Chapter 7 and 8

Ports

A port is a number that represents a process running on a network. Ports are associated with OSI Layer 5, but in every packet, there will be both a source and destination port embedded in the Transmission Control Protocol (TCP) or User Datagram Protocol (UDP) header. Both clients and servers use port numbers to identify themselves. Because a single device can be running many client and server processes at the same time, each process has its own port number to keep it unique. Ports are used to keep separate network conversations separate. For example, if you open several browsers at once, even to the same website, each browser has its own port number. This prevents the content in one browser from suddenly appearing in another browser. If your device offers any type of service on the network, that service will listen on its designated port for incoming connections. You can examine all of the ports that are being listened on or are currently in use on your device by entering the netstat -na command at a command prompt. All ports are assigned a number in a range from 0 to 65,535. The Internet Assigned Numbers Authority (IANA) separates port numbers into three blocks: well-known ports, which are preassigned to system processes by IANA; registered ports, which are available to user processes and are listed as a convenience by IANA; and dynamic ports, which are assigned by a client operating system as needed when there is a request for service. There are three blocks of port numbers that are commonly used. Well-known ports Port range: 0 to 1,023 These ports are preassigned for use by common, or well-known, services. Often, the services that run on these ports must be started by a privileged user. Services in this range include Hypertext Transfer Protocol (HTTP) on TCP port 80, Internet Message Access Protocol (IMAP) on TCP port 143, and DNS on UDP port 53. In addition, port 0 is not used. Registered ports Port range: 1,024 to 49,151 These ports are registered by software makers for use by specific applications and services that are not as well-known as the services in the well-known range. Services in the registered port range include Socket Secure (SOCKS) proxy on TCP port 1080, and Xbox Live on TCP and UDP port 3074. Dynamic or private ports Port range: 49,152 to 65,535 These ports are set aside for use by unregistered services and services (typically, client applications) needing a temporary connection. Well-Known TCP and UDP Port Numbers Commonly used port numbers are listed in the following table, along with the protocols and services that use those ports. Port number, Protocol, service name, service 7 TCP, UDP echo, Echo 20 TCP ftp-data, File Transfer (default data) 21 TCP ftp, file trnsfer (control) 22 TCP, UDP ssh, secure shell (ssh) 23 TCP, UDP telnet, telnet 25 TCP, smtp, smtp 53 TCP, UDP, dns, DNS 67 TCP, UDP bootps, DHCP (BOOTP) server 68 TCP, UDP bootpc, DHCP (BOOTP) client 69 UDP tftp, TFTP 80 TCP http, HTTP 110 TCP pop3, Post office protocol, version 3 (POP3) 123 UDP ntp, network time protocol (NTP) 137 UDP netbios, Network basic input/output system (NetBIOS) naming service 138 UDP netbios, NetBIOS datagram distribution service 139 TCP, netbios, NetBIOS session service 143 TCP, UDP, imap, IMAP 161 UDP, snmp, Simple network management protocol (SNMP) 194 TCP, irc, internet relay chat (IRC) 389 TCP, UDP ldap, Lightweight directory access protocol (LDAP) 443 TCP, https, HTTP server 445 TCP, smb, Server message block (SMB) 546 TCP, UDP dhcpv6-client, DHCPv6 client 547 TCP, UDP dhcpv6-server, DHCPv6 server 1720 TCP h.323, H.323 Call Setup 2427 UDP mgcp, Media Gateway Control Protocol (MGCP) gateway traffic 2727 TCP mgcp, MGCP callagent traffic 3389 TCP, UDP rdp, Remote Desktop Protocol (RDP) 5004 TCP, UDP rtp, Real-time Transport Protocol (RTP) media data 5005 TCP, UDP rtp, RTP control protocol 5060 TCP, UDP sip, Session Initiation Protocol (SIP) unencrypted signaling traffic 5061 TCP, UDP sip, SIP encrypted traffic Note: Although port 3389 does not fall within the specified range of well-known ports, this is one of the port numbers that you will need to remember. Note: The complete list of well-known ports and other port number assignments is available online at www.iana.org/assignments/port-numbers

Satellite transmissions systems

A satellite-based network offers immense geographical coverage, allowing for high-speed connections anywhere in the world to transmit data between endpoints. Satellite transmission systems are used as an alternative to conventional communications, and as a cost-effective method to transmit information to different locations globally. Satellite communications systems use line-of- sight (LoS) microwave transmission. A satellite system consists of two segments: space and ground. Space A space segment contains one or more satellites organized into a constellation and a ground station that provides operational control of the satellites. Ground A ground segment provides access from Earth stations to the satellite to meet the communication needs of users. The ground segment contains terminals that utilize the communication capabilities of the space segment. The ground segment contains three basic types of terminals. • Fixed terminals access satellites while they are stationary. • Transportable terminals are portable, but remain stationary during transmission. • Mobile terminals can communicate with satellites even when they are in motion. Satellites are used for a variety of purposes and each satellite service has different requirements. Satellite Internet The satellite Internet is a method of connecting to the Internet by using a satellite network. This method can be broadly classified as a one-way or two- way connection, based on how the request for an Internet connection reaches the satellite. In a one-way connection, the request for an Internet connection goes to the ISP via a phone line and is forwarded to the satellite. Satellite phone network A satellite phone is a telephone system that relies on the satellite network to provide services, instead of the local telephone switch infrastructure. Satellite phones can be handheld or fixed, usually connected to an antenna at the top of a building. When a call is made from a satellite phone to another satellite phone, the call is routed directly via the satellite. If a call is made to a regular phone, the satellite routes the call to the landline or cellular network via an Earth station known as the gateway. The gateway converts the signals so that the landline or cellular network can read them. Satellite phones work well in open spaces, but they do not have a good reception within buildings and enclosed spaces. Satellite television Satellite television is a method of relaying video and audio signals directly to a subscriber's television set by using satellites. A satellite TV network consists of a programming source that provides the original program. The satellite TV provider, also known as the direct broadcast (DB) center, then broadcasts these channels to the satellites, which receive the signals and rebroadcast them to Earth. The subscriber's dish antenna picks up the signals and sends them to the TV via a receiver, also known as the set-top box (STB). The satellite TV technology overcomes the disadvantage of broadcast networks, where an LoS arrangement is necessary. VSAT A very small aperture terminal (VSAT) is a telecommunication Earth station that consists of an antenna to transmit and receive signals from satellites. The size of a VSAT ranges from 1.2 to 2.4 meters in diameter. A network of VSATs provides a cost-effective solution to users who need to connect several sites or offices that are dispersed geographically. VSATs support transmission of voice, data, and video. A typical VSAT network consists of an antenna placed on top of a building and connected to a transceiver and modem by a cable. The modem converts the signals from the satellite into data or voice signals, and vice versa. VSAT networks can be connected in a point-to-point, star, or mesh network. GPS The global positioning system (GPS) is a navigational system that consists of a network of 27 satellites: 24 active and 3 in the standby mode. These satellites are arranged in such a pattern that at least four of them are visible from any part of the world. A GPS receiver receives the distance and time information from the four visible satellites and uses that information to calculate its current position. A GPS receiver needs an unobstructed view of the sky.

Asynchronous transfer mode

Asynchronous transfer mode (ATM) is a cell-switching network technology that supports high-speed ATM transfer of voice, video, and data in LANs, WANs, and telephone networks. ATM was standardized by the International Telecommunication Union (ITU) in 1988. It operates at Layers 1 and 2 of the OSI model. It was designed to be the "glue" that would connect all manner of disparate networks together, making LAN-to-WAN connectivity seamless. Due to its complexity and cost, relatively few companies have implemented it in their campus LAN backbones. However, ATM WAN implementations have become reasonably popular because of their versatility and high bandwidth availability. Information is transferred in fixed-size packets, called cells, each consisting of 53 bytes. ATM networks are made up of switches, which transport data cells among networks. ATM provides connection-oriented, end-to-end QoS. Unlike frame relay, it can guarantee QoS for a particular virtual channel (circuit) because all switches along a path agree upon the bandwidth before the connection is established. ATM uses a Layer 2 addressing scheme that is similar to telephone numbers. It has its own variant of the Address Resolution Protocol (ATM ARP), which maps Layer 3 network addresses such as Internet Protocol (IP) or Internet Packet Exchange (IPX) to the Layer 2 address. It can carry all manner of data, from very low-speed telemetry and low priority email and file transfers to voice, video, and high-speed real-time implementations such as high-definition video and teleradiology. Cell sequencing numbers guarantee that the data is reassembled in the proper order at the receiving end. ATM offers reliable QoS. In the early 1990s, it was is envisioned as the underlying technology of broadband Integrated Services Digital Network (BISDN), the anticipated standard for the emerging information superhighway. In the end, it was not ATM and BISDN, but IP and packet switching that became the de facto standard for the World Wide Web. ATM is still used in the backbone of most service providers. It is also the underlying technology for digital subscriber line (DSL). ATM handles broadband applications efficiently, and at the same time, allows users to assign priority to traffic on the network. ATM, like its predecessor frame relay, is currently being phased out, with major carriers migrating their customers' backbones to IP over multiprotocol label switching (MPLS). The versatility of ATM can be attributed to a variety of features. Bandwidth options Provides a wide range of high-bandwidth options. Although it has no inherent bandwidth limitations, typical ATM implementations can support from 51.84 Mbps to 2.488 Gbps, with 155 Mbps to 622 Mbps being the most common. Interfaces capable of 160 Gbps ATM throughput were introduced to the IEEE as early as 2001. In the other direction, both consumer and business DSL implementations tend to be lower speed, typically ranging from 192 Kbps to 100 Mbps. Types of traffic Fixed cell size Allows the capability to carry data, voice, and video simultaneously on the same channel. The fixed 53-byte cell size enables ATM to be implemented in hardware, reducing overhead and drain on resources required to move data on a network. QoS Built-in QoS features in the design aid in the flow of data between endpoints of the ATM network. Traffic contracting and shaping Traffic contracting assigns a set data rate to an endpoint. When an endpoint connects to an ATM network, it enters into a contract with the network for service quality. The ATM network will not contract more services than it can provide. Traffic shaping optimizes data flow on an ATM network. It includes control of bursts and optimizing bandwidth allocation. Real-time and non-real-time data support Real-time data support is used for time-sensitive data such as voice or video and travels at a higher priority than non-real-time data. ATM Network Interface Types ATM network interfaces connect ATM devices and fall into two categories: User-to-Network Interface (UNI) and Network-to-Network Interface (NNI). The UNI, as a user device, is an ATM border device that connects one ATM network to another ATM network or a LAN. NNI is a switch that is inside an ATM network. Individual devices can connect to an ATM network, but this is rare. ATM Connections ATM is not a channelized service and does not waste channels by assigning them to nodes that are not talking. In a situation when the device is offline, ATM does not hold the channel. It makes that bandwidth available to other nodes, exhibiting traffic contracting to allocate the necessary bandwidth without wasting it by reservation. An ATM switch makes virtual connections with other switches to provide a data path from endpoint to endpoint. Individual connections are called virtual channels (VCs). VCs support the connection-oriented transport between endpoints and are identified by a virtual channel identifier (VCI). VCs with a common path are tied together into virtual paths (VPs) and are identified by a virtual path identifier (VPI). You can form a transmission path (TP) by combining multiple VPs. Note: An ATM endpoint (or end system) contains an ATM network interface adapter.

