Communication Protocols

 

Communication Protocols

Approximately 30 years ago, communication protocols were developed so that individual stations could be connected to form a local area network (LAN). This group of computers and other devices, dispersed over a relatively limited area and connected by a communications link, enabled any station to interact with any other on the network. These networks allowed stations to share resources, such as laser printers and large hard disks. This chapter and Chapter 2 discuss the communication protocols that became a set of rules or standards designed to enable these stations to connect with one another and to exchange information. The protocol generally accepted for standardizing overall computer communications is a seven-layer set of hardware and software guidelines known as the Open Systems Interconnection (OSI) model. Before one can accurately define, implement, and test (hack into) security policies, it is imperative to have a solid understanding of these protocols. These chapters will cover the foundation of rules as they pertain to TCP/IP, ARP, UDP, ICMP, IPX, SPX, NetBIOS, and NetBEUI.

 

A Brief History of the Internet

During the 1960s, the U.S. Department of Defense’s Advanced Research Projects Agency (ARPA, later called DARPA) began an experimental wide area network (WAN) that spanned the United States. Called ARPANET, its original goal was to enable government affiliations, educational institutions, and research laboratories to share computing resources and to collaborate via file sharing and electronic mail. It didn’t take long, however, for DARPA to realize the advantages of ARPANET and the possibilities of providing these network links across the world. By the 1970s, DARPA continued aggressively funding and conducting research on ARPANET, to motivate the development of the framework for a community of networking technologies. The result of this framework was the Transmission Control Protocol/Internet Protocol (TCP/IP) suite. (A protocol is basically defined as a set of rules for communication over a computer network.) To increase acceptance of the use of protocols, DARPA disclosed a less expensive implementation of this project to the computing community. The University of California at Berkeley’s Berkeley Software Design (BSD) UNIX system was a primary target for this experiment. DARPA funded a company called Bolt Beranek and Newman, Inc. (BBN) to help develop the TCP/IP suite on BSD UNIX. This new technology came about during a time when many establishments were in the process of developing local area network technologies to connect two or more computers on a common site. By January 1983, all of the computers connected on ARPANET were running the new TCP/IP suite for communications. In 1989, Conseil Europeén pour la Recherche Nucléaire (CERN), Europe’s high-energy physics laboratory, invented the World Wide Web (WWW). CERN’s primary objective for this development was to give physicists around the globe the means to communicate more efficiently using hypertext. At that time, hypertext only included document text with command tags, which were enclosed in <angle brackets>. The tags were used to markup the document’s logical elements, for example, the title, headers and paragraphs. This soon developed into a language by which programmers could generate viewable pages of information called Hypertext Markup Language (HTML). In February 1993, the National Center for Supercomputing Applications at the University

of Illinois (NCSA) published the legendary browser, Mosaic. With this browser, users could view HTML graphically presented pages of information. At the time, there were approximately 50 Web servers providing archives for viewable HTML. Nine months later, the number had grown to more than 500. Approximately one year later, there were more than 10,000 Web servers in 84 countries comprising the World Wide Web, all running on ARPANET’s backbone called the Internet. Today, the Internet provides a means of collaboration for millions of hosts across the world. The current backbone infrastructure of the Internet can carry a volume well over 45 megabits per second (Mb), about one thousand times the bandwidth of the original ARPANET. (Bandwidth is a measure of the amount of traffic a media can handle at one time. In digital communication, this describes the amount of data that can be transmitted over a communication line at bits per second, commonly abbreviated as bps.)

 

Internet Protocol (IP)

The Internet Protocol (IP) part of the TCP/IP suite is a four-layer model (see Figure 1.1). IP is designed to interconnect networks to form an Internet to pass data back and forth. IP contains addressing and control information that enables packets to be routed through this Internet. (A packet is defined as a logical grouping of information, which includes a header containing control information and, usually, user data.) The equipment—that is, routers—that encounter these packets, strip off and examine the headers that contain the sensitive routing information. These headers are modified and reformulated as a packet to be passed along.

Packet headers contain control information (route specifications) and user data. This information can be copied, modified, and/or spoofed (masqueraded) by hackers.

