCP/IP protocols with the operating system. Nevertheless, TCP/IP protocols are available for all widely-used operating systems today and native TCP/IP support is provided in OS/2, OS/400, and Windows 9x/NT/2000, as well as most Unix variants.
Figure 2 shows the TCP/IP protocol architecture; this diagram is by no means exhaustive, but shows the major protocol and application components common to most commercial TCP/IP software packages and their relationship.
The sections below will provide a brief overview of each of the layers in the TCP/IP suite and the protocols that compose those layers. A large number of books and papers have been written that describe all aspects of TCP/IP as a protocol suite, including detailed information about use and implementation of the protocols. Some good TCP/IP references are:
TCP/IP Illustrated, Volume I: The Protocols by W.R. Stevens (Addison-Wesley, 1994)
Troubleshooting TCP/IP by Mark Miller (John Wiley & Sons, 1999)
Guide to TCP/IP, 2/e by Laura A. Cappell and Ed Tittel (Thomson Course Technology, 2004)
TCP/IP: Architecture, Protocols, and Implementation with IPv6 and IP Security by S. Feit (McGraw-Hill, 2000)
Internetworking with TCP/IP, Vol. I: Principles, Protocols, and Architecture, 2/e, by D. Comer (Prentice-Hall, 1991)
"TCP/IP Tutorial" by T.J. Socolofsky and C.J. Kale (RFC 1180, Jan. 1991)
"TCP/IP and tcpdump Pocket Reference Guide", developed by the author for The SANS Institute
3.1. The Network Interface Layer
The TCP/IP protocols have been designed to operate over nearly any underlying local or wide area network technology. Although certain accommodations may need to be made, IP messages can be transported over all of the technologies shown in the figure, as well as numerous others. It is beyond the scope of this paper to describe most of these underlying protocols and technologies.
Two of the underlying network interface protocols, however, are particularly relevant to TCP/IP. The Serial Line Internet Protocol (SLIP, RFC 1055) and Point-to-Point Protocol (PPP, RFC 1661), respectively, may be used to provide data link layer protocol services where no other underlying data link protocol may be in use, such as in leased line or dial-up environments. Most commercial TCP/IP software packages for PC-class systems include these two protocols. With SLIP or PPP, a remote computer can attach directly to a host server and, therefore, connect to the Internet using IP rather than being limited to an asynchronous connection.
3.1.1. PPP
It is worth spending a little bit of time discussing PPP because of its importance in Internet access today. As its name implies, PPP was designed to be used over point-to-point links. In fact, it is the prevalent IP encapsulation scheme for dedicated Internet access as well as dial-up access. One of the significant strengths of PPP is its ability to negotiate a number of things upon initial connection, including passwords, IP addresses, compression schemes, and encryption schemes. In addition, PPP provides support for simultaneous multiple protocols over a single connection, an important consideration in those environments where dial-up users can employ either IP or another network Layer protocol. Finally, in environments such as ISDN, PPP supports inverse multiplexing and dynamic bandwidth allocation via the Multilink-PPP (ML-PPP) described in RFCs 1990 and 2125.
PPP generally uses an HDLC-like (bit-oriented protocol) frame format as shown in Figure 3, although RFC 1661 does not demand use of HDLC. HDLC defines the first and last two fields in the frame:
Flag: The 8-bit pattern "01111110" used to delimit the beginning and end of the transmission.
Address: For PPP, uses the 8-bit broadcast address, "11111111".
Frame Check Sequence (FCS): An 8-bit remainder from a cyclic redundancy check (CRC) calculation, used for bit error detection.
RFC 1661 actually describes the use of the three other fields in the frame:
Protocol: An 8- or 16-bit value that indicates the type of datagram carried in this frame‘s Information field. This field can indicate use of a particular Network Layer protocol (such as IP, IPX, or DDP), a Network Control Protocol (NCP) in support of one of the Network Layer protocols, or a PPP Link-layer Control Protocol (LCP). The entire list of possible PPP values in this field can be found in the IANA list of PPP protocols.
Information: Contains the datagram for the protocol specified in the Protocol field. This field is zero or more octets in length, up to a (default) maximum of 1500 octets (although a different value can be negotiated).
Padding: Optional padding to add length to the Information field. May be required in some implementations to ensure some minimum frame length and/or to ensure some alignment on computer word boundaries.
The operation of PPP is basically as follows:
After the link is physically established, each host sends LCP packets to configure and test the data link. It is here where the maximum frame length, authentication protocol (Password Authentication Protocol, PAP, or Challenge-Handshake Authentication Protocol, CHAP), link quality protocol, compression protocol, and other configuration parameters are negotiated. Authentication, if it used, will occur after the link has been established.
After the link is established, one or more Network Layer protocol connections are configured using the appropriate NCP. If IP is to be used, for example, it will be set up using PPP‘s IP Control Protocol (IPCP). Once each of the Network Layer protocols has been configured, datagrams from those protocols can be sent over the link. Control protocols may be used for IP, IPX (NetWare), DDP (AppleTalk), DECnet, and more.
The link will remain configured for communications until LCP and/or NCP packets close the link down.
3.2. The Internet Layer
The Internet Protocol (RFC 791), provides services that are roughly equivalent to the OSI Network Layer. IP provides a datagram (connectionless) transport service across the network. This service is sometimes referred to as unreliable because the network does not guarantee delivery nor notify the end host system about packets lost due to errors or network congestion. IP datagrams contain a message, or one fragment of a message, that may be up to 65,535 bytes (octets) in length. IP does not provide a mechanism for flow control.
The basic IP packet header format is shown in Figure 4. The format of the diagram is consistent with the RFC; bits are numbered from left-to-right, starting at 0. Each row represents a single 32-bit word; note that an IP header will be at least 5 words (20 bytes) in length. The fields contained in the header, and their functions, are:
Version: Specifies the IP version of the packet. The current version of IP is version 4, so this field will contain the binary value 0100. [NOTE: Actually, many IP version numbers have been assigned besides 4 and 6; see the IANA‘s list of IP Version Numbers.]
Internet Header Length (IHL): Indicates the length of the datagram header in 32 bit (4 octet) words. A minimum-length header is 20 octets, so this field always has a value of at least 5 (0101) Since the maximum value of this field is 15, the IP Header can be no longer than 60 octets.
Type of Service (TOS): Allows an originating host to request different classes of service for packets it transmits. Although not generally supported today in IPv4, the TOS field can be set by the originating host in response to service requests across the Transport Layer/Internet Layer service interface, and can specify a service priority (0-7) or can request that the route be optimized for either cost, delay, throughput, or reliability.
Total Length: Indicates the length (in bytes, or octets) of the entire packet, including both header and data. Given the size of this field, the maximum size of an IP packet is 64 KB, or 65,535 bytes. In practice, packet sizes are limited to the maximum transmission unit (MTU).
Identification: Used when a packet is fragmented into smaller pieces while traversing the Internet, this identifier is assigned by the transmitting host so that different fragments arriving at the destination can be associated with each other for reassembly.