DHCP Options

DHCP options enable you to configure specific values such as the address of the default gateway, the DNS server, the domain name suffix of the interface, and other IP-related information, relieving the administrator from having to manually configure these parameters. You can modify scope options at the scope, server, class, or client-specific levels. You can modify DHCP options on a DHCP server by using the DHCP Options Properties dialog box. There are different categories of options for DHCP. These options will always apply to all clients unless they are overridden by other settings at the client's end. Global options Globally for all DHCP servers and their clients Scope options To clients that obtain leases within a particular scope Class options To clients that specify a class when obtaining a scope lease Reserved client options To any client with a scope reservation for its IP address DHCP Reservations Reservations are lease assignments in DHCP that enable you to configure a permanent IP address for a particular client on the subnet. DHCP reservations are based on the client's media access control (MAC) address. Reserved IP addresses differ from statically configured IP addresses; when there are any changes in network parameters on the DHCP server, IP addresses receive the changes when they renew their leases. The DHCP Lease Process The DHCP lease process contains four main phases. CompTIA® Network+® (Exam N10-006) | 253 The DHCP Lease 1. Discover: Once a node comes online and loads a simple version of TCP/IP and it's ready to communicate with a DHCP server, it transmits a broadcast called a DHCP discover to the network's broadcast address of 255.255.255.255 to see if any DHCP servers are online and then request an IP address. 2. Offer: DHCP servers that are online respond with a directed lease offer packet that contains an IP address that the node can lease. 3. Request: The node accepts the first offer it receives and returns a request to lease the IP address from the DHCP server, called a DHCP request. 4. Acknowledge: The DHCP server acknowledges the request from the node with a DHCP ACK, which has the IP address and settings required for the leasing time and starts the lease. The DHCP server also updates the IP address in its database as being in use, to avoid reassigning the address. Note: For additional information, check out the LearnTO Follow the DHCP Lease Process presentation in the LearnTOs for this course on your CHOICE Course screen. BOOTP BOOTP (the Bootstrap Protocol) is the predecessor of DHCP. It was developed to assign IP addresses to diskless workstations that had no way of storing their operating system. After providing the IP address from a configuration server, the workstation would then seek to download its operating system from the network. With BOOTP, clients do not lease their addresses. Instead, they are assigned an IP address from a list on the configuration server. Although BOOTP has largely been superseded by DHCP, it is still used by enterprise organizations to roll out large numbers of "bare metal" boxes that have no OS initially. The devices boot from their network cards, sending out a BOOTP request for an IP address. After obtaining their address, they then seek to discover a Trivial File Transfer Protocol (TFTP) server that they can download their operating system from. BOOTP uses the same port, UDP 67, as DHCP. Most DHCP servers can be configured to respond to both BOOTP and DHCP client requests. DHCP Relay Agent A DHCP relay agent is a service that captures a BOOTP or DHCP broadcast and forwards it through the router as a unicast transmission to the DHCP server on another subnet. BOOTP uses a local broadcast that cannot be sent through routers on the network. As an administrator of a TCP/IP network using DHCP, you must either have a DHCP server on each subnet and configure the router to forward the broadcasts, or configure a DHCP relay agent. Having multiple DHCP servers also ensures a higher degree of fault tolerance as the unavailability of a DHCP server on a subnet does not prevent nodes from requesting or renewing their leases. The Internet Protocol Helper (IP Helper) is an API used by C and C++ programmers to retrieve and modify network configuration settings on the local computer. An IP Helper address can be used to forward DHCP broadcasts to their destination. One DHCP server can be used to examine address leases for all network devices and manage network IP subnetworks. The DHCP server returns an offer to the relay agent, which in turn presents the offer to the client. Once the client has its lease, it also has the DHCP server's IP address, so it does not need to use the relay agent to renew the lease. An important factor you need to consider on a network with multiple subnets is that the routers on the network must be RFC 1542-compliant to allow a DHCP server to receive the broadcast message from a node. IP Addresses Recovery The DHCP lease process is important to the overall performance of a DHCP system. By leasing addresses to clients instead of permanently assigning them, a DHCP server can recover addresses leased to offline clients that no longer need the addresses. A typical DHCP lease lasts for eight days, but the lease can be as short as one day, one hour, or even less depending on organizational requirements. For example, many organizations limit wireless DHCP leases to one day for security purposes. APIPA Automatic Private IP Addressing (APIPA) is a service that enables a DHCP client device to configure itself automatically with an IP address in the range of 169.254.0.1 to 169.254.255.254, in case no DHCP servers respond to the client's DHCP discover broadcast. In case of a DHCP server failure, when the clients on the network cannot obtain IP addresses, the clients can use APIPA to assign themselves an IP address in the 169.254.x.x address range to enable communication with other clients. Thus, APIPA enables DHCP clients to initialize TCP/IP and communicate on the local subnet even in the absence of an active DHCP scope. APIPA addresses are not routable, so devices with APIPA addresses cannot communicate outside of the local subnet. Note: If a client cannot reach destinations outside of the local subnet, check the device's IP address. If the client shows an APIPA address, it signals that the client is configured to use DHCP and that a DHCP server is unavailable. APIPA is not a practical replacement for receiving a DHCP lease. Instead, its most common usage is as a diagnostic tool. The presence of an APIPA address informs IT support personnel that a client attempted to receive a DHCP lease but failed, and therefore self-assigned its own IP address. APIPA Support APIPA is available on client systems including: Windows® 7 and Windows 8, and server operating systems including: Windows 2008, Windows 2008 R2, Windows 2012, and Windows 2012 R2, as well as Macintosh®. Because APIPA requires no administrative configuration, it was once thought that APIPA addressing could be used for small offices where local subnet communication is all that is required. In reality, however, nearly all offices in today's modern networks implement Internet access. APIPA cannot assign the address of the default gateway or DNS server. To assign these values, the administrator would have to manually configure these settings in every client. Additionally, you cannot have a self-assigned IP address with a manually configured default gateway. This makes APIPA an untenable addressing alternative.

Frame relay

Frame relay is a WAN protocol that functions at the Physical and Data Link layers (Layers 1 and 2) of the OSI model. It is a packet-switched technology that allows transmission of data over a shared network medium and bandwidth using virtual circuits. As virtual circuits consume bandwidth only when they transport data, each device can use more bandwidth and transmit data at higher speeds. Frame relay provides reliable communication lines and efficient error-handling mechanisms that discard erroneous data frames. Frame relay is the successor of X.25, and the predecessor of ATM. Mostly implemented at Layer 2, it eliminates the error correction features found in X.25, depending instead on a reliable digital network infrastructure. In frame relay, either a permanent or switched (on demand) virtual circuit is established in the provider's network for customer traffic. It uses traffic shaping and congestion management techniques, with upstream routers (configured as frame relay switches) matching the speed of the next hop, and even discarding lower priority traffic, if necessary. Because of its "bursty" nature, frame relay was not originally suited for real-time voice or video, although later developments sought to remedy this. Frame relay can still be found in some networks, but has largely been replaced by MPLS VPNs. Frame Relay Characteristics Frame relay link speeds can range from 56 Kbps to 1.544 Mbps, with the lower speeds, such as 56, 64, 128, 384, and 512 Kbps, being the most popular. Unlike a dedicated point-to-point lease line, frame relay has the concept of a committed information rate (CIR), which is the minimum bandwidth that a customer's virtual circuit is guaranteed to have. If the network is not busy, the circuit bandwidth may be allowed to exceed the CIR. If the network is congested, however, any traffic that exceeds the CIR is marked "discard eligible" and will be dropped. Some providers sell service plans with a CIR of 0, meaning that your traffic will be the lowest priority among all the customers and will always be dropped first. Frame relay uses a Layer 2 address called a data link connection identifier (DLCI). Each customer's connection to the provider's DCE (a frame relay switch known as a point of presence, or POP) has its own DLCI number, distinguishing it from other customer connections to that particular POP. The POP then maps the customer's DLCI to a specific virtual circuit inside the cloud. Frame Relay Network Components Frame relay uses DCEs and DTEs to connect to the appropriate frame relay network, referred to as the Frame Relay Bearer Service (FRBS). Inside the FRBS—or frame relay network cloud—is a network of switches that makes connections between endpoints. A virtual circuit is established between two DTE devices. DTE equipment can consist of a single network device such as a router. A DCE typically is a CSU/DSU that sends signals to an edge system (ES), a switch on the frame relay network. The virtual circuits used in frame relay prevent you from seeing the complexity of communication inside the cloud. There are two types of virtual circuits: permanent and switched. Permanent virtual circuits (PVCs) are created by service providers inside their devices and the circuit is constant. Switched virtual circuits (SVCs) are established during data transmission and when the data "conversation" is over, the connection is closed. Advantages and Disadvantages of Frame Relay The advantages of frame relay are: • It offers facilities like that of a leased line, but at a significantly lower cost. • It delivers increased performance with reduced network complexity. • It can be implemented over the existing technology. • It can be easily configured to combine traffic from different networking protocols. • It offers a pay-as-you-go structure. • It can carry traffic that is not IP traffic. The disadvantages of frame relay are: • Data transmission may exceed network capacity as clients use a common network, and this results in the slowing down of the network. • The "bursty" nature of traffic in a frame relay cloud, along with the use of variable-length frames, makes it difficult to provide QoS. During its most popular years in the 1990s, it was considered unsuitable for real-time traffic such as voice or video. By 1997, the Frame Relay Forum finally developed a standard for Voice over Frame Relay (VoFR).

SONET/SDH

The Synchronous Optical Network (SONET) is a standard for synchronous data transport over a fiber optic cable. SONET is the U.S. version of the standard published by ANSI, whereas SDH is the European version of the standard published by the International Telecommunications Union (ITU). SONET has two specifications: the Optical Carrier (OC) specification for fiber optic cabling and the Secure Transfer specification (STS) for copper wire, although SONET over copper has severe limitations. SONET is deployed in a self-healing dual-fiber ring topology, similar to Fiber Distributed Data Interface (FDDI). When one ring works, the other is a standby. Whenever the working ring fails, SONET recognizes the failure and switches over to the second ring. SONET is most widely used inside service providers to act as a high-speed backbone for other systems, such as frame relay, ATM, and Metro-Ethernet. It operates at Layer 1 of the OSI model. SONET/SDH can be used on an ATM network, and connections to the lines can be made by using single-mode or multi-mode optical fiber. In such a setup, ATM would be the switching technology, and SONET/SDH would be the transmission technology on the network. SONET is divided into three areas. Each area is controlled by an integrated management system. Local collector ring A local collector ring interfaces with users and comprises digital cross-connect switches (DCSs) at the user's location or connects to the user's location by a T- carrier. The DCS acts as a concentrator to transmit signals from a user to the SONET ring. It supports connections from different technologies and from multiple users. The technologies that can connect to the ring include ATM, T1, or T3 lines; ISDN; or DSL voice. Regional network A regional network combines multiple collector rings by using add/drop multiplexers (ADMs). The ADM allows data from collector rings to be added to the regional ring. The data that is not accepted by the service requester is discarded or sent back to the ADM. By managing bandwidth on the regional network, it becomes more efficient. When data moves between two networks that the same regional network supports, the connection can be through the regional network. Broadband backbone network The broadband backbone network routes data between regional networks. It is capable of carrying a large amount of data simultaneously in the ring, and the requester picks the data as it is transmitted. The key advantages of SONET are its excellent bandwidth management, built-in fault recovery features, and support for data transfer speeds of up to 2.48 Gbps. A particular advantage to SONET deployments is its interoperability. The technology often is used to aggregate multiple lines (T1, T3 for example). SONET's transmission bandwidth ranges from 51.84 Mbps to 2.48 Gbps. Its hardware actually operates at speeds in the 10 Gbps range, but the SONET standard has not been expanded to include it. The ITU is an international organization within the United Nations that defines global technical standards for telecommunications. ITU also coordinates the widespread use of the radio spectrum, ensuring interference-free wireless communications. ITU also sponsors exhibitions and forums to exchange ideas and discuss issues affecting international telecommunications. Dense wavelength division multiplexing (DWDM) is a multiplexing technology that uses light wavelengths to transmit data. DWDM is often used as an alternative to SONET to carry Metro-Ethernet. It operates at Layer 1 of the OSI model. Signals from multiple sources using different technologies are carried simultaneously on separate light wavelengths. DWDM can multiplex up to 80 separate data channels into a lightstream for transmission over an optical fiber. Data from different protocols and technologies such as IP, SONET, and ATM can all travel simultaneously within an optical fiber. SONET is combined with WDM functions by sending SONET data streams out on different colors of light. The sending SONET multiplexer connects light streams to the WDM card. At the receiving end, the fiber demultiplexes the light into a single color stream and sends it to SONET equipment. Coarse wavelength division multiplexing (CWDM) is a method of combining multiple signals on laser beams at various wavelengths for transmission along fiber optic cables, such that the number of channels is fewer than in DWDM. It uses increased channel spacing to allow less sophisticated and thus cheaper transceiver designs. DWDM and CWDM are based on the same concept of using multiple wavelengths of light on a single fiber, but differ in the spacing of the wavelengths, number of channels, and the ability to amplify the multiplexed signals in the optical space.

Services and Daemons

A daemon is a background process that performs a specific operation. Daemon is a UNIX term, though daemons are supported on other operating systems. Daemons on Windows are referred to as system agents or services.