One of the IP’s primary functions is to provide a permanently established connection (termed connectionless), unreliable, best-effort delivery of datagrams through an Internetwork. Datagrams can be described as a logical grouping of information sent as a network layer unit over a communication medium. IP datagrams are the primary information units in the Internet. Another of IP’s principal responsibilities is the fragmentation and reassembly of datagrams to support links with different transmission sizes.

Figure 1.1 The four-layer TCP/IP model.

 

 

 

 

 

Figure 1.2 An IP packet.

 

During an analysis session, or sniffer capture, it is necessary to differentiate between different types of packet captures. The following describes the IP packet and the 14 fields therein, as illustrated in Figure 1.2.

 

 

 

 

 

Version. The IP version currently used.

IP Header Length (Length). The datagram header length in 32-bit words.

Type-of-Service (ToS). How the upper-layer protocol (the layer immediately above, such as transport protocols like TCP and UDP) intends to handle the current datagram and assign a level of importance.

Total Length. The length, in bytes, of the entire IP packet.

Identification. An integer used to help piece together datagram fragments.

Flag. A 3-bit field, where the first bit specifies whether the packet can be fragmented. The second bit indicates whether the packet is the last fragment in a series. The final bit is not used at this time.

Fragment Offset. The location of the fragment’s data, relative to the opening data in the original datagram. This allows for proper reconstruction of the original datagram.

Time-to-Live (TTL). A counter that decrements to zero to keep packets from endlessly looping. At the zero mark, the packet is dropped.

Protocol. Indicates the upper-layer protocol receiving the incoming packets.

Header Checksum. Ensures the integrity of the IP header.

Source Address/Destination Address. The sending and receiving nodes (station, server, and/or router).

Options. Typically, contains security options.

Data. Upper-layer information.

Key fields to note include the Source Address, Destination Address, Options, and Data.

 

Now let’s look at actual sniffer snapshots of IP Headers in Figures 1.3a and 1.3b to compare with the fields in the previous figure.

Figure 1.3a

Extracted during the transmission of an Internet Control Message Protocol (ICMP) ping test (ICMP is explained later in this chapter).

 

 

 

 

 

 

 

Figure 1.3b

Extracted during the transmission of a NetBIOS User Datagram Protocol (UDP) session request (these protocols are described later in this chapter and in Chapter 2).

 

 

 

 

 

 

 

 

IP Datagrams, Encapsulation, Size, and Fragmentation

IP datagrams are the very basic, or fundamental, transfer unit of the Internet. An IP datagram is the unit of data commuted between IP modules. IP datagrams have headers with fields that provide routing information used by infrastructure equipment such as routers (see Figure 1.4).

 

Figure 1.4 An IP datagram.

 

 

 

Be aware that the data in a packet is not really a concern for the IP. Instead, IP is concerned with the control information as it pertains to the upper-layer protocol. This information is stored in the IP header, which tries to deliver the datagram to its destination on the local network or over the Internet. To understand this relationship, think of IP as the method and the datagram as the means.

The IP header is the primary field for gathering information, as well as for gaining control.

 

It is important to understand the methods a datagram uses to travel across networks. To sufficiently travel across the Internet, over physical media, we want some guarantee that each datagram travels in a physical frame. The process of a datagram traveling across media in a frame is called encapsulation. Now, let’s take a look at an actual traveling datagram scenario to further explain these traveling datagram methods (see Figure 1.5). This example includes corporate connectivity between three branch offices, over the Internet, linking Ethernet, Token Ring, and FDDI (Fiber Distributed Data Interface) or fiber redundant Token Ring networks.

Figure 1.5 Real-world example of a traveling datagram.

An ideal situation is one where an entire IP datagram fits into a frame; and the network it is traveling across supports that particular transfer size. But as we all know ideal situations are rare. One problem with our traveling datagram is that networks enforce a maximum transfer unit (MTU) size, or limit, on the size of transfer. To further confuse the issue, different types of networks enforce their own MTU; for example, Ethernet has an MTU of 1500, FDDI uses 4470 MTU, and so on. When datagrams traveling in frames cross network types with different specified size limits, routers must sometimes divide the datagram to accommodate a smaller MTU. This process is called fragmentation.