Flags: Also used for fragmentation and reassembly. The first bit is called the More Fragments (MF) bit, and is used to indicate the last fragment of a packet so that the receiver knows that the packet can be reassembled. The second bit is the Don‘t Fragment (DF) bit, which suppresses fragmentation. The third bit is unused (and always set to 0).
Fragment Offset: Indicates the position of this fragment in the original packet. In the first packet of a fragment stream, the offset will be 0; in subsequent fragments, this field will indicates the offset in increments of 8 bytes.
Time-to-Live (TTL): A value from 0 to 255, indicating the number of hops that this packet is allowed to take before discarded within the network. Every router that sees this packet will decrement the TTL value by one; if it gets to 0, the packet will be discarded.
Protocol: Indicates the higher layer protocol contents of the data carried in the packet; options include ICMP (1), TCP (6), UDP (17), or OSPF (89). A complete list of IP protocol numbers can be found at the IANA‘s list of Protocol Numbers. An implementation-specific list of supported protocols can be found in the protocol file, generally found in the /etc (Linux/Unix), c:\windows (Windows 9x, ME), or c:\winnt\system32\drivers\etc (Windows NT, 2000) directory.
Header Checksum: Carries information to ensure that the received IP header is error-free. Remember that IP provides an unreliable service and, therefore, this field only checks the IP header rather than the entire packet.
Source Address: IP address of the host sending the packet.
Destination Address: IP address of the host intended to receive the packet.
Options: A set of options which may be applied to any given packet, such as sender-specified source routing or security indication. The option list may use up to 40 bytes (10 words), and will be padded to a word boundary; IP options are taken from the IANA‘s list of IP Option Numbers.
3.2.1. IP Addresses
IP addresses are 32 bits in length (Figure 5). They are typically written as a sequence of four numbers, representing the decimal value of each of the address bytes. Since the values are separated by periods, the notation is referred to as dotted decimal. A sample IP address is 208.162.106.17.
1 1 1 1 1 1 1 1 1 1 2 2 2 2 2 2 2 2 2 2 3 3 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 --+-------------+------------------------------------------------ Class A |0| NET_ID | HOST_ID | |-+-+-----------+---------------+-------------------------------| Class B |1|0| NET_ID | HOST_ID | |-+-+-+-------------------------+---------------+---------------| Class C |1|1|0| NET_ID | HOST_ID | |-+-+-+-+---------------------------------------+---------------| Class D |1|1|1|0| MULTICAST_ID | |-+-+-+-+-------------------------------------------------------| Class E |1|1|1|1| EXPERIMENTAL_ID | --+-+-+-+--------------------------------------------------------
FIGURE 5. IP Address Format.
IP addresses are hierarchical for routing purposes and are subdivided into two subfields. The Network Identifier (NET_ID) subfield identifies the TCP/IP subnetwork connected to the Internet. The NET_ID is used for high-level routing between networks, much the same way as the country code, city code, or area code is used in the telephone network. The Host Identifier (HOST_ID) subfield indicates the specific host within a subnetwork.
To accommodate different size networks, IP defines several address classes. Classes A, B, and C are used for host addressing and the only difference between the classes is the length of the NET_ID subfield:
A Class A address has an 8-bit NET_ID and 24-bit HOST_ID. Class A addresses are intended for very large networks and can address up to 16,777,214 (224-2) hosts per network. The first bit of a Class A address is a 0 and the NETID occupies the first byte, so there are only 128 (27) possible Class A NETIDs. In fact, the first digit of a Class A address will be between 1 and 126, and only about 90 or so Class A addresses have been assigned.
A Class B address has a 16-bit NET_ID and 16-bit HOST_ID. Class B addresses are intended for moderate sized networks and can address up to 65,534 (216-2) hosts per network. The first two bits of a Class B address are 10 so that the first digit of a Class B address will be a number between 128 and 191; there are 16,384 (214) possible Class B NETIDs. The Class B address space has long been threatened with being used up and it is has been very difficult to get a new Class B address for some time.
A Class C address has a 24-bit NET_ID and 8-bit HOST_ID. These addresses are intended for small networks and can address only up to 254 (28-2) hosts per network. The first three bits of a Class C address are 110 so that the first digit of a Class C address will be a number between 192 and 223. There are 2,097,152 (221) possible Class C NETIDs and most addresses assigned to networks today are Class C (or sub-Class C!).
The remaining two address classes are used for special functions only and are not commonly assigned to individual hosts. Class D addresses may begin with a value between 224 and 239 (the first 4 bits are 1110), and are used for IP multicasting (i.e., sending a single datagram to multiple hosts); the IANA maintains a list of Internet Multicast Addresses. Class E addresses begin with a value between 240 and 255 (the first 4 bits are 1111), and are reserved for experimental use.
Several address values are reserved and/or have special meaning. A HOST_ID of 0 (as used above) is a dummy value reserved as a place holder when referring to an entire subnetwork; the address 208.162.106.0, then, refers to the Class C address with a NET_ID of 208.162.106. A HOST_ID of all ones (usually written "255" when referring to an all-ones byte, but also denoted as "-1") is a broadcast address and refers to all hosts on a network. A NET_ID value of 127 is used for loopback testing and the specific host address 127.0.0.1 refers to the localhost.
Several NET_IDs have been reserved in RFC 1918 for private network addresses and packets will not be routed over the Internet to these networks. Reserved NET_IDs are the Class A address 10.0.0.0 (formerly assigned to ARPANET), the sixteen Class B addresses 172.16.0.0-172.31.0.0, and the 256 Class C addresses 192.168.0.0-192.168.255.0.
An additional addressing tool is the subnet mask. Subnet masks are used to indicate the portion of the address that identifies the network (and/or subnetwork) for routing purposes. The subnet mask is written in dotted decimal and the number of 1s indicates the significant NET_ID bits. For "classful" IP addresses, the subnet mask and number of significant address bits for the NET_ID are:
Class
Subnet Mask
Number of Bits A 255.0.0.0 8 B 255.255.0.0 16 C 255.255.255.0 24
Depending upon the context and literature, subnet masks may be written in dotted decimal form or just as a number representing the number of significant address bits for the NET_ID. Thus, 208.162.106.17 255.255.255.0 and 208.162.106.17/24 both refer to a Class C NET_ID of 208.162.106. Some, in fact, might refer to this 24-bit NET_ID as a "slash-24."
Subnet masks can also be used to subdivide a large address space into subnetworks or to combine multiple small address spaces. In the former case, a network may subdivide their address space to define multiple logical networks by segmenting the HOST_ID subfield into a Subnetwork Identifier (SUBNET_ID) and (smaller) HOST_ID. For example, user assigned the Class B address space 172.16.0.0 could segment this into a 16-bit NET_ID, 4-bit SUBNET_ID, and 12-bit HOST_ID. In this case, the subnet mask for Internet routing purposes would be 255.255.0.0 (or "/16"), while the mask for routing to individual subnets within the larger Class B address space would be 255.255.240.0 (or "/20").