Cable internet access

Cable Internet access uses a cable television connection and a cable modem to provide high-speed Internet access to homes and small businesses. With cable Internet access, your data is carried on two premium TV channels, one for transmit and one to receive. Cable access is contention-based, with users arranged in contention groups of nodes that split television and data signals at the cable provider's end. The speed of the network varies depending on the number of nodes in the contention group.

The MTR Utility

The My traceroute (MTR) utility combines ping and traceroute into a single function. MTR displays the routers traversed, the average time taken for round trip, and packet loss of each router. This utility helps network administrators identify latency or packet loss between two routers. MTR is used on UNIX-based systems. Note: The General Public License (GNU) is responsible for licensing and distributing MTR.

RDP

The Remote Desktop Protocol (RDP) is a proprietary protocol created by Microsoft for connecting to and managing devices that are not necessarily located at the same place as the administrator. It uses port 3389, runs on TCP, and works on the Application layer (Layer 7) of the OSI model. It and the remote desktop software allow a user to remotely log in to a networked device. The desktop interface, or application GUI, of the remote device looks as if it were accessed locally. RDP is a multiple-channel-capable protocol that allows for separate virtual channels for carrying device communication and presentation data from the server, as well as encrypted client mouse and keyboard data. RDP provides an extensible base and supports up to 64,000 separate channels for data transmission and provisions for multipoint transmission.

SMB

The Server Message Block (SMB) is a protocol that works on the Application layer (Layer 7) of the OSI SMB model and helps share resources such as files, printers, and serial ports among devices. SMB uses port 445 and runs on TCP. In a TCP/IP network, NetBIOS clients, such as Windows systems, use NetBIOS over TCP/IP to connect to servers, and then issue SMB commands to complete tasks such as accessing shared files and printers. Samba is a well-known open-source product that uses SMB to enable UNIX and Windows devices for sharing directories and files. Although the SMB protocol is primarily used in Microsoft networks, there are products such as NAS appliances that use SMB to facilitate file sharing across different operating system platforms. Linux can also support SMB, as well as act as a file and print server for Windows clients, if you enable the built-in Samba service on the Linux device.

SNMP

The Simple Network Management Protocol (SNMP) is an Internet protocol that enables administrators to monitor and manage network devices and traffic. Working at the Application layer (Layer 7) of the OSI model and the Application layer of the TCP/IP model, SNMP uses ports 161 and 162 to collect information from and send configuration commands to networking devices such as routers, switches, servers, workstations, printers, and any other SNMP-enabled devices. SNMP generally runs over UDP.

T-Carrier Systems

The T-carrier system is a digital and packet-switched system designed to carry multiplexed telephone connections. It makes communications more scalable than analog, circuit-switched systems. T- carrier systems use two twisted pairs of copper wires. The first pair is used for transmission and the second pair for reception. Therefore, T-carrier systems support full-duplex communication. T1 and T3 are the two most common T-service levels. E-1 and E-3 are European leased lines, roughly equivalent to United States T-1 and T-3 lines. T-services can be used to support a point-to-point WAN where the service provider sets up a dedicated connection between two T-service endpoints. T-services connect a customer's office with the service provider's network. The internal connection is over frame relay. The T-service can also connect an office to the telephone company for remote access. Individual remote clients dial in to a number and the service provider before being routed to the office through the T-service. This way, a server can service multiple dial-in connections without needing many modems. T-carriers operate at Layer 1 of the OSI model, and refer to the speed of the link. They can be used by several Layer 2 WAN protocols, including frame relay, primary rate ISDN, and PPP and high- level data link control (HDLC) in dedicated leased lines. Digital Signal Services Digital signal (DS) services are a hierarchy of different digital signals that transfer data at different rates. The T-carrier system is the most common physical implementation of the American National Standards Institute (ANSI) Digital Signal Hierarchy (DSH) specifications. DSH is a channelized data transmission standard used to multiplex several single data or voice channels for a greater total bandwidth. It was established in the 1980s, primarily for use with digital voice phones. In T-carrier implementations, DSH systems have become the standard building block of most channelized systems in the United States today. DSH defines a hierarchy of DSx specifications numbered DS0 to DS5. The basic DS0 level specifies a single voice channel of 64 Kbps. A DS1 signal bundles 24 DS0 channels and uses a T1 carrier line. The different types of DS services vary depending upon their data transmission rates. • DS0: Carries data at the rate of 64 Kbps. • DS1: Carries data at the rate of 1.5 Mbps. • DS2: Carries data at the rate of 6.3 Mbps. • DS3: Carries data at the rate of 44.4 Mbps. • DS4: Carries data at the rate of 274.2 Mbps. T-Lines In order to implement a different DS service, telephone companies use T-lines whose carrying capacities match the data rates of DS services. Depending on the number of DS0 links bundled together, you can get different amounts of bandwidth. Often, you will hear of links being referred to as fractional T1s, meaning that the customer has purchased less than the full 24 DS0s required to create a T1. E-Carrier Systems The E-carrier system is a dedicated digital line that transmits voice or data. It is used in Europe, Mexico, and South America. The different E-carriers transmit data at different rates. T-Carrier and E-Carrier Levels Level, T-carrier, e-carrier Level zero (channel data rate), 64 Kbps (DS0), 64 Kbps First level, 1.544 Mbps (DS1, 24 Ch., T1), 2.048 Mbps (E1, 32 Ch.) Intermediate level, T- carrier only, 3.152 Mbps (DS1C, 48 Ch.), - Second level, 6.312 Mbps (DS2, 96 Ch., T2), 8.448 Mbps (E2, 128 Ch.) Third level, 44.736 Mbps (DS3, 672 Ch., T3), 34.368 Mbps (E3, 512 Ch.) Fourth level, 274.176 Mbps (DS4, 4032 Ch.), 139.264 Mbps (E4, 2048 Ch.) Fifth level, 400.352 Mbps (DS5, 5760 Ch.), 565.148 Mbps (E5, 8192 Ch.)

Commands and Utilities for IP Networks

There are several commands and utilities available for you to configure, manage, and troubleshoot IP networks. Some you've already seen, such as using ipconfig to see if the default gateway is configured, and using ping to check for basic network connectivity. Other commands and utilities that you will probably find useful include: • route • tracert in Windows, and traceroute in UNIX and Linux • pathping • mtr utility in UNIX and Linux

Voice over IP (VoIP)

Voice over IP (VoIP) is a voice-over-data implementation in which voice signals are transmitted in real VoIP or near-real time over IP networks. In VoIP telephony, analog voice signals are converted into digital signals. As in a typical packet-switched network, digital signals are broken down into packets, to transmit voice as data. After reassembling the packets, the digital signals are reconverted into audio signals. When you make a telephone call, the network connection transmits signals over data networks, and transfers them to the standard phone system if the called party does not have a VoIP service. Conversely, when you dial a number that maps to a VoIP device, VoIP routes the call to the IP host. VoIP relies on the existing, robust infrastructure of IP networks and the near-universal implementation of IP. It also eliminates per-call costs, especially for long-distance calls, because it uses data channels to transmit voice signals. Unified voice services integrate a telecommunication network with an IP network. Typically, an IP- based PBX phone system is used to implement unified voice services. These services are implemented using VoIP phones. The phones can be hardware-based phones which look like traditional phones, but contain components that allow the phone to connect to the network. VoIP phones can also be software-based, where a device with a microphone and a sound card are used to make and receive calls. A VoIP provider or a SIP server are required to make and receive calls. Compared to traditional circuit-switched telephony, VoIP telephony provides various benefits for users and is thus gaining popularity. Cost reduction The most attractive benefit of VoIP telephony is the cost savings it offers. You can make a call to anywhere in the world, yet pay at the rates of downloads. For a business, the savings are especially considerable. Mobility Depending on the setup, you can make a VoIP call from anywhere you have Internet access. Reduced infrastructure With no need to provide for the cabling for a separate phone system, VoIP telephony reduces infrastructure and its inherent costs. Integrated communication As it is based on IP, some VoIP software integrates the transmission of not just voice data, but other forms of data. Thus, in addition to speaking with someone else, you can send image files and exchange video, such as through a webcam. Complementary features VoIP service providers usually offer many features for free, such as the caller ID and call forwarding, which are typically charged by fixed line service providers. Although VoIP telephony is gaining in popularity, it has many issues that need to be addressed before replacing or even competing with traditional telephony. Connectivity Because of the variable latency and unreliability of the Internet, it is not always a dependable choice for VoIP calls. Connections to the Internet are still not completely reliable with most providers, and there are times when you are not able to go online or you get disconnected often. An option would be to switch to a more reliable provider. Voice delivery As voice is delivered as packets, there may be periods of silence resulting from delays in packet delivery. This can not only be annoying, but it also consumes online time, as a conversation may take longer to complete. Power outage During a power outage, you are not able to go online and therefore cannot make a VoIP call. This is usually not a problem with traditional telephony, as phone companies provide for reserve power. An option would be to install a backup system. Security With the increasing popularity of VoIP telephony, security vulnerabilities, though not a big concern presently, are bound to increase. Hackers could not only listen to and intercept sensitive data, but even break in to systems and accounts to utilize VoIP services. Emergency 911 calls An emergency call from a traditional phone, in the event the caller is unable to speak, can be traced. However, it is difficult to trace a VoIP call, as voice packets bear an IP address rather than a location address. The problem gets more complicated if the person is using a portable device.

VoIP protocols

A VoIP session may use one or more protocols, depending on session parameters. Some consumer products such as SkypeTM and Google HangoutsTM do not use traditional VoIP protocols such as SIP or RTP. Skype for Business, however, does implement these protocols. Session Initiation Protocol (SIP) Initiates, modifies, and terminates a session. It is a signaling protocol for multimedia communication sessions. SIP must work with other protocols because it is responsible only for the signaling portion of a communication session. Session Description Protocol (SDP) Describes the content of a multimedia communication session. Real-Time Transport Protocol (RTP) Transmits audio or video content and defines the packet for delivery including the type of content, sequence numbering, time stamping, and delivery monitoring. Has no specific UDP or TCP port number; rather, it has a dynamic range of port numbers, a feature that makes traversing firewalls difficult. Real-Time Transport Control Protocol (RTCP) Monitors QoS in RTP transmissions. Acts as a partner to RTP to package and deliver data but does not transport data. Microsoft notification protocol (MSNP) Is an instant messaging protocol used by Skype. These real-time protocols are designed to be used in both multicast and unicast network services. The quality of a voice service is affected by latency and jitter on a packet network. Therefore, there is a need to ensure QoS for protecting voice from data and to ensure that other critical data applications, which compete with voice, do not lose out on bandwidth. The QoS implementation should also take care of packet loss, delays, and efficient use of bandwidth. Latency is the time delay for a packet to go from the source to the destination and back to the source. Jitter is the variability of latency over time across a network. Jitter should be minimum for real-time applications using voice and video. There are two items that can aid QoS, Class of Service (CoS) and Differentiated Services Code Point (DSCP). CoS is a parameter used in data and voice protocols to differentiate the types of payloads contained in the packet being transmitted. The focus is generally to assign priorities to the data payload or access levels to the telephone call. DSCP is a field in an IP packet that enables different levels of service to be assigned to network traffic. This is achieved by marking each packet on the network with a DSCP code and associating it with the corresponding level of service.

Introduction to WAN Technologies

A WAN is a network that spans a large area, often across multiple geographical locations. WANs typically connect multiple LANs and other networks by using long-range transmission media. This facilitates communication among users and devices in different locations. WANs can be private, such as those built and maintained by large, multinational corporations, or they can be public, such as the Internet. When a WAN includes sites and networks around the world, it is considered a global area network (GAN). Besides geographical coverage, the primary distinction between a LAN and a WAN is the Layer 2 protocol that each uses. With a few exceptions, most Layer 2 protocols were designed to work in either one network type or the other, but not in both. In addition, a company typically owns its own LAN infrastructure and equipment, but it will pay a service provider to connect the company's LANs through the provider's WANs.

Leased data lines

A dedicated line is a telecommunication path that is available 24 hours a day for use by a designated user; dedicated lines and leased lines are essentially the same thing. With dedicated or leased lines, bandwidth availability varies with technology, but is usually between 56 Kbps and 2 Mbps. A company can lease the connection for a fixed fee, typically based on the distance between endpoints. Leasing a line can be advantageous because it guarantees a fixed bandwidth over a dedicated line. While still used in some situations, leased lines are used less frequently with the rise of other high- speed connectivity options. Many companies find it less expensive to deploy VPNs over higher- speed Internet connections than spending on dedicated leased lines.