Routers provide the fragmentation process of datagrams, and as such, become vulnerable to passive and intrusive attacks.

IP Addresses, Classes, Subnet Masks Communicating on the Internet would be almost impossible if a system of unique addressing were not used. To prevent the use of duplicate addresses, routing between nodes is based on addresses assigned from a pool of classes, or range of available addresses, from the InterNetwork Information Center (InterNIC). InterNIC assigns and controls all network addresses used over the Internet by assigning addresses in three classes (A, B, and C), which consist of 32-bit numbers. By default, the usable bits for Classes A, B, and C are 8, 16, and 24 respectively. Addresses from this pool have been assigned and utilized since the 1970s, and they include the ranges shown in Figure 1.6; an example of an IP address is shown in Figure 1.7.

Figure 1.6 IP address chart by class.

Figure 1.7 IP address example with four octets.

The first octet (206) indicates a Class C (Internet-assigned) IP address range with the format Network.Network.Network.Host with a standard mask binary indicating 255.255.255.0. This means that we have 8 bits in the last octet for hosts. The 8 bits that make up the last, or fourth, octet are understood by infrastructure equipment such as routers and software in the following manner:

In this example of a full Class C, we only have 254 usable IP addresses for hosts; 0 and 255 cannot be used as host addresses because the network number is 0 and the broadcast address is 255. With the abundant utilization of Class B address space and the flooding of requested Class C addresses, a Classless Interdomain Routing (CIR) system was introduced in the early 1990s. Basically, a route is no longer an IP address; a route is now an IP address and mask, allowing us to break a network into subnets and supernets. This also drastically reduces the size of Internet routing tables.

It is important to understand IP address masking and subnetting for performing a security analysis, penetration hacking, and spoofing. There’s more information on these topics later in this chapter.

 

Subnetting, VLSM, and Unraveling IP the Easy Way

Subnetting is the process of dividing an assigned or derived address class into smaller, individual, but related, physical networks. Variable-length subnet masking (VLSM) is the broadcasting of subnet information through routing protocols (covered in the next chapter). A subnet mask is a 32-bit number that determines the network split of IP addresses on the bit level.

Figure 1.8 Real-world IP network example.

Example 1 Let’s take a look at a real-world scenario of allocating IP addresses for a routed network (Figure 1.8).

Given: 206.0.125.0 (NIC assigned Class C). In this scenario, we need to divide our Class C address block to accommodate three usable subnets (for offices A, B, and C) and two subnets for future growth. Each subnet or network must have at least 25 available node addresses. This process can be divided into five steps.

Step 1 Four host addresses will be required for each of the office’s router interfaces: Router 1 Ethernet 0, Router 2 Ethernet 0/Ethernet 1, and Router 3 Token Ring 0 (see Figure 1.9).

Step 2 Only one option will support our scenario of five subnets with at least 25 IP addresses per network (as shown in the Class C subnet chart in Figure 1.10).

 

Figure 1.9 Real-world network example interface requirement chart.

 

 

 

 

 

Figure 1.10 Class C subnet chart by number of subnets versus number of hosts per subnet.

Bits in Subnet Mask: Keeping in mind the information given earlier, let’s further explore the subnet mask bit breakdown. When a bit is used, we indicate this with a 1:

 

 

 

Number of Subnets: Remember, in this scenario we need to divide our Class C address block to accommodate three usable subnets (for offices A, B, and C) and two subnets for future growth with at least 25 available node addresses per each of the five networks.

To make this process as simple as possible, let’s start with the smaller number—that is, 5 for the required subnets or networks, as opposed to 25 for the available nodes needed per network. To solve for the required subnets in Figure 1.9), we’ll start with the following equation, where we’ll solve for n in 2n – 2, being sure to cover the required five subnets or networks.

Let’s start with the power of 2 and work our way up:

22 – 2 = 2 23 – 2 = 6 24 – 2 = 14

The (3rd power) in the equation indicates the number of bits in the subnet mask. Here we see that 23 – 2 = 6 subnets if we use these 3 bits. This will cover the required five subnets with an additional subnet (or network) left over.