But how a subnet mask work? To determine the subnet portion of the address, we simply perform a bit-by-bit logical AND of the IP address and the mask. Consider the following example: suppose we have a host with the IP address 172.20.134.164 and a subnet mask 255.255.0.0. We write out the address and mask in decimal and binary as follows:
172.020.134.164 10101100.00010100.10000110.10100100 AND 255.255.000.000 11111111.11111111.00000000.00000000 --------------- ----------------------------------- 172.020.000.000 10101100.00010100.00000000.00000000
From this we can easily find the NET_ID 172.20.0.0 (and can also infer the HOST_ID 134.164).
As an aside, most ISPs use a /30 address for the WAN links between the network and the customer. The router on the customer‘s network will generally have two IP addresses; one on the LAN interface using an address from the customer‘s public IP address space and one on the WAN interface leading back to the ISP. Since the ISP would like to be able to ping both sides of the router for testing and maintenance, having an IP address for each router port is a good idea.
By using a /30 address, a single Class C address can be broken up into 64 smaller addresses. Here‘s an example. Suppose an ISP assigns a particular customer the address 24.48.165.130 and a subnet mask 255.255.255.252. That would look like the following:
024.048.165.130 00011000.00110000.10100101.10000010 AND 255.255.255.252 11111111.11111111.11111111.11111100 --------------- ----------------------------------- 024.048.165.128 00011000.00110000.10100101.10000000
So we find the NET_ID to be 24.48.165.128. Since there‘s a 30-bit NET_ID, we are left with a 2-bit HOST_ID; thus, there are four possible host addresses in this subnet: 24.48.165.128 (00), .129 (01), .130 (10), and .131 (11). The .128 address isn‘t used because it is all-zeroes; .131 isn‘t used because it is all-ones. That leave .129 and .130, which is ok since we only have two ends on the WAN link! So, in this case, the customer‘s router might be assigned 24.48.165.130/30 and the ISP‘s end of the link might get 24.48.165.129/30. Use of this subnet mask is very common today (so common that there is a proposal to allow the definition of 2-address NET_IDs specifically for point-to-point WAN links).
A very good IP addressing tutorial can be found in Chuck Semeria‘s "Understanding IP Addressing: Everything You Ever Wanted to Know." If you are really interested in subnet masks, there are a number of subnet calculators on the Internet, including jafar.com‘s IP Subnet/Supernet Calculator, Net3 Group Inc.‘s IP Subnet Calculator, and Super Shareware‘s Subnet Calculator.
A last and final word about IP addresses is in order. Most Internet protocols specify that addresses be supplied in the form of a fully-qualified host name or an IP address in dotted decimal form. However, spammers and others have found a way to obfuscate IP addresses by supplying the IP address as a single large decimal number. Remember that IP addresses are 32-bit quantities. We write the address in dotted decimal for the convenience of humans; the computer still interprets dotted decimal as a 32-bit quantity. Therefore, writing the address as a single large decimal number will still allow the computer to see the address as a 32-bit number. For that reason, the following URLs will all take you to the same Web page:
http://www.garykessler.net
http://209.198.111.31
http://3519442719
3.2.2. Conserving IP Addresses: CIDR, DHCP, NAT, and PAT
The use of class-based (or classful) addresses in IP is one of the reasons that IP address exhaustion has been a concern since the early 1990s. Consider an organization, for example, that needs 1000 IP addresses. A Class C address is obviously to small so a Class B address would get assigned. But a Class B address offers more than 64,000 address, so over 63,000 addresses are wasted in this assignment.
An alternative approach is to assign this organization a block four Class C addresses, such as 192.168.128.0, 192.168.129.0, 192.168.130.0, and 192.168.131.0. By using a 22-bit subnet mask 255.255.252.0 (or "/22") for routing to this "block," the NET_ID assigned to this organization is 192.168.128.0.
This use of variable-size subnet masks is called Classless Interdomain Routing (CIDR), described in RFCs 1518 and 1519. In the example here, routing information for what is essentially four Class C addresses can be specified in a single router table entry.
But this concept can be expanded even more. CIDR is an important contribution to the Internet because it has dramatically limited the size of the Internet backbone‘s routing tables. Today, IP addresses are not assigned strictly on a first-come, first-serve basis, but have been preallocated to various numbering authorities around the world. The numbering authorities in turn, assign blocks of addresses to major (or first-tier) ISPs; these address blocks are called CIDR blocks. An ISP‘s customer (which includes ISPs that are customers of a first-tier ISP) will be assigned an IP NET_ID that is part of the ISP‘s CIDR block. So, for example, let‘s say that Gary Kessler ISP has a CIDR block containing the 256 Class C addresses in the range 196.168.0.0-196.168.255.0. This range of addresses could be represented in a routing table with the single entry 196.168.0.0/16. Once a packet hits the Gary Kessler ISP, it will be routed it to the correct end destination.
But don‘t stop now! By shrinking the size of the subnet mask so that a single NET_ID refers to multiple addresses (resulting in shrinking router tables), we could extend the size of the subnet mask to actually assign to an organization something smaller than a Class C address. As the Class C address space falls in danger of being exhausted, users are under increasing pressure to accept assignment of these sub-Class C addresses. An organization with just a few servers, for example, might be assigned, say, 64 addresses rather than the full 256. The standard subnet mask for a Class C is 24 bits, yielding a 24-bit NET_ID and 8-bit HOST_ID. If we use a "/26" mask (255.255.255.192), we can assign the same "Class C" to four different users, each getting 1/4 of the address space (and a 6-bit HOST_ID). So, for example, the IP address space 208.162.106.0 might be assigned as follows:
Note that in ordinary Class C usage, we would lose two addresses from the space — 0 and 255 — because addresses of all 0s and all 1s cannot be assigned as a HOST_ID. In the usage above, we would lose eight addresses from this space, because 0, 64, 128, and 192 have an all 0s HOST_ID and 63, 127, 191, and 255 have an all 1s HOST_ID. Each user, then, has 62 addresses that can be assigned to hosts.
The pressure on the Class C address space is continuing in intensity. Today, the pressure is not only to limit the number of addresses assigned, but organizations need to show why they need as many addresses as they want. Consider a company with 64 hosts and 3 servers. The ISP may request that that company only obtain 32 IP addresses. The rationale: the 3 servers need 3 addresses but the other hosts might be able to "share" the remaining pool of 27 addresses (recall that we lost HOST_ID addresses 0 and 31).
A pool of IP addresses can be shared by multiple hosts using a mechanism called Network Address Translation (NAT). NAT, described in RFC 1631, is typically implemented in hosts, proxy servers, or routers. The scheme works because every host on the user‘s network can be assigned an IP address from the pool of RFC 1918 private addresses; since these addresses are never seen on the Internet, this is not a problem.
FIGURE 6. Network Address Translation (NAT).
Consider the scenario shown in Figure 6. When the user accesses a Web site on the Internet, the NAT server will translate the "private" IP address of the host (192.168.50.50) into a "public" IP address (220.16.16.5) from the pool of assigned addresses. NAT works because of the assumption that, in this example, no more than 27 of the 64 hosts will ever be accessing the Internet at a single time.