Digital subscriber line (DSL)

A digital subscriber line (DSL) is a point-to-point, public network access broadband Internet connection method that transmits digital signals over existing phone lines. DSL accomplishes this connection by transporting voice as low-frequency signals and data as high-frequency signals. It has become a popular way to connect small businesses and households to the Internet because of its affordability and high download speeds. However, distance and the quality of lines affect the total bandwidth available to a customer. DSL uses ATM as its underlying technology. DSL operates at Layers 1 and 2 of the OSI model. DSL then uses a tunneling protocol, the Point-to-Point Protocol over Ethernet (PPPoE) to carry data on top of the ATM infrastructure. PPPoE is used to provision DSL services for customers. It provides the ability for the DSL modem to "dial" the provider, authenticate the user, and then compress and encrypt the packets. It also allows the provider to monitor and regulate bandwidth utilization by the customer. If you have an analog (POTS) telephone on the same line as your computer (highly likely in a consumer environment), you will have to use a DSL filter to separate the low-frequency analog voice from the high-frequency DSL data. Connecting the DSL filter improperly to your phone system causes considerable data noise on the telephone, while disrupting service to the device. DSL is commonly referred to as xDSL, denoting the different types of DSL technologies. Asymmetric DSL ADSL is the most popular DSL technology. It allows residential customers to access the Internet and receive phone calls simultaneously. Provides high bandwidth, high-speed transmission over regular telephone lines. Called asymmetric as most of the bandwidth is used for information moving downstream. Widely used where users download more information than what they send. Offers speeds of up to 8 Mbps. Symmetric DSL Unlike ADSL, SDSL provides symmetric connectivity to users. Although it also uses telephone lines, it offers other services on the same line. Often used by small and medium businesses who don't need the service guarantees of frame relay or the higher performance of a leased line. Offers speeds of up to 2.3 Mbps. High-bit-rate DSL Unlike other types of DSL, where downloads speeds tend to be significantly faster than upload speeds, HDSL receives and sends data at the same speed. To accomplish this, it requires two lines that are separate from the normal phone line. HDSL provides transfer rates of 1.54 Mbps, which are comparable to a T1 line. Very high-bit-rate DSL VDSL is an asymmetric solution that provides extremely fast connections over short distances on standard copper phone wiring. VDSL offers speeds up to 52 Mbps.

Domain Names

A domain is a grouping of devices on the Internet or on another network based on the nature of their operations. A domain enables communication between its systems as a unit and other networks on the Internet, instead of maintaining individual connections for each of its systems. Although there are several types of domains, some of the common ones are commercial, governmental, and educational domains. Domains are identified by their unique names; for example, com, gov, and edu. A domain name is a unique name that identifies an entity on the Internet. Also known as site names, domain names appear as part of the complete address of a web resource. They are usually registered by organizations as their website address. A period is used to separate domain name labels, which can have no more than 63 characters. A domain name identifies a collection of devices on the network of a particular domain. A host name is a unique name that identifies a specified device in a network. Therefore, host names are subsets of domain names.

FQDN

A host name combined with the host's domain name forms the node's Fully Qualified Domain Name FQDN (FQDN). A name resolution service maps the FQDN of the node to its IP address so that users can use names instead of IP addresses to communicate with other network nodes and the Internet. FQDNs are written using standard dot-delimited notation, and a dot separates each section of the name. The maximum length of an FQDN is 255 characters; each dot-delimited section can be up to 63 characters long. A network node can have more than one host name assigned to it. Its primary name is its host name; the other names are called canonical names (CNAMEs), or aliases. You can use the hostname command in either Windows or Linux to discover the host part of your computer's DNS name. In Windows, you can use the command ipconfig /all to discover your FQDN. In Linux, you can use the command hostname --fqdn.

Passive optical network (PON)

A passive optical network (PON) is a system that brings optical fiber cabling and signals all or most of the way to the end user. Depending on where the PON terminates, the system can be described as fiber-to-the-curb (FTTC), fiber-to-the-building (FTTB), or fiber-to-the-home (FTTH). A PON consists of an OLT at the service provider's central office and a number of ONUs near end users. A PON reduces the amount of fiber and central office equipment required compared with point-to- point architectures. A passive optical network is a form of fiber optic access network.

Sockets

A socket is an identifier for an application process on a TCP/IP network. It is the combination of the IP address (or host name) and port number. For example, if your web server is installed on 193.44.234.3, the socket for the HTTP process is 193.44.234.3:80. A socket can be open for any protocol, or it can be limited to a specific protocol (as defined in firewall rules). Winsock is the technical specification that defines how Windows network software accesses network services such as TCP/IP. It defines a standard interface between a Windows TCP/IP client application and the underlying TCP/IP protocol stack. The Berkeley sockets interface is the interface between the TCP/IP based network and the network applications that use it. It is the socket interface used by UNIX and Linux.

Wireless WAN

A wireless WAN (WWAN) uses wireless network technology to allow users to check email, surf the WWAN web, and connect to corporate resources accessible within wireless network boundaries. Users connect to a WWAN by using a WWAN card. WWANs use a number of technologies to transfer data and connect to the Internet, such as PPP. Each of these technologies, however, falls into one of three families: GSM/UMTS, cdmaOne/CDMA2000, and WiMAX. The GSM/UMTS and cdmaOne/CDMA2000 protocols started out as mobile phone technologies but now support data transmission. WWAN technologies also use the Wireless Application Protocol (WAP), which enables you to access the Internet from your mobile device. Note: Wireless Application Protocol shares its acronym with wireless access point. The following table compares coverage, speeds, security, and costs of WLANs and WWANs. Coverage Used in a single building of an organization, a home, or a hotspot such as a coffee shop. Usually limited to 100 meters. Used wherever a cellular network provider has coverage—can be regional, national, or even global. Speed Typically 1 to 4 Mbps depending on the number of users that share the connection. Typically 30 to 50 Kbps. Security Susceptible to hacking and interoperability issues between WLANs. Operates on a globally allocated frequency that does not require a license. Tightly regulated frequencies spectrum requiring licenses to operate within the frequency. WWANs incorporate military security technology with a high- level of authentication and encryption. Cost No cost for the wireless connection within the range but a cost to access the Internet via the WLAN access point. The subscription fee is similar to a mobile phone contract. Can be a monthly fee, per minute or per megabyte charge.

Point-to-multipoint connectivity

Another common WAN topology is point-to-multipoint. This is a physical star, in which a central site is the hub, and multiple branch sites are spokes. Logically, a point-to-multipoint connection behaves like a hub. All nodes belong to the same subnet, even though traffic physically passes through the central site. Point-to-multipoint is often used in frame relay networks.

Circuit Switching and Packet Switching

As you know, switching is a technique used for transmitting information over a network to the destination network device. The two main types of switching are circuit switching and packet switching. In circuit switching, one endpoint creates a single path connection to another, depending on the requirement. In circuit switching, the word "circuit" refers to the connection path between endpoints. Once the circuit is established, data is transmitted through that path until the circuit is active. Bandwidth is dedicated to the connection until it is not needed anymore. There is no guarantee that data will be transmitted through the same path through the network in different sessions. The public switched telephone network (PSTN) is an example of a circuit switching network. In packet switching networks, data to be transmitted is broken into small units known as packets that move in sequence through the network. Each packet takes the best route available at any given time rather than following an established circuit path. Each data packet contains all of the routing and sequencing information required to transmit it from one endpoint to another, after which the data is reassembled. Packet switching assumes that a network is constantly changing and adjustments need to be made to compensate for network congestion or broken links. Packet switching is not the best choice for streaming media such as live video and audio feeds. Because all packets do not necessarily arrive at the destination in order, or soon after each other, time-sensitive applications can end up stuttering or delayed, or a streaming connection may drop entirely.

Cell switching networks

Cell switching networks are similar to packet switching networks, except that data is transmitted as fixed-length cells instead of in variable-length packets. If data does not fill up an entire cell, the remainder of the space is filled with blank or filler data until the cell reaches its fixed size. The advantage of cell switching over packet switching is its predictability. Cell switching technologies make it easy to track how much data is moving on a network. Cell Switching Networks

Static IP Address Assignment

Configuring TCP/IP statically on a network requires that an administrator visit each node to manually enter IP address information for that node. If the node moves to a different subnet, the administrator must manually reconfigure the node's TCP/IP information for its new network location. In a large network, configuring TCP/IP statically on each node can be very time consuming, and prone to errors that can potentially disrupt communication on the network. Static addresses are typically assigned only to systems with a dedicated functionality, such as router interfaces, network-attached printers, or servers that host applications on a network.

DHCP

DHCP is a network service that automatically assigns IP addresses and other TCP/IP configuration DHCP information on network nodes configured as DHCP clients. A DHCP server allocates IP addresses to DHCP clients dynamically, and should be configured with at least one DHCP scope. The scope defines the group of IP addresses that a DHCP server can use. When a DHCP server enables the scope, it automatically leases TCP/IP information to DHCP clients for a defined lease period (normally eight days). The scope contains a range of IP addresses and a subnet mask, and can contain other options, such as a default gateway and Domain Name System (DNS) addresses. A scope also needs to specify the duration of the lease, and usage of an IP address after which the node needs to renew the lease with the DHCP server. The DHCP server determines this duration, which can be set for a defined time period or for an unlimited length of time. The Dynamic Host Configuration Protocol version 6 (DHCPv6) is a network protocol for configuring IPv6 hosts with IP addresses, IP prefixes, and other configuration data required to operate in an IPv6 network. It is the IPv6 equivalent of DHCP for IPv4 networks. IPv6 hosts may automatically generate IP addresses internally using stateless address autoconfiguration, or they may be assigned configuration data with DHCPv6.

The DNS Hierarchy

DNS names are built in a hierarchical structure. This allows DNS servers on the Internet to use a minimum number of queries to locate the source of a domain name. The top of the structure— represented by a period—contains root name servers. Below that is the top-level domain name, then the first-level domain name, and so on, until the FQDN for an individual host is complete.

Dial-up connections

Dial-up connections are PSTN connections that use modems, existing phone lines, and long-distance carrier services to provide low-cost, low-bandwidth WAN connectivity and remote network access. Generally limited to 56 Kbps, dial-up connections are sometimes used as backups for higher- bandwidth WAN services. Dial-up connections have two major drawbacks: They are slow and they can have a considerable connection wait time. Despite these limitations, dial-up connections are still used because they provide enough bandwidth for affordable basic Internet connectivity services over the existing telephone infrastructure, especially in geographical areas where other connectivity methods are not available.

Digital network hierarchies

Digital networks have two hierarchical structures that define them: the plesiochronous digital hierarchy (PDH) and the synchronous digital hierarchy (SDH). PDH networks carry data over fiber optic or microwave radio systems. In this type of network, the different parts are ready, but are not synchronized. They have largely replaced PDH for a synchronized network in which the movement of data is highly synchronized along different parts. In SDH, data moves on an optical fiber using LEDs. Basic data transmission occurs at a rate of 155.5 Mbps.

Global system for mobile communications (GSM) and code division multiple access (CDMA)

Global System for Mobile Communications (GSM) and Code Division Multiple Access (CDMA) are both standards that describe protocols for 2G digital cellular networks used by mobile phones. They both use radio signals for voice and data communications. They both have derivatives for use with 3G phones. GSM uses Universal Mobile Telecommunications System (UMTS) and CDMA uses CDMA2000. The major difference between the two is how the carrier connects to the phone and how they turn voice data into radio waves. Enhanced Data rates for GSM Evolution (EDGE) is a 3G standard based on GSM. It is approximately three times as fast as GSM and provides up to 384 Kbps speeds. Mobile phone carriers T-Mobile® and AT&T® use GSM for their cell phone networks, whereas Sprint, Virgin Mobile®, and Verizon Wireless® use the CDMA standard. With 4G phones, GSM and CDMA can be used with the main standards, LTE, and WiMax. Since 4G technologies work using an IP network, the radio signals from both are translated into electronic data for use on the network or the phone.