Number of Hosts per Subnet: Now let’s determine the number of bits left over for available host addresses. In this scenario, we will be using 3 bits in the mask for subnetting. How many are left over?

Out of the given 32 bits that make up IP addresses, the default availability (for networks versus hosts), as previously explained, for Classes A, B, and C blocks are as follows:

Class A: 8 bits

Class B: 16 bits

Class C: 24 bits

Our scenario involves a Class C block assigned by InterNIC. If we subtract our default bit availability for Class C of 24 bits (as shown) from the standard 32 bits that make up IP addresses, we have 8 bits remaining for networks versus hosts for Class C blocks.

Next, we subtract our 3 bits used for subnetting from the total 8 bits remaining for network versus hosts, which gives us 5 bits left for actual host addressing:

Let’s solve an equation to see if 5 bits are enough to cover the required available node addresses of at least 25 per subnet or network:

25 – 2 = 30

Placing the remaining 5 bits back into our equation gives us the available node addresses per subnet or network, 25 – 2 = 30 host addresses per six subnets or networks (remember, we have an additional subnet left over).

From these steps, we can divide our Class C block using 3 bits to give us six subnets with 30 host addresses each.

Step 3

Now that we have determined the subnet mask, in this case 255.255.255.224 (3 bits), we need to calculate the actual network numbers or range of IP addresses in each network. An easy way to accomplish this is by setting the host bits to 0. Remember, we have 5 bits left for hosts:

With the 5 host bits set to 0, we set the first 3 bits to 1 in every variation, then calculate the value (for a shortcut, take the first subnet value=32 and add it in succession to reveal all six subnets):

Now let’s take a look at the network numbers of our subnetted Class C block with mask 255.255.255.224:

Step 4

Now that we have solved the network numbers, let’s resolve each network’s broadcast address by setting host bits to all 1s. The broadcast address is defined as the system that copies and delivers a single packet to all addresses on the network. All hosts attached to a network can be notified by sending a packet to a common address known as the broadcast address:

Let’s take a look at the network broadcast addresses of our subnetted Class C block with mask 255.255.255.224:

Step 5

So what are the available IP addresses for each of our six networks anyway? They are the addresses between the network and broadcast addresses for each subnet or network (see Figure 1.11).

Figure 1.11

Available IP addresses for our networks.

 

 

 

 

 

 

 

Unraveling IP with Shortcuts

Let’s take a brief look at a shortcut for determining a network address, given an IP address. Given: 206.0.139.81 255.255.255.224. To calculate the network address for this host, let’s map out the host octet (.81) and the subnet-masked octet (.224) by starting from the left, or largest, number:

Now we can perform a mathematic “logical AND” to obtain the network address of this host (the value 64 is the only common bit):

We simply put the 1s together horizontally, and record the common value (205.0.125.64).

 

ARP/RARP Engineering: Introduction to Physical Hardware Address Mapping

Now that we have unearthed IP addresses and their 32-bit addresses, packet/datagram flow and subnetting, we need to discover how a host station or infrastructure equipment, such as a router, match an IP address to a physical hardware address. This section explains the mapping process that makes communication possible. Every interface, or network interface card (NIC), in a station, server, or infrastructure equipment has a unique physical address that is programmed by and bound internally by the manufacturer. One goal of infrastructure software is to communicate using an assigned IP or Internet address, while hiding the unique physical address of the hardware. Underneath all of this is the address mapping of the assigned address to the actual physical hardware address. To map these addresses, programmers use the Address Resolution Protocol (ARP). Basically, ARP is a packet that is broadcasted to all hosts attached to a physical network. This packet contains the IP address of the node or station with which the sender wants to communicate. Other hosts on the network ignore this packet after storing a copy of the sender’s IP/hardware address mapping. The target host, however, will reply with its hardware address, which will be returned to the sender, to be stored in its ARP response cache. In this way, communication between these two nodes can ensue (see Figure 1.12).

The hardware address is usually hidden by software, and therefore can be defined as the ultimate signature or calling card for an interface.

