EXAM #2 FOR INTRODUCTION TO COMPUTER NETWORKS

2002
NAME: ____________________________ SSN: ___________________
CODENAME: ________________________
Welcome to Exam #2 in Introduction to Networks. Read each problem carefully.
There are seven required problems (each worth 14 points - 2 points for writing
your name on the exam) and one extra credit problem worth 10 points. You may
have with you a calculator, pencils, blank paper, lucky rabbit's foot, and one
8.5 x 11 inch "formula sheet". On this formula sheet you may have anything you
want (definitions, formulas, old exam answers, etc.) as hand written by you on
both sides of the sheet. Photocopies, scans, or computer generated text are not
allowed on this sheet. Good luck and be sure to show your work!
PROBLEM #1
Answer the following questions about bridges:
a) Precisely define what is a bridge (in the context of networks, obviously).
--A bridge is a MAC layer filter. It filters frames by MAC addresses. Frames
on "this side" of a bridge are filtered-out (i.e., not forwarded).
--b) What is the difference between a bridge and a repeater. Describe what a
repeater does and what is its purpose. Does a bridge achieve this same
purpose?
--A bridge is layer 2 and a repeater is layer 1. A repeater passes through
all frames and regenerates the digital signal. When the signal is
regenerated it is brought back to the "full" voltage level of the signal
(e.g. as sent at the sending host). A bridge also regenerates the digital
signal so it achieves the same purpose in this respect.
--c) Ethernet uses a type of bridging called "transparent bridge". This
type of bridging also goes by two other names. What are these names?
Explain where all three names come from (i.e., what they mean).
---

Fall

The term TB came into being because this bridge type is totally "transparent"
to end stations. No changes to end stations are needed when a TB bridge is
installed. A TB bridge is also called a "learning bridge" because it
automatically learns its bridge tables. Another name is "spanning tree bridge"
because the configuration of the network must be a spanning tree for this
type of bridge to work.
--d) Give the transparent bridge forwarding and learning algorithm.
--Receive a frame
If (DA is in table) and (DA is local) then not forward else forward
If (SA in table) check/update direction
if (SA not in table) add to table
--PROBLEM #2
Answer the following questions about bridges:
a) Briefly describe source routing.
--In source routing a Routing Information Field (RIF) is included in a frame as
part of the LAN header. The RIF contains hop ids for the frame to follow.
Each bridge that receive a frame forwards it only if that bridges hop id is
in the RIF.
--b) What are advantages of source route bridging? What are advantages of
transparent bridging?
--An advantage of source routing is that it simplifies bridge design. Bridging
tables and complicated CAMs are not needed. An advantage of transparent
bridging is that it requires no changes to the end stations (but, source
routing does require that end stations do discovery and otherwise insert RIF
fields in all frames).
--PROBLEM #3
Answer the following questions about sockets programming:

a) What is the history (the "roots") of the sockets interface?

--Sockets developed with Unix. BSD flavor Unix to be exact. This explains the
file I/O paradigm of sockets.
--b) Sketch the flowchart for a sockets client/server model. For each step of
the flowchart give the appropriate sockets function name.
--SERVER

CLIENT

socket()
socket()
|
|
V
|
bind()
|
|
|
V
V
listen() <---------- connect()
|
|
V
|
accept()
|
|
|
V
V
send()/recv() <--> send()/recv()
|
|
V
V
close()
close()
--c) When we say that some sockets functions "block", what do we mean? Give
an example of a sockets function that blocks.
--This means that the function waits on an external event (e.g., a connection
establishment) and does not return until this event has occurred. This could
take many seconds in some cases. For example, the accept() function will not
return until a connection has been established. Other blocking functions
include connect(), recv(), and send().
--PROBLEM #4

Consider the following network consisting of two three-station Ethernets

attached via an IP router. Hosts A through F have MAC addresses of aaa,
bbb, ccc, ddd, eee, and fff, respectively. Router port 1 has MAC address
xxx, port 2 has MAC address yyy. The host id is shown above each host, the
netid below the Ethernet.
Router
+----------+
|
|
001 002 003
|1
2|
004 005 006
A B C
+-+------+-+
D E F
| | |
|
|
| | |
--+-----+-----+---------+- -+---------+-----+-----+--001.128.123
001.128.124
a) Give the ARP cache for station C assuming that C has recently
communicated with all other stations.
--IP
MAC
-------------------------001.128.123.001 aaa
001.128.123.002 bbb
001.128.123.004 xxx
001.128.123.005 xxx
001.128.123.006 xxx
--b) Give the routing table for the router.
--net_id
port
---------------------001.128.123 1
001.128.124 2
--c) If A sends D a packet, explain what takes place and sketch the
packet. In your packet sketch carefully identify the contents of
the address fields.
--<-- MAC --> <--------------- IP -------------->
+-----+-----+-----------------+-----------------+--| xxx | aaa | 001.128.123.001 | 001.128.124.004 | ... on LAN #1