High speed packet access (HSPA)

HSPA High Speed Packet Access (HSPA) refers to a family of technologies based on the 3GPP Release 5 specification, which offers high data rate services in mobile networks. HSPA offers a downlink speed of up to 14 Mbps and an uplink speed of up to 5.8 Mbps, making it possible for users to upload or download data at a high speed without having to wait for cellular service providers to upgrade their hardware. The HSPA family includes High Speed Downlink Packet Access (HSDPA), High Speed Uplink Packet Access (HSUPA), and HSPA+. HSPA+ uses multicarrier technologies in which multiple 5 MHz carriers are aggregated and a bigger data channel is used for data transmission. This large data channel also decreases latency and provides an increased capacity for bursty traffic, such as web applications. Evolved HSPA also aims to use an all-IP architecture, where all base stations will be connected to the Internet via the ISP's edge routers.

HTTPS

HTTP Secure (HTTPS) is a secure version of HTTP that provides a secure connection between a web browser and a server. HTTPS runs at the Application layer (Layer 7) of the OSI model and the Application layer of the TCP/IP model. It uses port 443 and runs on TCP. HTTPS uses the Secure Sockets Layer (SSL) security protocol to encrypt data. Not all web browsers and servers support HTTPS, though. Note: HTTPS is also referred to as HTTP over SSL.

Host Names

Host Names A host name is a unique name given to a node on a TCP/IP network. It enables users and technicians to recognize the node more easily. A naming service, which is software that runs on one or more nodes, maps the node address to an IP address or a MAC address.

IP Configuration Utilities

IP Configuration Utilities You can use the IP configuration utility for your operating system to view and change TCP/IP configuration information. ipconfig Displays connection-specific DNS suffix, IP address, subnet mask, and default gateway information. Must be run from a command line. To display additional information about a computer's IP configuration, use the command ipconfig /all. Supported on all Windows server systems and client systems. ifconfig Displays the status of currently active network interface devices. Using options, you can dynamically change the status of the interfaces and their IP address. Supported on Linux and UNIX. dhclient Enables you to configure and manage DHCP settings on the network interfaces of a computer. Supported on Linux and UNIX. Note: You can manually release and renew a DHCP lease in Linux by issuing the following command at a command prompt: sudo dhclient -v -r ipconfig Options for DHCP The Windows ipconfig utility provides options for managing dynamic address leases: • ipconfig /release forces the release of an IP address used by a client. • ipconfig /renew requests the renewal of an IP address for a client. The system first attempts to obtain a DHCP address, and if a DHCP server fails to respond, it will switch to APIPA addressing.

ISDN

ISDN is a digital circuit switching technology that carries both voice and data over digital phone lines or PSTN wires. It was the first widely used technique to bring simultaneous voice and data to a customer's home. It uses identifiers similar to a telephone number to establish a demand-dial connection to another ISDN device. ISDN uses digital channels to carry the payload and manage the call. The two types of channels are the "B" (bearer) channel, and the "D" (delta) channel. B channels have a bandwidth of 64 kb. D channels have a bandwidth of either 16 kb or 64 kb, depending on the type of service. ISDN comes in two service types: • Basic Rate Interface (BRI): 2 B channels + one 16 kb D channel for 128 kb throughput • Primary Rate Interface (PRI): 23 B channels + one 64 kb D channel for (near) T1 throughput ISDN was the first successful attempt to digitize the "last mile" of existing copper wire between the customer and the phone company. ISDN works at Layers 1, 2, and 3 of the OSI model. At Layer 1, the frames are 48 bits long and differ in structure depending on the direction of traffic. At Layer 2, the connection is balanced, meaning there is no master/slave relationship where the master (DCE) sets the clock rate and the slave (DTE) must adhere to the DCE's speed. Both the DCE and DTE are treated equally by the protocol. At Layer 3, end-to-end connections are created for user-to-user, circuit-switched or packet-switched functionality similar to X.25. ISDN has largely been replaced by DSL, but can still be found in areas where the distance to the customer premises exceeds DSL's capabilities. Specialized equipment is required to use it, or to adapt it to existing telephone systems. ISDN Hardware ISDN hardware includes terminal equipment (TE), terminal adapters (TAs), network termination (NT) devices, line termination (LT), and exchange termination (ET) equipment. TEs are communications equipment that stations use to accomplish tasks at both ends of a communications link. TAs form the hardware interface between a computer and an ISDN line. NTs are devices that connect the local telephone exchange lines to a customer's telephone or data equipment. ISDN lines terminate at a customer's premises by using an RJ-45 connector in a configuration called a U-interface, which usually connects to a network termination unit (NTU). The NTU can directly connect to ISDN-aware equipment, such as phones or ISDN network interface cards (NICs) in devices. This type of equipment is called Terminal Equipment type 1 (TE1).

The DNS Name Resolution Process

In the DNS process, DNS servers work together as needed to resolve names on behalf of DNS clients. Step 1: Client request The DNS request is passed to a DNS Client service for resolution by using locally cached information on the client. If the DNS request cannot be resolved locally, it sends a DNS query to the DNS resolver. A DNS name resolution request message is generated by the resolver, which is transmitted to the DNS server address specified during configuration. Step 2: Preferred DNS server The DNS server, upon receiving the request, first checks if the requested name is in its DNS cache entries or its local DNS database, and returns the IP address to the client. If there is no match for the requested name, the DNS server sends the request to a root name server asking which DNS server has the entries for the appropriate top-level domain. Step 3: Root name server Upon receiving the request, the root name server reads the top-level domain of that name and sends a message that contains the IP address of the server for that top-level domain. The root name server then sends a reply to the client's DNS server. Step 4: Top-level domain server The client's DNS server contains the IP address of the top-level domain of the requested name. The DNS server then contacts the top-level domain's DNS server to resolve the name. The top-level domain server reads the second-level domain of the requested name, and if it can resolve the name, it sends the desired IP address back to the client's DNS server. Step 5: Other domain servers If the top-level domain cannot resolve the name because of additional levels in the FQDN, it sends the IP address to the second-level DNS server. Step 6: Host name resolution This communication between DNS servers continues until it reaches the level in the DNS hierarchy where a DNS server can resolve the host name. Step 7: Host address The preferred DNS server provides the client with the IP address of the target host. Note: For additional information, check out the LearnTO Follow the DNS Name Resolution Process presentation in the LearnTOs for this course on your CHOICE Course screen. Recursive and Iterative Name Queries There are two kinds of DNS queries: recursive and iterative. • A recursive query is when the client requests that its preferred DNS server find data on other DNS servers. A recursive request starts with the client requesting a name to be resolved to an IP address of its preferred DNS server. If the preferred server cannot resolve the name, it sends a request, on behalf of the client, to another DNS server. • An iterative query occurs when the client requests only the information a server already has in its cache for a particular domain name. If the receiving server cannot resolve the request, it notifies the client, but does not forward the request on to any other server. Recursive queries usually take place between end-user client systems and their preferred DNS servers. Once the recursive query is in process, queries between DNS servers are usually iterative. In most cases, a DNS client will perform a recursive query, and wait upon its DNS server to locate the address for it. The DNS server in turn will either perform an iterative search on the Internet, or will also do a recursive query, asking another DNS server to perform the search for it.

Instant messaging

Instant messaging (IM) is a type of chat that has real-time text transmission over the Internet or a local connection. Short messages are typically transmitted bidirectionally between two parties when a user selects Send. Some IM applications can use push technology to provide real-time text, in which messages are transmitted character by character, as they are typed. Some instant messaging applications include file transfer, selectable hyperlinks, VoIP, or video chat. Instant messaging systems tend to provide connections between specified known users. Depending on the IM protocol, the technical architecture can be peer-to-peer (direct point-to-point transmission) or client-server (a central server retransmits messages from the sender to the receiver).

IMAP4

Internet Message Access Protocol version 4 (IMAP4) is a protocol used for retrieving messages from a mail IMAP4 server. IMAP4 works at the Application layer (Layer 7) of the OSI model and the Application layer of the TCP/IP model. IMAP4 uses port 143 for regular transmissions and port 993 for encrypted transmissions, and it runs on TCP. Though it is similar to POP3, IMAP4 is more powerful and offers several functions. They include: • A user can check an email header and also look for a specific string of characters in the contents of a message before downloading it. • Messages can also remain on the server while the client works with them as if they were local. • Users can search through messages by keywords and choose which messages to download locally. • Messages in the user's mailbox can be marked with different status flags, such as deleted or replied to. The messages and their status flags stay in the mailbox until explicitly removed by the user. • An email message containing multimedia files can be partially downloaded, saving bandwidth. • A user can create, rename, or delete mailboxes on a mail server, and also arrange mailboxes in a hierarchical manner in a folder for email storage. • Unlike POP3, IMAP4 enables users to access folders other than their mailbox. Note: Because IMAP4 is designed to store messages on the server, it is much easier for users to access their email messages—both new and saved—from multiple devices. Note: IMAP was developed at Stanford University in 1986.

End-to-End Communication

It can be helpful to understand how communication traverses the network. Imagine host A sends a message to host Z, which is on a different network. Host A first sends its message by using the MAC address to the local default gateway on its network, in this case a router. That router then sends the message to another router by using an IP address. This next router then sends the message over its network by using a MAC address to another router. The next hop may be another network segment or even the Internet. This process will repeat until the message finally arrives at a router on the destination network. That router will then send the message to host Z by using the MAC address.

Point to point connectivity

Like LANs, WANs use bus, star, mesh, and ring topologies. Some simple WAN implementations also use point-to-point and point-to-multipoint connections. A point-to-point connection is a simple WAN topology that provides a direct connection between two nodes. A point-to-point connection is a type of bus, but with only two nodes on it. It has its own IP subnet, with a /30 subnet mask. The nodes, typically routers, are at either end of the link. This topology is used most commonly by dedicated leased lines and dial-up connections.

Long-term evolution (LTE)

Long-term evolution (LTE) is a radio technology for wireless broadband access. It was introduced in 3GPP Release 8. LTE will be backward compatible with the Global System for Mobile Communications (GSM) and the High Speed Packet Access (HSPA) protocol. This compatibility will enable users to make voice calls and have access to data networks even when they are in areas without LTE coverage. LTE will offer data rates about 100 times faster than 3G networks, a downlink rate that exceeds 100 Mbps, and an uplink rate of more than 50 Mbps.

Metro-ethernet

Metro-Ethernet is a metropolitan area network (MAN) that uses Ethernet standards. Metro- Ethernets can connect LANs and individual users to a WAN or to the Internet. Organizations in large cities can use Metro-Ethernet to connect branch locations or offices to an intranet. A typical Metro-Ethernet has a star network or mesh network topology with servers or routers interconnected through cable or fiber optic media. For example, Comcast Business offers a Metro-Ethernet service for businesses with different locations within a city to communicate with using a wider bandwidth. Metro-Ethernet operates at Layer 2 of the OSI model. The Metro Ethernet Forum does not specify exactly how Metro-Ethernet must be provided. Carriers are free to use pure Ethernet, Synchronous Optical Networking (SONET), MPLS, or a combination of IP-related protocols. Metro-Ethernet topology can be ring, star, or full or partial mesh. Popular implementations of Metro-Ethernet currently offer 1 Gbps over fiber optic cable at a distance of 100 kilometers, or 100 Gbps at a distance of 10 km. Recent developments promise to provide terabit data rates. Metro-Ethernet can be connected to using Layer 2 switches or Layer 3 routers. Companies can extend their VLANs to other locations by using 802.1q VLAN tagging over Metro-Ethernet. Although it is inexpensive and easy to implement, Metro-Ethernet does not currently scale as well as other MPLS implementations. As such, large organizations are using it as their core backbone, particularly for replicating between data centers or for aggregating call center traffic to their data centers. The following diagram shows the use of Metro-Ethernet, MPLS, and the Internet for WAN connectivity.