 

Figure 1.12 ARP resolution.

ARP Encapsulation and Header Formatting

It is important to know that ARP is not an Internet protocol; moreover, ARP does not leave the local logical network, and therefore does not need to be routed. Rather, ARP must be broadcasted, whereby it communicates with every host interface on the network, traveling from machine to machine encapsulated in Ethernet packets (in the data portion).

 

Figure 1.13 An ARP/RARP packet.

illustrates the encapsulation of an ARP packet including the Reverse Address Resolution Protocol (RARP) (which is discussed in the next section). The packet components are defined in the following list

 

 

 

 

 

 

 

RARP Transactions, Encapsulation

The Reverse Address Resolution Protocol (RARP), to some degree, is the opposite of ARP. Basically, RARP allows a station to broadcast its hardware address, expecting a server daemon to respond with an available IP address for the station to use. Diskless machines use RARP to obtain IP addresses from RARP servers. It is important to know that RARP messages, like ARP, are encapsulated in Ethernet frames (see Figure 1.14, Excerpt from Figure 1.13). Likewise, RARP is broadcast from machine to machine, communicating with every host interface on the network.

RARP Service

The RARP Daemon (RARPd) is a service that responds to RARP requests. Diskless systems typically use RARP at boot time to discover their 32-bit IP address, given their 48-bit hardware Ethernet address. The booting machine sends its Ethernet address, encapsulated in a frame as a RARP request message. The server running RARPd must have the machine’s name-to-IP-address entry, or it must be available from the Domain Name Server (DNS) with its name-to-Ethernet-address. With these sources available, the RARPd server maps this Ethernet address with the corresponding IP address.

 

Transmission Control Protocol

IP has many weaknesses, one of which is unreliable packet delivery—packets may be dropped due to transmission errors, bad routes, and/or throughput degradation. The Transmission Control Protocol (TCP) helps reconcile these issues by providing reliable, stream-oriented connections. In fact,

TCP/IP is predominantly based on TCP functionality, which is based on IP, to make up the TCP/IP suite. These features describe a connection-oriented process of communication establishment. There are many components that result in TCP’s reliable service delivery. Following are some of the main points:

Streams. Data is systematized and transferred as a stream of bits, organized into 8-bit octets or bytes. As these bits are received, they are passed on in the same manner.

Buffer Flow Control. As data is passed in streams, protocol software may divide the stream to fill specific buffer sizes. TCP manages this process, and assures avoidance of a buffer overflow. During this process, fast-sending stations may be stopped periodically to keep up with slow-receiving stations.

Virtual Circuits. When one station requests communication with another, both stations inform their application programs, and agree to communicate. If the link or communications between these stations fail, both stations are made aware of the breakdown and inform their respective software applications. In this case, a coordinated retry is attempted.

Full Duplex Connectivity. Stream transfer occurs in both directions, simultaneously, to reduce overall network traffic.

Figure 1.15 TCP windowing example.

 

Sequencing and Windowing

TCP organizes and counts bytes in the data stream using a 32-bit sequence number. Every TCP packet contains a starting sequence number (first byte) and an acknowledgment number (last byte). A concept known as a sliding window is implemented to make stream transmissions more efficient. The sliding window uses bandwidth more effectively, because it will allow the transmission of multiple packets before an acknowledgment is required. Figure 1.15 is a real-world example of the TCP sliding window. In this example, a sender has bytes to send in sequence (1 to 8) to a receiving station with a window size of 4. The sending station places the first 4 bytes in a window and sends them, then waits for an acknowledgment (ACK=5). This acknowledgment specifies that the first 4 bytes were received. Then, assuming its window size is still 4 and that it is also waiting for the next byte (byte 5), the sending station moves the sliding window 4 bytes to the right, and sends bytes 5 to 8. Upon receiving these bytes, the receiving station sends an acknowledgment (ACK=9), indicating it is waiting for byte 9. And the process continues. At any point, the receiver may indicate a window size of 0, in which case the sender will not send any more bytes until the window size is greater. A typical cause for this occurring is a buffer overflow.