But suppose that assumption is wrong. Another enhancement, called Port Address Translation (PAT) or Network Address Port Translation (NAPT), allows multiple hosts to share a single IP address by using different "port numbers" (ports are described more in Section 3.3).
FIGURE 7. Port Address Translation (PAT).
Port numbers are used by higher layer protocols (e.g., TCP and UDP) to identify a higher layer application. A TCP connection, for example, is uniquely identified on the Internet by the four values (aka 4-tuple) . The server‘s port number is defined by the standards while client port numbers can be any number greater than 1023. The scenario in Figure 7 shows the following three connections:
The client with the "private" IP address 192.168.50.50 (using port number 12002) connects to a Web server at address 98.10.10.5 (port 80).
The client with the "private" IP address 192.168.50.6 (using port number 22986) connects to the same Web server at address 98.10.10.5 (port 80).
The client with the "private" IP address 192.168.50.6 (using port number 8931) connects to an FTP server at address 99.12.18.6 (port 21).
PAT works in this scenario as follows. The router (running PAT software) can assign both local hosts with the same "public" IP address (220.16.16.5) and differentiate between the three packet flows by the source port.
A final note about NAT and PAT. Both of these solutions work and work fine, but they require that every packet be buffered, dissassembled, provided with a new IP address, a new checksum calculated, and the packet reassembled. In addition, PAT requires that a new port number be placed in the higher layer protocol data unit and new checksum calculated at the protocol layer above IP, too. The point is that NAT, and particularly PAT, results in a tremendous performance hit.
One advantage of NAT is that it makes IP address renumbering a thing of the past. If a customer has an IP NET_ID assigned from its ISP‘s CIDR block and then they change ISPs, they will get a new NET_ID. With NAT, only the servers need to be renumbered.
Another way to deal with renumbering is to dynamically assign IP addresses to host systems using the Dynamic Host Configuration Protocol (DHCP). DHCP is also an excellent solution for those environments where users move around frequently; it prevents the user from having to reconfigure their system when they move from, say, the Los Angeles office network to the New York office. For an introduction to DHCP, see RFC 2131 or "The Dynamic Host Configuration Protocol (DHCP) and Windows NT" by G. Kessler and C. Monaghan.
3.2.3. The Domain Name System
While IP addresses are 32 bits in length, most users do not memorize the numeric addresses of the hosts to which they attach; instead, people are more comfortable with host names. Most IP hosts, then, have both a numeric IP address and a name. While this is convenient for people, however, the name must be translated back to a numeric address for routing purposes.
Earlier discussion in this paper described the domain naming structure of the Internet. In the early ARPANET, every host maintained a file called hosts that contained a list of all hosts, which included the IP address, host name, and alias(es). This was an adequate measure while the ARPANET was small and had a slow rate of growth, but was not a scalable solution as the network grew.
[NOTE: A hosts file is still found on Unix systems although usually used to reconcile names of hosts on the local network to cut down on local DNS traffic; the file can usually be found in the /etc directory. On Microsoft Windows systems, the HOSTS file can typically be found in the c:\windows folder; in Windows NT and 2000, it can be found in c:\winnt\system32\drivers\etc.]
To handle the fast rate of new names on the network, the Domain Name System (DNS) was created. The DNS is a distributed database containing host name and IP address information for all domains on the Internet. There is a single authoritative name server for every domain that contains all DNS-related information about the domain; each domain also has at least one secondary name server that also contains a copy of this information. Thirteen root servers around the globe (most in the U.S., actually, with the remainder in Asia and Europe) maintain a list of all of these authoritative name servers.
When a host on the Internet needs to obtain a host‘s IP address based upon the host‘s name, a DNS request is made by the initial host to a local name server. The local name server may be able to respond to the request with information that is either configured or cached at the name server; if necessary information is not available, the local name server forwards the request to one of the root servers. The root server, then, will determine an appropriate name server for the target host and the DNS request will be forwarded to the domain‘s name server.
Name server data files contain the following types of records including:
A-record: An address record maps a hostname to an IP address.
PTR-record: A pointer record maps an IP address to a hostname.
NS-record: A name server record lists the authoritative name server(s) for a given domain.
MX-record: A mail exchange record lists the mail servers for a given domain. As an example, consider the author‘s e-mail address, kumquat@sover.net. Note that the "sover.net" portion of the address is a domain name, not a host name, and mail has to be sent to a specific host. The MX-records in the sover.net name database specifies the host mail.sover.net is the mail server for this domain.
CNAME-record: Canonical name records provide a mechanism of assigning aliases to host names, so that a single host with a IP address can be known by multiple names.
More information about the DNS can be found from the World Internetworking Alliance (WIA) Web site. Additional DNS references include DNS and BIND by P. Albitz and C. Liu (O‘Reilly & Associates) and "Setting up Your own DNS" by G. Kessler. The concepts, structure, and delegation of the DNS are described in RFCs 1034 and 1591. In addition, the IANA maintains a list of DNS parameters.
[ANOTHER NOTE: For Microsoft NetBIOS applications, the moral equivalent to the DNS is the Windows Internet Name Service (WINS), used to reconcile the NetBIOS name of a computer (e.g., \\ALTAMONT) to an IP address. A local WINS database can be created in the LMHOSTS file.]
3.2.4. ARP and Address Resolution
Early IP implementations ran on hosts commonly interconnected by Ethernet local area networks (LAN). Every transmission on the LAN contains the local network, or medium access control (MAC), address of the source and destination nodes. MAC addresses are 48-bits in length and are non-hierarchical, so routing cannot be performed using the MAC address. MAC addresses are never the same as IP addresses.
When a host needs to send a datagram to another host on the same network, the sending application must know both the IP and MAC addresses of the intended receiver; this is because the destination IP address is placed in the IP packet and the destination MAC address is placed in the LAN MAC protocol frame. (If the destination host is on another network, the sender will look instead for the MAC address of the default gateway, or router.)
Unfortunately, the sender‘s IP process may not know the MAC address of the intended receiver on the same network. The Address Resolution Protocol (ARP), described in RFC 826, provides a mechanism so that a host can learn a receiver‘s MAC address when knowing only the IP address. The process is actually relatively simple: the host sends an ARP Request packet in a frame containing the MAC broadcast address; the ARP request advertises the destination IP address and asks for the associated MAC address. The station on the LAN that recognizes its own IP address will send an ARP Response with its own MAC address. As Figure 2 shows, ARP message are carried directly in the LAN frame and ARP is an independent protocol from IP. The IANA maintains a list of all ARP parameters.
Other address resolution procedures have also been defined, including:
Reverse ARP (RARP), which allows a disk-less processor to determine its IP address based on knowing its own MAC address
Inverse ARP (InARP), which provides a mapping between an IP address and a frame relay virtual circuit identifier
ATMARP and ATMInARP provide a mapping between an IP address and ATM virtual path/channel identifiers.
LAN Emulation ARP (LEARP), which maps a recipient‘s ATM address to its LAN Emulation (LE) address (which takes the form of an IEEE 802 MAC address).