Multiprotocol label switching (MPLS)

Multiprotocol label switching (MPLS) is a high-performance, multi-service switching technology that is used in packet data networks. It is defined by a set of Internet Engineering Task Force (IETF) specifications that enable Layer 3 devices such as routers to establish and manage network traffic. It ensures faster switching of data as it follows label switching that helps save processing time of packets by the label-switching routers. MPLS is a packet-forwarding technology that uses labels to make its forwarding decisions. The labels are special headers that are 4 bytes long, inserted between the Layer 2 and Layer 3 headers of the packet. A router that has MPLS enabled on its interface is referred to as a label switching router (LSR). It uses the label, rather than the Layer 3 header, to forward the packet to its neighbor. In this way, costly routing table lookups are avoided. Each router rebuilds the label with information for the next hop. MLPS provides the following benefits: • Virtual private networking (VPN). • Traffic engineering (TE). • Quality of Service (QoS). • Any Transport over MPLS (AToM). The labels contain a special designation called the Forward Equivalence Class (FEC). The FEC is applied to a particular stream of packets. It may correspond with a prefix (destination network), or may be based on a class of service (such as IP precedence). A downstream router sends FECs to its upstream neighbor. The upstream neighbor in turn places the correct FEC in the label of a particular packet and passes the packet to the appropriate downstream neighbor. In this way, the path for a particular traffic stream in MPLS is pre-determined. Traffic always follows the same path, with the packets being quickly switched from one router to the next. MPLS has succeeded frame relay and ATM as the dominant private WAN service. While expensive, it has very good reliability (99.9 - 99.99 percent). Companies use it mostly to connect branch offices to each other or to the corporate data center. Most providers have migrated, or are in the process of migrating, their frame relay/ATM customers to MPLS.

Name Resolution

Name resolution is the process of identifying a network node by translating its host or domain name into the corresponding IP address. Several popular name resolution systems are available, but the most prevalent is DNS, which is used on the Internet and practically all TCP/IP based networks. DNS services on an IP network are comparable in functionality to directory assistance in the telephone system. When a client needs the number, it contacts the DNS server, provides the name, and requests the DNS server to look up the number for it.

Static and Dynamic IP Addressing

On a Transmission Control Protocol/Internet Protocol (TCP/IP) network, you can assign IP address information statically to nodes by manually entering IP addressing information on each individual network node. Or, you can assign IP addresses dynamically, by using the Dynamic Host Configuration Protocol (DHCP) service.

Non-Real-Time UC Technologies

Other non-real-time UC technologies include: • Unified messaging: The integration of different electronic messaging and communications media technologies into a single interface. Traditional communications systems deliver messages into different types of stores such as voice mail systems, email servers, and stand-alone fax machines. With unified messaging, all types of messages are stored in one system. • Voice mail: A computer-based system that allows users and subscribers to exchange personal voice messages, and to process transactions relating to individuals, organizations, products and services, by using a telephone. • Email: A method of exchanging digital messages from an author to one or more recipients. Email operates across the Internet or other networks. • SMS: A text messaging service component of a phone, web, or mobile communication system. It uses standardized communications protocols to allow fixed line or mobile phone devices to exchange short text messages. • Fax: A telephonic transmission of scanned printed material, normally to a telephone number connected to a printer or other output device.

Other Real-Time UC Technologies

Other real-time UC technologies include: • Desktop sharing: Technologies and products that allow remote access and remote collaboration on a person's computer desktop through a graphical terminal emulator. • Speech recognition: Technology that translates spoken words into text. • Data sharing: Technologies that allow the sharing of data such as screen sharing or interactive whiteboards. • Call control: Software that decodes addressing information and routes telephone calls from one endpoint to another. It also creates the features that can be used to adapt standard switch operation to the needs of users.

POP3

Post Office Protocol 3 (POP3) is a protocol used to retrieve email messages from a mailbox on a mail server. POP3 works at the Application layer (Layer 7) of the OSI model and the Application layer of the TCP/IP model. POP3 uses port 110 for regular transmissions and port 195 for encrypted transmissions, and it runs on TCP. With POP3, email messages wait in the mailbox on the server until the client retrieves them. The client can start the transfer on a set schedule, or transfer messages manually. Once the client retrieves and downloads the messages, the server deletes them unless the client configures options to leave the messages on the server. The client then works with the locally cached email messages. Because POP3 is designed by default to download messages to the local device and delete them from the email server, it is not the best email protocol to use when users need to access their email from multiple devices. This is because when they use POP3, they end up with their email messages downloaded and split among the devices they use instead of having all their messages in one central location. Or, if they leave their messages on the server, they will have to delete old messages manually to avoid exceeding mailbox size limits, which may also lead to messages being split across multiple devices.

Presence Information

Presence is knowing where a computing device is, usually linked to a person, and if it is available, in real time. A user's client provides presence information via a network connection to a presence service, and can be made available for distribution to other users to convey their availability for communication. Presence information has wide application in many communication services such as instant messaging, phones, and GPS, among others.

Satellite media

Satellite media provide for long-range, global WAN transmissions. A physical link transfers the signal to a satellite link at some point for transmission, and the satellite link then transmits the signal back to a physical link at the other end of the transmission for data delivery. Due to the greater distances the signal must travel, average latency is high, so satellite transmissions do not always work well for real-time applications. Weather conditions also affect the signal. Satellite services provide varying speeds depending on the service agreement. Satellite Internet access is an example of direct, unbounded WAN transmissions. Depending upon the provider, satellite TV customers can choose to receive Internet access through the same satellite dish that receives their TV signals.

SSh

Secure Shell (SSH) is a program that enables a user or an application to log on to another device over a network, execute commands, and manage files. SSH operates at the Application layer (Layer 7) of the OSI model and the Application layer of the TCP/IP model. It uses port 22 and runs on TCP. It creates a shell or session with a remote system, offers strong authentication methods, and ensures that communications are secure over insecure channels. With the SSH slogin command, the login session, including the password, is encrypted and protected against attacks. Secure Shell works with many different operating systems, including Windows, UNIX, and Macintosh. Windows does not provide a native SSH client. You will have to download and install an SSH client such as PuTTY, WinSCP, or Teraterm Pro with the TTSSH extension. Note: SSH is a replacement for the UNIX-based rlogin command, which can also establish a connection with a remote host, but transmits passwords in cleartext. There are two versions of Secure Shell available: SSH1 and SSH2. They are two different protocols and encrypt different parts of a data packet. To authenticate systems, SSH1 employs user keys to identify users; host keys to identify systems; session keys to encrypt communication in a single session; and server keys, which are temporary keys that protect the session key. SSH2 is more secure; it does not use server keys. SSH2 includes a secure replacement for FTP called Secure File Transfer Protocol (SFTP). Because they are different protocol implementations, SSH1 and SSH2 are not compatible with each other. Note: Note that the acronym SFTP is used both for Secure File Transfer Protocol as well as for the now obsolete Simple File Transfer Protocol. All traffic (including passwords) is encrypted by SSH to eliminate connection hijacking, eavesdropping, and other network-level attacks, such as IP source routing, IP spoofing, and DNS spoofing. When you implement SSH with encryption, any attacker who manages to gain access to your network can neither play back the traffic nor hijack the connection. They can only force SSH to disconnect.

Telnet

Telecommunications Network (Telnet) is a terminal emulation protocol that enables users at one site to simulate a session on a remote host as if the terminal were directly attached. It performs this simulation by translating keystrokes from the user's terminal into instructions that the remote host recognizes, and then carrying the output back and displaying it in a format native to the user's terminal. Telnet operates at the Application layer (Layer 7) of the OSI model and the Application layer of the TCP/IP model. It uses port 23 and runs on TCP. You can connect to any host that is running a Telnet daemon or service. Connection-oriented, Telnet handles its own session negotiations and assists network administrators in remote administration such as connecting to a remote server or to a service such as FTP. However, it is not considered a secure protocol, since it transmits in cleartext. Many systems, such as a UNIX host or an IBM® mainframe running TCP/IP, include Telnet daemons. There is also a Telnet server service in older versions of Windows, such as Windows XP and Windows Server® 2003. Telnet is not installed by default in Windows Server 2012 R2. Microsoft provides directions for installing Telnet; you can view them by visiting the URL: http:// technet.microsoft.com/en-us/library/cc770501(WS.10).aspx. Windows includes a basic Telnet client utility. It is installed when you install TCP/IP on your Windows system. It includes Video Terminal 100 (VT100), VT52, and TeleTYpe (TTY) terminal emulation. It does not include the Telnet daemon or service, but the Telnet service can be enabled on Windows Server computers. Telnet is defined in RFC 854, and uses the following defaults: • Uses TCP Port 23; however, you can specify a different port if the host to which you are connecting is configured to use a different port. • Uses 25 lines in the buffer, but you can configure it for up to 399 lines. • Uses VT100 as the default terminal emulation, but some versions allow you to configure your system with VT220, VT52, and TTY terminal emulation support.

DNS Components

The DNS database is divided logically into a hierarchical grouping of domains. It is divided physically into files called zones. The zone files contain the actual IP-to-host name mappings for one or more domains. The zone file is stored on the DNS server that is responsible for resolving host names for the domains contained in the zone. For example, a zone might be responsible for mapping host names to IP addresses within the gcinteriors domain within the .com namespace. Each network node in that domain will have a host record within the domain's zone file. The record includes the node's host name, FQDN, and assigned IP address. For example, a host named 2012srv in the gcinteriors.com domain might have an IP address of 74.43.216.152. That host would have a host record that maps the 2012srv.gcinteriors.com name to the IP address of 74.43.216.152. That host record will appear in the gcinteriors.com zone file on the DNS server that is responsible for the gcinteriors.com domain. Records can be entered into a DNS database either statically or dynamically. A static record is entered manually by an administrator and does not change unless the administrator manually updates it. A network node can request to add a dynamic DNS record that can change dynamically. Dynamic DNS is a method of automatically updating a name server in DNS, often in real time, with the active DNS configuration of its configured host names, addresses, or other information. For example, if a client is using DHCP to get its IP address, each time it leases a new address, it can request an update of its DNS host record.

DNS

The Domain Name System (DNS) is a TCP/IP name resolution service that translates FQDNs into IP addresses. It consists of a system of hierarchical databases that are stored on separate DNS servers on all networks that connect to the Internet. These servers list IP addresses and related device names. Because DNS servers store, maintain, and update databases, they respond to DNS client name resolution requests to translate host names into IP addresses. All these servers work together to resolve FQDNs. On internal networks, a local DNS service can resolve host names without using external DNS servers.

FTP

The File Transfer Protocol (FTP) is a TCP/IP protocol that enables the transfer of files between a user's workstation and a remote host. The FTP daemon or service must be running on the remote host, and an FTP utility may need to be installed on the client. FTP commands must be entered in lowercase and are available both as Windows command-line and UNIX commands. It works on the Application layer (Layer 7) of the OSI model and the Application layer of the TCP/IP model. FTP works on two TCP channels: TCP port 20 for data transfer and TCP port 21 for control commands. These channels work together to enable users to execute commands and transfer data simultaneously. A server-based program answers requests from FTP clients for download. A command line utility enables users to connect to an FTP server and download files. You can initiate an FTP session by entering: ftp FQDN/IP address of remote host You can use several options with the FTP command line utility. -v Prevent remote server command responses being shown . -n Suppress auto-logon at initial connection. -i Disable interactive prompting when transferring multiple files. -d Enable debugging, displaying all commands passed between the FTP client and server. -s Disable wildcard character support. -s: [filename] Run all the FTP commands contained in the [filename] file. -a Allow use of any local interface during data connection binding. -w: [windowsize] Override the default transfer buffer size. Trivial File Transfer Protocol (TFTP) is a simple version of FTP that uses UDP as the transport protocol, and does not require logon to the remote host. As it uses UDP, it does not support error correction but provides for higher data integrity. It is commonly used for bootstrapping and loading applications and not for file transfer. FTP traffic is not encrypted and all transmissions are in clear text. User names, passwords, commands, and data can be read by anyone able to perform packet capture (sniffing) on the network. Most Internet browsers can support FTP in a graphical user interface (GUI) mode. A connection to an FTP site can be made by browsing the Internet, logging on, and connecting. Once connected, you can drag files on and off the FTP site the same way you would from File Explorer. There are also a number of third-party FTP utilities that you can use for connecting to and uploading files to your FTP site. To access most FTP servers, the client needs to connect using a valid user name and password. Some FTP servers allow limited access through an anonymous connection. If anonymous access is disabled on the remote host, users will need login credentials. To use this option, log on using the user name anonymous and enter your email address for the password. When connecting to an FTP server, logging on poses the biggest problems. You need to provide the correct credentials to log on to the FTP server. Most users are granted only read permissions and to upload files, you need to ensure that you have the necessary permissions.