 

TCP Packet Format and Header Snapshots

Keeping in mind that it is important to differentiate between captured packets—whether they are TCP, UDP, ARP, and so on—take a look at the TCP packet format in Figure 1.16, whose components are defined in the following list:

Figure 1.16 A TCP packet.

 

 

 

 

 

 

 

 

 

 

 

Ports, Endpoints, Connection Establishment

TCP enables simultaneous communication between different application programs on a single machine. TCP uses port numbers to distinguish each of the receiving station’s destinations. A pair of endpoints identifies the connection between the two stations, as mentioned earlier. Colloquially, these endpoints are defined as the connection between the two stations’ applications as they communicate; they are defined by TCP as a pair of integers in this format: (host, port). The host is the station’s IP address, and port is the TCP port number on that station. An example of a station’s endpoint is: 206.0.125.81:1026 (host)(port) An example of two stations’ endpoints during communication is:

This technology is very important in TCP, as it allows simultaneous communications by assigning separate ports for each station connection. When a connection is established between two nodes during a TCP session, a three-way handshake is used. This process starts with a one-node TCP request by a SYN/ACK bit, and the second node TCP response with a SYN/ACK bit. At this point, as described previously, communication between the two nodes will proceed. When there is no more data to send, a TCP node may send a FIN bit, indicating a close control signal. At this intersection, both nodes will close simultaneously.

Figure 1.17a Extracted from an HTTP Internet Web server transmission.

 

 

 

 

 

 

 

Figure 1.17b Extracted from a sliding window sequence transmission.

 

User Datagram Protocol

The User Datagram Protocol (UDP) operates in a connectionless fashion; that is, it provides the same unreliable, datagram delivery service as IP. Unlike TCP, UDP does not send SYN/ACK bits to assure delivery and reliability of transmissions. Moreover, UDP does not include flow control or error recovery functionality. Consequently, UDP messages can be lost, duplicated, or arrive in the wrong order. And because UDP contains smaller headers, it expends less network throughput than TCP and so can arrive faster than the receiving station can process them. UDP is typically utilized where higher-layer protocols provide necessary error recovery and flow control. Popular server daemons that employ UDP include Network File System (NFS), Simple Network Management Protocol (SNMP), Trivial File Transfer Protocol (TFTP), and Domain Name System (DNS), to name a few.

 

UDP Formatting, Encapsulation, and Header Snapshots

UDP messages are called user datagrams. These datagrams are encapsulated in IP, including the UDP header and data, as it travels across the Internet. Basically, UDP adds a header to the data that a user sends, and passes it along to IP. The IP layer then adds a header to what it receives from UDP. Finally, the network interface layer inserts the datagram in a frame before sending it from one machine to another. As just mentioned, UDP messages contain smaller headers and consume fewer overheads than TCP. The UDP datagram format is shown in Figure 1.18, and its components are defined in the following list.

 

Multiplexing, Demultiplexing, and Port Connections

UDP provides multiplexing (the method for multiple signals to be transmitted concurrently into an input stream, across a single physical channel) and demultiplexing (the actual separation of the streams that have been multiplexed into a common stream back into multiple output streams) between protocol and application software. Multiplexing and demultiplexing, as they pertain to UDP, transpire through ports. Each station application must negotiate a port number before sending a UDP datagram. When UDP is on the receiving side of a datagram, it checks the header (destination port field) to determine whether it matches one of station’s ports currently in use. If the port is in use by a listening application, the transmission proceeds; if the port is not in use, an ICMP error message is generated, and the datagram is discarded.

Figure 1.19 Extracted after the IP portion of a domain name resolution from a DNS request transmission

 

Internet Control Message Protocol

The Internet Control Message Protocol (ICMP) delivers message packets, reporting errors and other pertinent information to the sending station or source. Hosts and infrastructure equipment use this mechanism to communicate control and error information, as they pertain to IP packet processing.