[NOTE: IP hosts maintain a cache storing recent ARP information. The ARP cache can be viewed from a Unix or DOS (in Windows 95/98/NT) command line using the arp -a command.]
3.2.5. IP Routing: OSPF, RIP, and BGP
As an OSI Network Layer protocol, IP has the responsibility to route packets. It performs this function by looking up a packet‘s destination IP NET_ID in a routing table and forwarding based on the information in the table. But it is routing protocols, and not IP, that populate the routing tables with routing information. There are three routing protocols commonly associated with IP and the Internet, namely, RIP, OSPF, and BGP.
OSPF and RIP are primarily used to provide routing within a particular domain, such as within a corporate network or within an ISP‘s network. Since the routing is inside of the domain, these protocols are generically referred to as interior gateways protocols.
The Routing Information Protocol version 2 (RIP-2), described in RFC 2453, describes how routers will exchange routing table information using a distance-vector algorithm. With RIP, neighboring routers periodically exchange their entire routing tables. RIP uses hop count as the metric of a path‘s cost, and a path is limited to 16 hops. Unfortunately, RIP has become increasingly inefficient on the Internet as the network continues its fast rate of growth. Current routing protocols for many of today‘s LANs are based upon RIP, including those associated with NetWare, AppleTalk, VINES, and DECnet. The IANA maintains a list of RIP message types.
The Open Shortest Path First (OSPF) protocol is a link state routing algorithm that is more robust than RIP, converges faster, requires less network bandwidth, and is better able to scale to larger networks. With OSPF, a router broadcasts only changes in its links‘ status rather than entire routing tables. OSPF Version 2, described in RFC 1583, is rapidly replacing RIP in the Internet.
The Border Gateway Protocol version 4 (BGP-4) is an exterior gateway protocol because it is used to provide routing information between Internet routing domains. BGP is a distance vector protocol, like RIP, but unlike almost all other distance vector protocols, BGP tables store the actual route to the destination network. BGP-4 also supports policy-based routing, which allows a network‘s administrator to create routing policies based on political, security, legal, or economic issues rather than technical ones. BGP-4 also supports CIDR. BGP-4 is described in RFC 1771, while RFC 1268 describes use of BGP in the Internet. In addition, the IANA maintains a list of BGP parameters.
As an alternative to using a routing protocol, the routing table can be maintained using "static routing." One example of static routing is the configuration of a default gateway at a host system; if the host needs to send an IP packet off of the local LAN segment, it is just blindly forwarded to the default gateway (router). Edge router‘s, too, commonly use static routing; the single router connecting a site to an ISP, for example, will usually just have a static routing table entry indicating that all traffic leaving the local LAN be forwarded to the ISP‘s access router. Since there‘s only a single path into the ISP, a routing protocol is hardly necessary.
All IP hosts and routers maintain a table that lists the most up-to-date routing information that that device knows. On a Windows system, you can examine the routing table by issuing a route print command; on Unix systems, use netstat -r.
Figure 2 shows the protocol relationship of RIP, OSPF, and BGP to IP. A RIP message is carried in a UDP datagram which, in turn, is carried in an IP packet. An OSPF message, on the other hand, is carried directly in an IP datagram. BGP messages, in a total departure, are carried in TCP segments over IP. Although all of the TCP/IP books mentioned above discuss IP routing to some level of detail, Routing in the Internet by Christian Huitema is one of the best available references on this specific subject.
3.2.6. IP version 6
The official version of IP that has been in use since the early 1980s is version 4. Due to the tremendous growth of the Internet and new emerging applications, it was recognized that a new version of IP was becoming necessary. In late 1995, IP version 6 (IPv6) was entered into the Internet Standards Track. The primary description of IPv6 is contained in RFC 1883 and a number of related specifications, including ICMPv6.
IPv6 is designed as an evolution from IPv4, rather than a radical change. Primary areas of change relate to:
Increasing the IP address size to 128 bits
Better support for traffic types with different quality-of-service objectives
Extensions to support authentication, data integrity, and data confidentiality
The architecture and structure of IPv6 addresses is described in RFC 2373. In July 1999, the IANA delegated the initial IPv6 address space to the worldwide regional registries in order to begin immediate worldwide deployment of IPv6 addresses. More information can be found at APNIC, ARIN, and RIPE.
For more information about IPv6, check out:
IPng: Internet Protocol Next Generation by Scott Bradner and Allison Mankin (Addison-Wesley, 1996)
IPv6: The New Internet Protocol by Christian Huitema (Prentice-Hall, 1996).
"IPv6: The Next Generation Internet Protocol" by Gary Kessler.
IPng and the TCP/IP Protocols by Stephen Thomas (John Wiley & Sons, 1996)
IPng Working Group page (IETF)
IP Next Generation Web Page (Sun)
6bone Web Page (LBL)
3.3. The Transport Layer Protocols
The TCP/IP protocol suite comprises two protocols that correspond roughly to the OSI Transport and Session Layers; these protocols are called the Transmission Control Protocol and the User Datagram Protocol (UDP). One can argue that it is a misnomer to refer to "TCP/IP applications," as most such applications actually run over TCP or UDP, as shown in Figure 2.
3.3.1. Ports
Higher-layer applications are referred to by a port identifier in TCP/UDP messages. The port identifier and IP address together form a socket, and the end-to-end communication between two hosts is uniquely identified on the Internet by the four-tuple (source port, source address, destination port, destination address).
Port numbers are specified by a 16-bit number. Port numbers in the range 0-1023 are called Well Known Ports. These port numbers are assigned to the server side of an application and, on most systems, can only be used by processes with a high level of privilege (such as root or administrator). Port numbers in the range 1024-49151 are called Registered Ports, and these are numbers that have been publicly defined as a convenience for the Internet community to avoid vendor conflicts. Server or client applications can use the port numbers in this range. The remaining port numbers, in the range 49152-65535, are called Dynamic and/or Private Ports and can be used freely by any client or server.
A complete list of port numbers that have been assigned can be found in the IANA‘s list of Port Numbers. An implementation-specific list of supported port numbers and services can be found in the services file, generally found in the /etc (Linux/Unix), c:\windows (Windows 9x, ME), or c:\winnt\system32\drivers\etc (Windows NT, 2000) directory.
3.3.2. TCP
TCP, described in RFC 793, provides a virtual circuit (connection-oriented) communication service across the network. TCP includes rules for formatting messages, establishing and terminating virtual circuits, sequencing, flow control, and error correction. Most of the applications in the TCP/IP suite operate over the reliable transport service provided by TCP.
The TCP data unit is called a segment; the name is due to the fact that TCP does not recognize messages, per se, but merely sends a block of bytes from the byte stream between sender and receiver. The fields of the segment (Figure 8) are:
Source Port and Destination Port: Identify the source and destination ports to identify the end-to-end connection and higher-layer application.
Sequence Number: Contains the sequence number of this segment‘s first data byte in the overall connection byte stream; since the sequence number refers to a byte count rather than a segment count, sequence numbers in contiguous TCP segments are not numbered sequentially.