HTTP

The Hypertext Transfer Protocol (HTTP) is a network protocol that works on the Application layer HTTP (Layer 7) of the OSI model and the Application layer of the TCP/IP model to provide web services. HTTP uses port 80 for communicating with web clients and servers and runs on TCP. HTTP enables clients to interact with websites by allowing them to connect to and retrieve web pages from a server. It defines the format and transmission of messages, as well as what actions web servers and clients' browsers should take in response to different commands. A stateless protocol in which each command executes independently of any prior commands, HTTP supports not only persistent connections to web resources to reduce reconnection times, but also pipelining and buffering to help in the transfer process. Web services are application components that communicate through open protocols and can be used by other applications. Web services are based on HTTP and XML. Web services enable any operating system to access the applications you publish. The data is encoded and decoded using XML. SOAP is used to transport the data via open protocols. Programmers can create application components that are reusable and can be accessed as services. Programmers can also use web services to link existing data from various applications to make the data available across all platforms. Note: Because HTTP is stateless, it is difficult to implement websites that react intelligently to user input. This limitation can be overcome with a number of add-on technologies, such as ActiveX®, Java®, JavaScript®, and cookies.

NTP

The Network Time Protocol (NTP) is an Internet protocol that synchronizes the clock times of devices NTP in a network by exchanging time signals. It works on the Application layer (Layer 7) of the OSI model and the Application layer of the TCP/IP model. Synchronization is done to the millisecond against the U.S. Naval Observatory master clocks. Running continuously in the background on a device, NTP sends periodic time requests to servers to obtain the server time stamp and then adjusts the client's clock based on the server time stamp received. Implementations send and receive time stamps by using UDP on port number 123. Note: The master time clocks are located in Washington, DC, and Colorado Springs, Colorado.

The optical carrier system

The Optical Carrier x (OCx) standard specifies the bandwidth for fiber optic transmissions. It is a channelized technology based on the same 64 Kbps channel as DSH but with a base rate of 810 channels. The OCx standard is open-ended, enabling manufacturers to add specifications as they develop hardware that supports faster transmission speeds. OCx specifications correspond with the data rates of SONET. As one OC channel corresponds to a data rate of 51.84 Mbps, using multiple channels increases the rate by 51.84 Mbps per channel. OC1 1 OC channel with a data rate of 51.84 Mbps. OC3 3 OC channels with a data rate of 155.52 Mbps. OC9 9 OC channels with a data rate of 466.56 Mbps. OC12 12 OC channels with a data rate of 622.08 Mbps. OC18 18 OC channels with a data rate of 933.15 Mbps. OC24 24 OC channels with a data rate of 1.24 Gbps. OC36 36 OC channels with a data rate of 1.87 Gbps. OC192 192 OC channels with a data rate of 9.95 Gbps. The passive optical network (PON) is a point-to-multipoint optical network that is used for broadcast transmissions using optical systems. As the optical transmission requires no power or active electronic parts when the signal passes through the network, it is referred to as passive. A PON contains a central office node, known as optical line termination (OLT) and optical network units (ONUs) near end users. An OLT can connect to up to 32 ONUs.

SMTP

The Simple Mail Transfer Protocol (SMTP) is a communications protocol for formatting and sending SMTP email messages from a client to a server or between servers. It works at the Application layer (Layer 7) of the OSI model and the Application layer of the TCP/IP model. Using port 25 or 587 for standard communications and port 465 for encrypted communications, SMTP runs on TCP. SMTP uses a store-and-forward process. In SMTP, the sender starts the transfer. SMTP can store a message until the receiving device comes online. At that point, it contacts the device and hands off the message. If all devices are online, the message is sent quickly. An SMTP message consists of a header and a content section. The header, or envelope, contains the delivery information of the message and uses a colon (:) as a separator character. The content portion contains the message text, which is a sequence of ASCII characters. Because of SMTP's store-and-forward capability, it is used to send data through unreliable wide area network (WAN) links if delivery time is not critical. Data is sent to the endpoint and continues to hop from server to server until it eventually reaches its destination. SMTP has a few limitations. The first one is related to the size of messages. Messages that are more than 64 KB cannot be handled by some older implementations. Another limitation involves timeouts. If the client and server timeouts are different, one of the systems may give up when the other is still busy, resulting in termination of the connection unexpectedly. Sometimes SMTP may also trigger infinite mail storms. For example, consider host 1 with Mailing List A containing a few entries and host 2 with Mailing List B containing both its own entries and that of Mailing List A. In such a case, email sent to Mailing List A and copied to Mailing List B could trigger sending multiple copies of the same email message to the same set of recipients. Furthermore, if host 1 fails when mail is being forwarded, host 2 will try resending it to host 1. This generates a heavy amount of traffic on the network. Extended SMTP (ESMTP) extends the capabilities of SMTP and helps to overcome some of these limitations.

The TCP/IP Protocol Stack

The TCP/IP protocol stack is the collection of protocols that work together to provide communications on IP-based networks such as the Internet. To send data over a TCP/IP network requires four steps or layers: • The Application layer encodes the data being sent. • The Transport layer splits the data into manageable chunks and adds port number information. • The Internet layer adds IP addresses stating where the data is from and where it is going. • The Network Access layer adds MAC address information to specify which hardware device the message came from and which hardware device the message is going to. The terms protocol stack and protocol suite are often used interchangeably. But, you also might find that the terms can be used to convey subtle differences, such as the stack being a complete set of protocols and the suite being a subset of the stack, often supplied by a particular vendor, or the suite being the definition of the protocols and the stack being the software implementation of the suite.

WAN Termination Equipment

The WAN Physical layer describes the interface between the data termination equipment (DTE) and the data communications equipment (DCE). In most cases, the DCE belongs to the service provider, and the DTE is the customer's device. The DCE will almost always be a modem or CSU/DSU that is installed on the customer's premises.

Legacy Name Resolution Methods

The most common name resolution methods that have been used in the past are the HOSTS file, NetBIOS, and Windows Internet Name Server (WINS). A HOSTS file is a plaintext file configured on a client device containing a list of IP addresses and their associated host names, separated by at least one space. Comments may be included after the host name if preceded by the # symbol and separated from the host name by at least one space. The hosts file exists in both Linux and Windows. It can be edited in both operating systems to manually map host names to IP addresses. The HOSTS file provides an alternative method of host name resolution. An external client can use a HOSTS file to resolve names on your internal network without accessing your internal DNS server. You have to manually configure each host name entry in a HOSTS file. A device will always consult its HOSTS file first to see if the desired IP address is there before querying a DNS server. The HOSTS file requires a lot of maintenance, so it is recommended that you use it only when other methods of host name resolution are not supported, or temporarily unavailable for troubleshooting purposes. NetBIOS is a simple, broadcast-based naming service. A NetBIOS name can be any combination of alphanumeric characters excluding spaces and the following characters: / : * ? " ; \ |. The length of the name cannot exceed 15 characters. The 16th character is reserved. WINS is an implementation of the NetBIOS Name Service (NBNS), a name server and service for NetBIOS computer names. It is an older type of naming service used on Windows-based networks.

UC hardware

The most prevalent types of hardware devices used to provide UC services include servers, devices, and gateways. UC servers Unified communications servers provide the voice, video, fax, messaging, and more services that users will use. It can be a proprietary appliance bundled with the desired features and powered Ethernet switchports, or it could also be a generic server with the appropriate hardware specifications and the UC software product installed on it. As with many UC products, it can also be a virtual server in a provider's cloud. UC devices Unified communications devices are client-side devices that allow end users to use unified communications services. They can include IP or video- enabled phones, headsets, webcams and busy lights, and other meeting room devices. UC gateways Unified communications gateways connect your private UC network with a public network. It is the interface that allows users to connect with the outside world. It also allows mobile users to connect from the outside into the private network. The public network can be the PSTN, the Internet, or a cellular provider—any network that will extend the reach of your UC system to everyone else. The data that your UC gateway processes will be voice, video, conferencing, collaboration, messaging—any service or protocol that you have chosen to implement in your UC system. The gateway can be a dedicated appliance or a generic server with software installed. It can also be a service in a provider's cloud.

pathping

The pathping command provides information about latency and packet loss on a network. pathping combines the functionality of the ping and tracert commands. Similar to ping, pathping sends multiple ICMP echo request messages to each router between two hosts over a period of time, and then displays results based on the number of packets returned by each router. It is similar to tracert as it identifies the routers that are on the path. In the output, it also displays the path to the remote host over a maximum of 30 hops. In addition, it displays details of packet transfer between the hosts in a time span of over 25 seconds, and the system names and their IP addresses. pathping can be used to isolate a router or subnet with issues as it can display the degree of packet loss at any given router or link. pathping Options The pathping command can be used with different options that allow you to customize the results of the command to your network requirements. -h maximum hops Specify the maximum number of hops to locate a destination. -i address Specify a source IP address. -n Specify that host name resolution can be skipped. -4 address Specify the IPv4 addresses that are to be used. -6 address Specify the IPv6 addresses that are to be used.

Ping

The ping command is used to verify the network connectivity of a device, and also to check to see if the target device is active. It verifies the IP address, host name, and reachability of the remote device by using and listening for echo replies. ping uses ICMP to check the connections with remote hosts by sending out echo requests as ICMP ECHO_REQUEST packets to the host whose name or IP address you specify on the command line. ping listens for reply packets. The ping command can be used in both Windows and Linux, as well as FreeBSD (which is what Mac OS X is built on), but the syntax options are not the same. For a list of options in Windows, enter ping /?. For Linux and FreeBSD, enter man ping. Syntax The syntax of the ping command is: ping target The target variable specifies the IP address or DNS name of a device on the network. ping uses the DNS setting to resolve the DNS name into an IP address. Options You can ping a device or an IP address. You can also ping the loopback address (127.0.0.1) to test whether TCP/IP has initialized on an individual device. If the computer has a default gateway, you can ping remote systems. To list other options for the ping command, enter ping /? at the command prompt. Some of the options include setting the packet size, changing the time-to-live (TTL) value, and specifying how many times to ping the host. Packet size By default, data packets are sent as 32 bytes. You can specify a larger size to test response time, the maximum size being 65,500 bytes. To change the packet size, use the -l option followed by the packet length. ping target [-l size] TTL A value that determines how many hops an IP packet can traverse before being discarded. Each hop is a router that was crossed. ping target [-i TTL] Packet count Specifies the number of packets with which a remote host is pinged. The default is four packets. You can specify a higher number of packets with the -n option. ping target [-n packet count] Continuous ping Pings the specified host until the command is interrupted by pressing Ctrl+C. ping target -t IPv6 Ping using IPv6 ping target -6 Note: To ping IPv6 on Linux, use the ping6 command. ICMP Blocking As a security measure, some public Internet hosts and routers might be configured to block incoming packets that are generated by the ping command. (They might also block packets from other TCP/IP diagnostic utilities such as the tracert command.) It is not the ping command that is blocked but the ICMP traffic that is blocked. Pinging these hosts will fail even if the host is online. Keep this in mind when you try to ping large public Internet sites; if you are trying to determine if one of these sites is up and running, a better method is simply to use a web browser to connect to the site directly.

tracert

The tracert command determines the route data takes to get to a particular destination. The node sends out messages with incrementally increasing TTL values, which cause the packets to expire at each successive router in the path. Internet Control Message Protocol (ICMP) "Time Exceeded" messages are then sent back from the routers to the node running tracert. Each time a packet is sent, the TTL value is reduced before the packet is forwarded, thus allowing TTL to count how many hops it is away from the destination. traceroute is the Linux equivalent of the tracert command, which is Windows-based. Note: If you run the tracert command repeatedly for the same destination, you will normally see different results after a relatively short period of time. This is because TCP/IP is auto- correcting and takes the fastest route possible across the global network of Internet routers. Network Firewalls If a network firewall is configured to not allow a tracert or ping through, you might not be able to trace the route all the way to the end; it might appear to end at the firewall. If you get the message "Destination Unreachable," a router is not able to figure out how to get to the next destination. Even though it does not tell you what is wrong, it alerts you to the router where the problem is occurring. You can use various options with the tracert command. tracert Options -d If you are having trouble resolving host names when using tracert, use the -d option to prevent tracert from trying to resolve host names. It also speeds up response time because it is not spending time resolving host names. -h max_hops The default number of hops tracert will attempt to reach is 30. Using the -h option, you can specify more or fewer hops for it to check. -j host-list You can use the -j option to force the outgoing datagram to pass through a specific router. -w timeout If many of your responses on the tracert are timing out, by using the -w option, you can increase the number of milliseconds to wait before continuing. If, after increasing the value, destinations are then reachable, you probably have a bandwidth issue to resolve.