 

 

ICMP Format, Encapsulation, and Delivery

ICMP message encapsulation is a two-fold process. The messages are encapsulated in IP datagrams, which are encapsulated in frames, as they travel across the Internet. Basically, ICMP uses the same unreliable means of communications as a datagram. This means that ICMP error messages may be lost or duplicated. The ICMP format includes a message type field, indicating the type of message; a code field that includes detailed information about the type; and a checksum field, which provides the same functionality as IP’s checksum (see Figure 1.20). When an ICMP message reports an error, it includes the header and data of the datagram that caused the specified problem. This helps the receiving station to understand which application and protocol sent the datagram. (The next section has more information on ICMP message types.)

Figure 1.20 Illustration of an ICMP datagram.

Figure 1.21 ICMP message chart.

 

ICMP Messages, Subnet Mask Retrieval

There are many types of useful ICMP messages; Figure 1.21 contains a list of several, which are described in the following list. Echo Reply (Type 0)/Echo Request (Type 8). The basic mechanism for testing possible communication between two nodes. The receiving station, if available, is asked to reply to the ping. An example of a ping is as follows:

STEP 1:

BEGIN ECHO REQUEST Ping 206.0.125.81 (at the command prompt)

Reply from 206.0.125.81: bytes-32 time<10ms TTL=128 (from receiving station 206.0.125.81)

Reply from 206.0.125.81: bytes-32 time<10ms TTL=128

Reply from 206.0.125.81: bytes-32 time<10ms TTL=128

Reply from 206.0.125.81: bytes-32 time<10ms TTL=128

Destination Unreachable (Type 3). There are several issuances for this message type, including when a router or gateway does not know how to reach the destination, when a protocol or application is not active, when a datagram specifies an unstable route, or when a router must fragment the size of a datagram and cannot because the Don’t Fragment Flag is set. An example of a Type 3 message is as follows:

STEP 1: BEGIN ECHO REQUEST Ping 206.0.125.81 (at the command prompt)

STEP 2: BEGIN ECHO REPLY Pinging 206.0.125.81 with 32 bytes of data:

Destination host unreachable.

 Source Quench (Type 4). A basic form of flow control for datagram delivery. When datagrams arrive too quickly at a receiving station to process, the datagrams are discarded. During this process, for every datagram that has been dropped, an ICMP Type 4 message is passed along to the sending station. The Source Quench messages actually become requests, to slow down the rate at which datagrams are sent. On the flip side, Source Quench messages do not have a reverse effect, whereas the sending station will increase the rate of transmission.

Route Redirect (Type 5). Routing information is exchanged periodically to accommodate network changes and to keep routing tables up to date. When a router identifies a host that is using a nonoptional route, the router sends an ICMP Type 5 message while forwarding the datagram to the destination network. As a result, routers can send Type 5 messages only to hosts directly connected to their networks.

Datagram Time Exceeded (Type 11). A gateway or router will emit a Type 11 message if it is forced to drop a datagram because the TTL (Time-to-Live) field is set to 0. Basically, if the router detects the TTL=0 when intercepting a datagram, it is forced to discard that datagram and send an ICMP message Type 11.

Datagram Parameter Problem (Type 12). Specifies a problem with the datagram header that is impeding further processing. The datagram will be discarded, and a Type 12 message will be transmitted.

Timestamp Request (Type 13)/Timestamp Reply (Type 14). These provide a means for delay tabulation of the network. The sending station injects a send timestamp (the time the message was sent) and the receiving station will append a receive timestamp to compute an estimated delay time and assist in their internal clock synchronization.

Figure 1.22 ICMP header sniffer capture.

Information Request (Type 15)/Information Reply (Type 16). As an alternative to RARP (described previously), stations use Type 15 and Type 16 to obtain an Internet address for a network to which they are attached. The sending station will emit the message, with the network portion of the Internet address, and wait for a response, with the host portion (its IP address) filled in.

Address Mask Request (Type 17)/Address Mask Reply (Type 18). Similar to an Information Request/Reply, stations can send Type 17 and Type 18 messages to obtain the subnet mask of the network to which they are attached. Stations may submit this request to a known node, such as a gateway or router, or broadcast the request to the network.

ICMP Header Snapshots

Figure 1.22 on page 35 contains snapshots of an ICMP Header. The first was extracted after the IP portion of an ICMP ping test transmission; the second was extracted during an unreachable ping test.

 

 

 

 

 

 

 

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