Acknowledgment Number: Used by the sender to acknowledge receipt of data; this field indicates the sequence number of the next byte expected from the receiver.
Data Offset: Points to the first data byte in this segment; this field, then, indicates the segment header length.
Control Flags: A set of flags that control certain aspects of the TCP virtual connection. The flags include:
Urgent Pointer Field Significant (URG): When set, indicates that the current segment contains urgent (or high-priority) data and that the Urgent Pointer field value is valid.
Acknowledgment Field Significant (ACK): When set, indicates that the value contained in the Acknowledgment Number field is valid. This bit is usually set, except during the first message during connection establishment.
Push Function (PSH): Used when the transmitting application wants to force TCP to immediately transmit the data that is currently buffered without waiting for the buffer to fill; useful for transmitting small units of data.
Reset Connection (RST): When set, immediately terminates the end-to-end TCP connection.
Synchronize Sequence Numbers (SYN): Set in the initial segments used to establish a connection, indicating that the segments carry the initial sequence number.
Finish (FIN): Set to request normal termination of the TCP connection in the direction this segment is traveling; completely closing the connection requires one FIN segment in each direction.
Window: Used for flow control, contains the value of the receive window size which is the number of transmitted bytes that the sender of this segment is willing to accept from the receiver.
Checksum: Provides bit error detection for the TCP segment. The checksum field covers the TCP segment header and data, as well as a 96-bit pseudo header that contains the IP header Source Address, Destination Address, and Protocol fields, as well as the TCP segment length.
Urgent Pointer: Urgent data is information that has been marked as high-priority by a higher layer application; this data, in turn, usually bypasses normal TCP buffering and is placed in a segment between the header and "normal" data. The Urgent Pointer, valid when the URG flag is set, indicates the position of the first octet of nonexpedited data in the segment.
Options: Used at connection establishment to negotiate a variety of options; maximum segment size (MSS) is the most commonly used option and, if absent, defaults to an MSS of 536. Another option is Selective Acknowledgement (SACK), which allows out-of-sequence segments to be accepted by a receiver. The IANA maintains a list of all TCP Option Numbers.
3.3.3. UDP
UDP, described in RFC 768, provides an end-to-end datagram (connectionless) service. Some applications, such as those that involve a simple query and response, are better suited to the datagram service of UDP because there is no time lost to virtual circuit establishment and termination. UDP‘s primary function is to add a port number to the IP address to provide a socket for the application.
Source Port: Identifies the UDP port being used by the sender of the datagram; use of this field is optional in UDP and may be set to 0.
Destination Port: Identifies the port used by the datagram receiver.
Length: Indicates the total length of the UDP datagram.
Checksum: Provides bit error detection for the UDP datagram. The checksum field covers the UDP datagram header and data, as well as a 96-bit pseudo header that contains the IP header Source Address, Destination Address, and Protocol fields, as well as the UDP datagram length.
3.3.4. ICMP
The Internet Control Message Protocol, described in RFC 792, is an adjunct to IP that notifies the sender of IP datagrams about abnormal events. This collateral protocol is particularly important in the connectionless environment of IP. ICMP is not a classic host-to-host protocols like TCP or UDP, but is host-to-host in the sense that one device (e.g., a router or computer) is sending a message to another device (e.g., another router or computer).
The commonly employed ICMP message types include:
Destination Unreachable: Indicates that a packet cannot be delivered because the destination host cannot be reached. The reason for the non-delivery may be that the host or network is unreachable or unknown, the protocol or port is unknown or unusable, fragmentation is required but not allowed (DF-flag is set), or the network or host is unreachable for this type of service.
Echo and Echo Reply: These two messages are used to check whether hosts are reachable on the network. One host sends an Echo message to the other, optionally containing some data, and the receiving host responds with an Echo Reply containing the same data. These messages are the basis for the Ping command.
Parameter Problem: Indicates that a router or host encountered a problem with some aspect of the packet‘s Header.
Redirect: Used by a host or router to let the sending host know that packets should be forwarded to another address. For security reasons, Redirect messages should usually be blocked at the firewall.
Source Quench: Sent by a router to indicate that it is experiencing congestion (usually due to limited buffer space) and is discarding datagrams.
TTL Exceeded: Indicates that a datagram has been discarded because the TTL field reached 0 or because the entire packet was not received before the fragmentation timer expired.
Timestamp and Timestamp Reply: These messages are similar to the Echo messages, but place a timestamp (with millisecond granularity) in the message, yielding a measure of how long remote systems spend buffering and processing datagrams, and providing a mechanism so that hosts can synchronize their clocks.
ICMP messages are carried in IP packets. The IANA maintains a complete list of ICMP parameters.
3.3.5. TCP Logical Connections and ICMP
It is imperative to understand how a TCP connection is established to get a good feel for how TCP operates. TCP connections have three main parts: connection establishment, data exchange, and connection termination. The example below shows a POP3 server (listening on TCP port 110) being contacted by a client (using TCP port 1967).
The connection establishment phase comprises a three-way handshake during which time the client and server exchange their initial sequence number (ISN) and acknowledge the other host‘s ISN. In this example, the client starts by sending the server a TCP segment with the syn-bit set and a Sequence Number of 800. The syn-bit tells the receiver (i.e., the server) that the sender (i.e., the client) is in "ISN initialization" mode and that the ISN hasn‘t yet been confirmed. The segment‘s Acknowledgement Number isn‘t shown because its value is, at this point, invalid.
The server responds with a segment with the syn- and ack-bits set, a Sequence Number of 1567, and an Acknowledgement Number of 801. The syn-bit and ISN of 1567 have the same meaning as above. The ack-bit indicates the value of the Acknowledgement Number field is valid and the ACK value of 801 is the way in which the server confirms the client‘s ISN.
The final part of the three-way handshake is when the client sends a segment with just the ack-bit set. Note that the Acknowledgement Number field (1568) is one greater than the server‘s ISN.
This three-way handshake is sometimes referred to as an exchange of "syn, syn/ack, and ack" segments. It is important for a number of reasons. For individuals looking at packet traces, recognition of the three-way handshake is how to find the start of a connection. For firewalls, proxy severs, intrusion detectors, and other systems, it provides a way of knowing the direction of a TCP connection setup since rules may differ for outbound and inbound connections.
The second part of the TCP connection is data exchange. The information here is more or less made up for example purposes only; it shows a POP server sending a banner message to the client system, the user sending the "quit" command, and the server signing off. (Note that the "\n" indicates an "end-of-line" indicator.) These segments show the changing of, and relationship between, the client‘s and server‘s sequence and acknowledgement numbers.
The final phase is connection termination. Although TCP connections are full-duplex (even if a given application does not allow two-way simultaneous communication), the TCP protocol views the logical connection as a pair of simplex links. Therefore, connection termination requires four segments or, more properly, two pair of segments. In this case, the client sends the server a segment with the fin- and ack-bits set; the server responds with a segment with just the ack-bit set and the Acknowledgment Number is incremented. The server then sends a fin/ack segment to the client.