Types of DNS Servers

There are different types of DNS servers, including default DNS servers and authoritative name servers. As with the default gateway, you can configure default DNS servers that match host names to IP addresses. These specialized servers maintain databases of IP addresses and their corresponding domain names. For example, when you type www.yahoo.com into your browser address bar, the name is resolved by DNS to the IP addresses of the Yahoo server farm. You can configure default DNS servers statically or automatically. • If you are configuring static IP addresses, include the IP addresses of the default DNS servers as you configure each client. • If you are using DHCP, use the DHCP scope options to specify the IP addresses of the default DNS servers. An authoritative name server (ANS) is a DNS server that possesses an actual copy of the records for a zone, as opposed to just caching a lookup from another DNS server. It responds to name-related queries in one or more zones. The most important function of the ANS is delegation, which means that part of a domain is delegated to other DNS servers. The start of authority (SOA) is the first DNS server to create the zone. It is typically the primary DNS server, meaning that it holds the only writable copy of the zone. Additional authoritative servers can be secondary DNS servers, meaning that they hold read-only copies that they obtain from the primary (their master). Primary and Secondary DNS Servers When configuring a client's DNS settings, it is common to specify both a primary and a secondary DNS server to provide a more reliable name resolution process. When two DNS servers are listed in a client's TCP/IP settings, the client queries the primary server first. If the primary server does not answer, the client queries the secondary server. If the primary server returns a "Name Not Found" message, the query is over and the client does not query the secondary server. This is because both DNS servers can do recursive and iterative queries, and both primary and secondary servers should be able to contact the same resources. If one cannot access the resource, the other will not be able to either.

VoIP software

There are various scenarios where VoIP can be used, such as phone calls, web conferences, or enabling voice mail and faxes to be delivered through email. These capabilities are achieved through different VoIP applications that are available. Microsoft Lync Microsoft Lync is a messaging application for smartphones and desktops that allows users to connect to other users and provides presence information, instant messaging, voice calls, video calls, and online meetings. Users can now also connect to Skype users. Skype Skype is a messaging application for smartphones and desktops that allows users to connect to other users and provides presence information, instant messaging, voice calls, and video calls. Users can now also connect to Microsoft Lync users. Google Hangouts Google Hangouts is a messaging application for smartphones and desktops that allows users to connect to other users and provides presence information, instant messaging, photos, voice calls, and video calls. GoToMeeting GoToMeeting is an online meeting application for smartphones and desktops that provides the ability to participate in online meetings, share desktops, and participate in video conferencing. Viber Viber is a messaging application for smartphones and desktops that allows users to connect to other users and provides presence information, instant messaging, photos, voice calls, and video calls. OnSIP OnSIP is a business-level VoIP service that provides an enterprise class phone system. Ekiga Ekiga is a phone and video conferencing application for desktops that allows users to connect to other users and provides instant messaging, voice calls, and video calls. Vonage Vonage is a telephone application and service for smartphones and desktops that allows users to make voice calls. Tango Tango is a messaging application for smartphones that allows users to connect to other users and provides presence information, social networking, instant messaging, photos, music, games, voice calls, and video calls. ShoreTel Sky ShoreTel Sky is a business-level VoIP service that provides an enterprise class phone system.

Dial-up and broadband connectivity

Three of the most common methods used to provide Internet connectivity to customers are dial-up, broadband DSL, and broadband cable. Dial-up lines Dial-up lines are local loop public switched telephone network (PSTN) connections that use modems, existing phone lines, and long-distance carrier services to provide low-cost, low-bandwidth WAN connectivity, and remote network access. PSTN is also known as the plain old telephone system (POTS). As a data carrier, the PSTN operates at Layers 1 and 2 of the OSI model. The Layer 2 protocol that manages the call is the Point-to-Point Protocol (PPP). PSTN is a telephone system that carries analog voice data. PSTN offers traditional telephone services to residences and establishments. PSTN includes telephones and fax machines that set up temporary but continuous connections. During a call, a circuit is established between two users and is kept open even during periods of silence. This provides guaranteed Quality of Service (QoS) but uses bandwidth inefficiently. Broadband DSL Broadband DSL offers high-speed Internet access with much higher speeds than dial-up connections. Telephone companies use DSL to offer data, video, and voice services over existing phone lines. Broadband cable Broadband cable offers high-speed Internet access with higher speeds than dial-up connections and broadband DSL. It also allows the simultaneous use of a telephone line.

Types of DNS Records

Types of DNS Records Different types of DNS records are available that serve specific purposes. Types of DNS Records Address (A) Maps a host name to its IP address by using a 32-bit IPv4 address. IPv6 address (AAAA) Maps a host name to its IP address by using a 128-bit IPv6 address. Canonical name (CNAME) Maps multiple canonical names (aliases) to an A record. Mail Exchanger (MX) Maps a domain name to a email server list. Name Server (NS) Assigns a DNS zone to access the given authoritative name servers Pointer (PTR) Maps an IP address to the host name for the purpose of reverse lookup. Start of Authority (SOA) Specifies authoritative information about a DNS zone. Service Locator (SRV) Specifies a generic service location record for newer protocols.

Unified communication (UC) technologies

Unified communication (UC) technologies are a group of integrated real-time communication services and non-real-time services that provides a consistent user experience across multiple devices and media types. Real-time communication services and products can be integrated with non-real-time services and products, with the ultimate result being that a user can send a message on one medium and receive the same communication on another medium. For example, if you receive a voice-mail message, you might choose to access it through email or a mobile phone. Medianets enable video teleconferencing (VTC or VC) over ISDN or other high-speed broadband connections using IP and SIP protocols. This technology enables IP and SIP protocols to carry both video and audio data so that a conference between multiple locations can be carried out. Real-time UC technologies include: • Voice over data systems, such as Voice over IP (VoIP). • Video conferencing. • Presence information. • Instant messaging. • Desktop sharing. • Data sharing. • Speech recognition. Non-real-time UC technologies include: • Voice mail • Email • SMS (text) messaging • Fax messaging

Video conferencing

Video conferencing uses telecommunication technologies that allow two or more locations to communicate by simultaneous two-way video and audio transmissions. Video conferencing is different from videophone calls in that it's designed to serve a conference or multiple locations rather than individuals. Video conferencing uses digital compression of audio and video streams in real time. The hardware or software that performs compression is called a codec.

Virtual Circuit Switching

Virtual circuit switching is a switching technique to transfer packets on logical circuits that do not have physical resources, such as frequencies or time slots allocated. This technique merges both packet and circuit switching techniques to its advantage. These logical paths are assigned to identities, rather than to physical locations, and can be either permanent or switched. Each of the packets carries a virtual circuit identifier (VCI) that is local to a link and updated by each switch on the path, from the source to the destination of the packet. There are two types of virtual circuits: permanent virtual circuits (PVCs) and switched virtual circuits (SVCs). Permanent PVCs are usually associated with leased lines. They connect two endpoints and are always on, which is why they are referred to as permanent. When a PVC is established, it is manually built and maintained by a telephone company (telco). The telco identifies the endpoints with a data link connection identifier (DLCI). PVCs provide a fast, reliable connection between endpoints because the connection is always on. Customers pay a fixed monthly fee per connection. Switched SVCs are associated with dial-up connections. SVCs provide more flexibility than PVCs and allow a single connection to an endpoint to be connected to multiple endpoints as needed. When a network device attempts to connect to a WAN, an SVC is requested and the carrier establishes the connection. Customers typically pay by connection time (like a long-distance phone call) and the monthly charge is less than that of a PVC. SVCs are useful when you need a part-time connection. But keep in mind that connection time can be slow, and if usage increases, so can an SVC's cost.

Voice-over-Data Systems

Voice-over-data systems are communication systems that replace traditional telephone links by transmitting analog voice communications over digital WAN technologies. Digital WANs provide more bandwidth than analog phone systems, and there is no long-distance service cost involved. Because voice communications are time-sensitive, the voice-over-data system must ensure that packets arrive complete and in sequence. In a voice-over-data system, voice software interfaces with an analog voice device, such as a microphone, to convert the analog voice into a data signal and to translate the dialing destination into a network address. Voice-over-data systems have included voice-over-frame-relay (VoFR), voice-over-ATM (VoATM), and voice over IP (VoIP).

WAN Devices

WAN connectivity devices enable you to connect LANs together. The most common WAN devices include those described in the following table. Modem A modem enables digital data to be sent over an analog medium such as a telephone wire or cable provider's line. Digital signals are converted into an analog format suitable for transmission through analog carriers and then restored to digital format on the receiving end. The three main types of modems that you will encounter are: • DSL modems: hardware devices that connect subscribers to a telephone line that provides the digital subscriber line service for connectivity to the Internet. This connectivity is sometimes referred to as DSL broadband. DSL speed varies widely and depends on several factors. It is best to communicate with the service providers to determine the bandwidth available for a particular location. • Cable modems: hardware devices that connect subscribers to the Internet service provider's (ISP's) cable systems. Service providers use a cable modem to connect the subscriber's device to the Internet by using twisted pair cabling and a network port or USB connection. On the other end, the cable modem connects to the wall jack by using coaxial cabling. Most cable companies provide access for up to 25 or 50 Mbps. If the cable system is fiber-based, speeds might reach up to 1 Gbps. Cable modems operate at the Physical (Layer 1) and Data Link (Layer 2) layers of the OSI model. • Dial-up modems: communication devices that convert a computer's digital signals into analog signals before transmission over telephone lines. A dial-up modem can be either internal or external. Internal dial-up modems exist as part of the motherboard and use the device's power supply; external dial-up modems connect via the serial or USB port as separate expansion boxes. Unlike internal dial-up modems, external modems require a separate power supply. The disadvantage of a dial-up modem is that it is slow when compared to broadband modems. Access server An access server manages dial-in and dial-out user communications. It can have a mixture of analog and digital interfaces and support hundreds of simultaneous users. Network access servers function as control points for roaming and remote users so that they can access internal resources (or connect to an ISP) from external locations. WAN switch A WAN switch is a multiport internetworking device that normally switches traffic and operates at the Data Link layer (Layer 2) of the OSI model. WAN switches can share bandwidth among allocated service priorities, recover from outages, and provide network design and management systems. CSU/DSU A Channel Service Unit/Data Service Unit (CSU/DSU) is a combination of two WAN connectivity devices that work together to connect a digital WAN line with a customer's LAN. The DSU receives the signal from the LAN and passes it to the CSU. The CSU converts the signal format to make it compatible with the Digital Data Service (DDS) on the WAN line. ISDN terminal adapter An Integrated Services for Digital Network (ISDN) terminal adapter is similar to a modem in that it joins Basic Rate Interface (BRI) connections to different physical interfaces on a router. Unlike a modem, it does not convert between analog and digital signaling. Note: You will discuss ISDN in greater depth later in this lesson.

Wireless interoperability for microwave access (WiMAX)

Wireless Interoperability for Microwave Access (WiMAX) is a packet-based wireless telecommunication technology that provides wireless broadband access over long distances. Based on the IEEE 802.16 standard, it is intended for wireless MANs. WiMAX provides fixed as well as mobile broadband access. It covers a range of about 30 miles for fixed stations and 3 to 10 miles for mobile stations. WiMAX also provides LoS and non-line-of-sight (NLoS) communication, and can provide connection speeds of about 70 Mbps. WiMAX operates in the wireless frequency ranges of between 2 and 11 GHz of the wireless spectrum. Note: WiMAX was created by an organization known as the WiMAX Forum. WiMAX offers two different services: LoS and NLoS. • Line-of-sight (LoS): Signals travel over a direct path from a transmitter to a receiver. • Non-line-of-sight (NLoS): Signals reach a receiver through reflections and diffractions. WiMAX is of two types: fixed and mobile. Fixed Optimized for fixed applications in LoS and NLoS environments. The main disadvantage of fixed WiMAX is its difficulty to compete with established wired technologies such as DSL in places where the wired telecommunication infrastructure is well developed. Mobile Optimized for portable and mobile applications in an NLoS environment. Mobile WiMAX includes additional features such as power management, handoff, frequency reuse, channel bandwidth scalability, and better NLoS performance and indoor penetration.

X.25 Switched networks

X.25 was the first widely implemented packet-switching network technology. It was developed in the 1970s before the OSI model, but its functionality corresponds to Layers 1, 2, and 3 of the OSI model. X.25 was designed to move data across the less-than-reliable analog long-distance public carrier lines of the time. Its emphasis on reliable delivery introduced a lot of overhead to the network, which reduced performance.