The paragraphs above describe a normal scenario setting up a TCP connection between a client and server. Two UDP hosts communicate in a similar fashion; one host sends a UDP datagram to the other which is presumably listening on the port indicated in the datagram.
But what happens if a host isn‘t listening on a port to which a connection is attempted or the host doesn‘t actually exist? Here‘s what happens in these "abnormal" conditions:
Host not listening on TCP port: If Host A attempts to contact Host B on a TCP port that Host B is not listening on, Host B responds with a TCP segment with the reset (RST) and acknowledge (ACK) flags set.
Host not listening on UDP port: If Host A attempts to contact Host B on a UDP port that Host B is not listening on, Host B sends an ICMP port unreachable message to Host A.
Host does not exist: If Host A attempts to contact Host B and Host B is not listening (e.g., Host B‘s IP address either doesn‘t exist or is unavailable), Host B‘s subnet‘s router will send an ICMP host unreachable message to Host A.
3.4. The TCP/IP Application Layer
The TCP/IP Application Layer protocols support the applications and utilities that are the Internet. This section will list a number of these applications and show a sample packet decode of all protocol layers.
3.4.1. TCP and UDP Applications
Commonly used protocols (as shown in Figure 2) include:
Archie: A utility that allows a user to search all registered anonymous FTP sites for files on a specified topic. Largely obsolete today, obviated by the World Wide Web.
BGP: The Border Gateway Protocol version 4 (BGP-4) is a distance vector exterior gateway routing protocol, commonly used between two ISPs or between a customer site and ISP if there are multiple links.
DNS: The Domain Name System (described in slightly more detail in Section 3.2.2 above) defines the structure of Internet names and their association with IP addresses, as well as the association of mail and name servers with domains.
Finger: Used to determine the status of other hosts and/or users (RFC 1288).
FTP: The File Transfer Protocol allows a user to transfer files between local and remote host computers (RFC 959).
Gopher: A tool that allows users to search through data repositories using a menu-driven, hierarchical interface, with links to other sites. Largely obsolete today, obviated by the World Wide Web (RFC 1436).
HTTP: The Hypertext Transfer Protocol is the basis for exchange of information over the World Wide Web (WWW). Various versions of HTTP are in use over the Internet, with HTTP version 1.0 (RFC 1945) being the most current. WWW pages are written in the Hypertext Markup Language (HTML), an ASCII-based, platform-independent formatting language (RFC 1866).
IMAP: The Internet Mail Access Protocol defines an alternative to POP as the interface between a user‘s mail client software and an e-mail server, used to download mail from the server to the client and providing significant flexibility in mailbox management.
OSPF: The Open Shortest Path First version 2 (OSPFv2) protocol is a link state routing protocol used within an organization‘s network. This is the preferred so-called interior gateway protocol.
Ping: A utility that allows a user at one system to determine the status of other hosts and the latency in getting a message to that host. Uses ICMP Echo messages. For more information and insight, see The Ping Page.
POP: The Post Office Protocol defines a simple interface between a user‘s mail client software (e.g., Eudora, Outlook, or the e-mail capability of your browser) and an e-mail server, used to download mail from the server to the client and allows the user to manage their mailboxes. The current version is POP3 (RFC 1460).
RADIUS: The Remote Authentication Dial-In User Service (RADIUS) is a remote-access protocol.
RIP: The Routing Information Protocol (RIP) is a distance-vector routing protocol used within an organization‘s network.
SSH: The Secure Shell is a protocol that allows remote logon to a host across the Internet, much like Telnet. Unlike Telnet, however, SSH encrypts passwords and data traffic.
SMTP: The Simple Mail Transfer Protocol is the standard protocol for the exchange of electronic mail over the Internet (RFC 821). SMTP is used between e-mail servers on the Internet or to allow an e-mail client to send mail to a server. RFC 822 specifically describes the mail message body format, and RFCs 1521 and 1522 describe MIME (Multipurpose Internet Mail Extensions). Reference books on electronic mail systems include !%@:: Addressing and Networks by D. Frey and R. Adams (O‘Reilly & Associates, 1993) and THE INTERNET MESSAGE: Closing the Book With Electronic Mail by M. Rose (PTR Prentice Hall, 1993).
SNMP: The Simple Network Management Protocol defines procedures and management information databases for managing TCP/IP-based network devices. SNMP (RFC 1157) is widely deployed in local and wide area networks. SNMP Version 2 (SNMPv2, RFC 1441) adds security mechanisms that are missing in SNMP, but is also very complex; widespread use of SNMPv2 has yet to be seen. Additional information on SNMP and TCP/IP-based network management can be found in SNMP by S. Feit (McGraw-Hill, 1994) and THE SIMPLE BOOK: An Introduction to Internet Management, 2/e, by M. Rose (PTR Prentice Hall, 1994).
SSL: The Secure Sockets Layer (SSL), designed by Netscape, provides a mechanism for secure communications over the Internet, based on certificates and public key cryptography. The most commonly known SSL application is HTTP over SSL, commonly designated as https. The newest version of SSL is called Transport Layer Security (TLS) (RFC 2246). SSL is not, however, HTTP-specific; protocols such as IMAP4 (imaps), FTP (ftps), Telnet (telnets), and POP3 (pop3s) all have definitions for operation over SSL.
TACACS+: The Terminal Access Controller Access Control System plus is a remote access protocol.
Telnet: Short for Telecommunication Network, a virtual terminal protocol allowing a user logged on to one TCP/IP host to access other hosts on the network (RFC 854).
TFTP: The Trivial File Transfer Protocol (TFTP) is used for some specialized simple file transfer applications.
Time/NTP: Time and the Network Time Protocol (NTP) are used so that Internet hosts can synchronize their system time from well-known Internet time servers.
Traceroute: A tool that displays the route taken by packets across the Internet between a local and remote host. The traceroute command is available on Linux/Unix systems; Windows systems starting with Windows 95 have a tracert command utility.
Whois/NICNAME: Utilities that search databases for information about Internet domains and domain contact information (RFC 3912).
A guide to using many of these applications can be found in "A Primer on Internet and TCP/IP Tools and Utilities" (FYI 30/RFC 2151) by Gary Kessler & Steve Shepard (also available in HTML, PDF, and Word).
3.4.2. Protocol Analysis
Full-blown protocol analysis is well beyond the scope of this paper. But a little introduction is ok!!
Today‘s protocol analyzers are usually software running on a computer or a specialized piece of hardware. In either case, the device‘s network interface card (NIC) operates in promiscuous mode so that the NIC captures every packet that flies by on the wire rather than only those packets addressed to this particular NIC. Most protocol analyzers also provide a display with at least a partial interpretation of the packets.
FIGURE 11. Protocol analyzer display.
Figure 11 shows the display from a GN Nettest WinPharoah protocol analyzer. In this case, we see the contents of a packet containing a POP3 message. The analyzer‘s display has three parts:
The top part shows a summary of the frames in the capture buffer. Note here that we see frames numbered 77-80 (column 1). The second column shows the frame length (in bytes); all use the Ethernet II frame format (column 3). The next two columns list the source and destination addresses; in this example, there are two communicating stations, named INSTRUCTOR (the server) and WINPHAROAH (the client). The summary column shows that these are POP3 commands and responses.
The middle section shows the packet decode; a detailed discussion of this is below.
The bottom section shows the frame in hexadecimal, as transmitted over the line. This is the raw bit stream.
The middle section is, indeed, the most interesting as this is where the frame contents are interpreted and displayed. The details of the IP, TCP, and POP3 protocols of Frame 80, the highlighted one, are shown here; the interpretation of the Ethernet frame itself is also available but is scrolled off the screen here.
Right after the Ethernet header information is the IP packet header. Note that this particular packet uses IP version 4, is 53 bytes in length, and carries a TCP segment. Note also that this packet was sent from the client (WINPHAROAH).
After IP is the TCP information. Note that the destination port number is 110, the port associated with a POP3 server. Since the POP3 server port is the destination, it means that this packet contains a POP command to the server from the client (which we knew anyway by looking at the summary of frame 80 above).
Finally we see the POP3 command itself. When a POP3 client connects to the server, the first thing it does is send the username using the POP3 USER command. If the username is valid, the server asks for a password, which is sent from the client in a POP3 PASS command, which is shown here. Note that the POP3 password is sent unencrypted!!!
The discussion here is only meant to readers a taste of one of the coolest tools that we get to play with in data communications; it is also an important tool for network managers and security administrators.
There are a fair number of free or inexpensive software packet sniffers that one can acquire for Linux or Windows systems. One of the most popular is tcpdump, which comes with many Linux distributions (e.g., Red Hat 7.1). WinDump is a tcpdump implementation for Windows, and the same group distributes Analyzer, a GUI packet sniffer. Ethereal is another GUI analyzer, with version for both Windows and Linux. More information on these packages can be found at my Packet Sniffing and Protocol Analysis Software page.
3.5. Summary
As this discussion has shown, TCP/IP is not merely a pair of communication protocols but is a suite of protocols, applications, and utilities. Increasingly, these protocols are referred to as the Internet Protocol Suite, but the older name will not disappear anytime soon.
Figure 12 shows the relationship between the various protocol layers of TCP/IP. Applications and utilities reside in host, or end-communicating, systems. TCP provides a reliable, virtual circuit connection between the two hosts. (UDP, not shown, provides an end-to-end datagram connection at this layer.) IP provides a datagram (DG) transport service over any intervening subnetworks, including local and wide area networks. The underlying subnetwork may employ nearly any common local or wide area network technology.
Note that the term gateway is used for the device interconnecting the two subnets, a device usually called a router in LAN environments or intermediate system in OSI environments. In OSI terminology, a gateway is used to provide protocol conversion between two networks and/or applications.
4. Other Information Sources
This memo has only provided background information about the TCP/IP protocols and the Internet. There is a wide range of additional information that the reader can access to further use and understand the tools and scope of the Internet. The real fun begins now!
Internet specifications, standards, reports, humor, and tutorials are distributed as Request for Comments (RFC) documents. RFCs are all freely available on-line, and most are available in ASCII text format.
Internet standards are documented in a subset of the RFCs, identified with an "STD" designation. RFC 2026 describes the Internet standards process and STD 1 always contains the official list of Internet standards.
For Your Information (FYI) documents are another RFC subset, specifically providing background information for the Internet community. The FYI notes are described in RFC 1150.
Frequently Asked Question (FAQ) lists may be found for a number of topics, ranging from ISDN and cryptography to the Internet and Gopher. Two such FAQs are of particular interest to Internet users: "FYI on Questions and Answers - Answers to Commonly asked ‘New Internet User‘ Questions" (RFC 1594) and "FYI on Questions and Answers: Answers to Commonly Asked ‘Experienced Internet User‘ Questions" (RFC 1207). All three of these documents point to even more information sources.
5. Acronyms and Abbreviations
ARP Address Resolution Protocol ARIN American Registry for Internet Numbers ARPANET Advanced Research Projects Agency Network ASCII American Standard Code for Information Interchange ATM Asynchronous Transfer Mode BGP Border Gateway Protocol BSD Berkeley Software Development CCITT International Telegraph and Telephone Consultative Committee CIX Commercial Internet Exchange CDPD Cellular Digital Packet Data protocol CSLIP Compressed Serial Line Internet Protocol DARPA Defense Advanced Research Projects Agency DDP Datagram Delivery Protocol DDS Digital data service DNS Domain Name System DOCSIS Data Over Cable System Interface Specification DoD U.S. Department of Defense DWDM Dense Wave Division Multiplexing FAQ Frequently Asked Questions lists FDDI Fiber Distributed Data Interface FTP File Transfer Protocol FYI For Your Information series of RFCs GOSIP U.S. Government Open Systems Interconnection Profile HDLC High-level Data Link Control HTML Hypertext Markup Language HTTP Hypetext Transfer Protocol IAB Internet Activities Board IANA Internet Assigned Numbers Authority ICANN Internet Corporation for Assigned Names and Numbers ICMP Internet Control Message Protocol IESG Internet Engineering Steering Group IETF Internet Engineering Task Force IMAP Internet Message Access Protocol InterNIC Internet Network Information Center IP Internet Protocol IPX Internetwork Packet Exchange ISDN Integrated Services Digital Network ISO International Organization for Standardization ISOC Internet Society ITU-T International Telecommunication Union Telecommunication Standardization Sector MAC Medium (or media) access control Mbps Megabits (millions of bits) per second NICNAME Network Information Center name service NSF National Science Foundation NSFNET National Science Foundation Network NTP Network Time Protocol OSI Open Systems Interconnection OSPF Open Shortest Path First PING Packet Internet Groper POP3 Post Office Protocol v3 PPP Point-to-Point Protocol RADIUS Remote Authentication Dial-In User Service RARP Reverse Address Resolution Protocol RIP Routing Information Protocol RFC Request For Comments SDH Synchronous Digital Hierarchy SLIP Serial Line Internet Protocol SMDS Switched Multimegabit Data Service SMTP Simple Mail Transfer Protocol SNAP Subnetwork Access Protocol SNMP Simple Network Management Protocol SONET Synchronous Optical Network SSL Secure Sockets Layer STD Internet Standards series of RFCs TACACS+ Terminal Access Controller Access Control System plus TCP Transmission Control Protocol TFTP Trivial File Transfer Protocol TLD Top-level domain UDP User Datagram Protocol WAP Wireless Application Protocol xDSL Digital Subscriber Line family of technologies
6. About the Author
Gary Kessler is Assistant Professor and program director of the Computer Networking and Computer & Digital Forensics programs at Champlain College, as well as the president and janitor of Gary Kessler Associates, an independent telecommunications consulting and training firm specializing in computer and network security, Internet access issues, and TCP/IP networking. He has written over 65 papers for industry publications. Gary‘s main areas of interest include TCP/IP and the Internet, computer and network forensics, network security, electronic commerce, and fast packet technologies and services. His e-mail address is kumquat@sover.net.
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