+-----+-----+-----------------+-----------------+--<-- MAC --> <--------------- IP -------------->

+-----+-----+-----------------+-----------------+--| ddd | yyy | 001.128.123.001 | 001.128.124.004 | ... on LAN #2
+-----+-----+-----------------+-----------------+----PROBLEM #5
Answer the following questions about IP and TCP:
a) How many bytes in the IP and TCP headers? List the key fields for the
IP and TCP headers.
--Each header is 20 bytes (without options). Key fields for the IP header
include source and destination IP address. For the TCP header key fields
are the source and destination port number, sequence and ack #, window
size, and segment checksum.
--b) Sketch the packet flows for a TCP connection establishment.
--SYN ->
<- SYN/ACK
ACK ->
--c) What is the key difference between TCP and UDP?
--TCP is assured delivery and UDP is not (UDP is datagram service).
--d) Why are IP addresses hierarchal with netid and hostid?
--IP address addresses are hierarchical to reduce the size of routing tables.
IP packets are routed only by netid until they reach their destination network
where ARP is then used to resolve hostid to MAC address.
---

PROBLEM #6
Answer the following questions about TCP
a) What is the consequence of having a time-out value (for segment
retransmission) that is too short (and define what is "too short")?
--A too short time-out value is one which is shorter than the round trip
time of a send packet and its returning ACK. A too short value will
result in a double send of the original packet.
--b) What is the consequence of having a time-out value that is too long?
--A too long time-out value will result in a sending host sitting idle
where is should be re-sending in the case of a lost packet of ACK.
--c) If RTT_i (the i_th RTT estimate) is not updated on time-outs what can
happen? Be precise (describing an example might be easiest and best).
--It the RTT estimate is not updated when time-outs occur the sender can
get stuck in a state with a too small RTT estimate. For example, if the
current path fails and a new loner path is used for subsequent packets then
the new path will have a longer RTT. This longer RTT will cause the sender
to always time-out and also never thus be able to update its RTT estimate.
Thus, the sender is stuck in "double send" mode for the duration of the
connection.
--PROBLEM #7
Answer the following questions about HTTP and web server implementations
a) What is HTTP, where does it "fit" in the stack, and at a very high
level how does it operate.
--HTTP is the HyperText Transfer Protocol. It fits above TCP. It is
a request/response protocol between a client and a server.
---

b) Describe the contents of a GET header.

--A GET header contains the GET command and file name requested (something
like GET \index.html HTTP\1.1). Optional fields following this command
include accept, accept encoding, accept language, host, and others.
--c) Give the flowchart (or steps) for the operation of a simple web server
(i.e., a web server that serves only static objects). Identify where
parallelism (e.g., as implemented with threads) is typically implemented.
--Wait for a connection
Make the connection
Receive a command and parse it
Respond with requested stuff (i.e., do the command)
Drop the connection
Typically the last three steps are "spun off" in a thread leaving the main
program to wait and make additional connections (and spin off more threads).
--EXTRA CREDIT PROBLEM:
Telnet is an application that allows a client to have a console for a remote
server run on his or her PC. As the user types on their keyboard, key
characters can be sent one at a time to the server. TCP/IP is used for
communications. What is the approximate overhead in a telnet session
given that each keystroke or character is sent individually to the server
(or from the server to the client). Show your calculations. Do you think
telnet really works this way? Can you suggest an improvement? What is the
approximate reduction in overhead from your improvement? Again, show your
calculations. The approach you use to this problem is much more important
that any "numbers" you come-up with.
--For each one byte carried there will be 16 bytes of LAN header and trailer,
20 bytes of IP header, and 20 bytes of TCP header. This is an efficiency
of 1 / (1 + 16 + 20 + 20) = 1.75% (or overhead of over 98%). If each character
typed is ACKed, then the efficiency is approximately cut in half... or less
than 1% efficiency (greater than 99% overhead). Telnet actually sends an entire
typed line (a command ended by CRLF) and not each character. Assuming 20
bytes per line and ACKs, we get an efficiency of about

20 / (20 + 16 + 20 + 20 + 16 + 20 + 20) = 15.2%.

This is still very low efficiency and high overhead (about 85% overhead).
---

EXAM #1 FOR INTRODUCTION TO COMPUTER NETWORKS

2002
NAME: ____________________________ SSN: ___________________
CODENAME: ________________________
Welcome to Exam #1 in Introduction to Networks. Read each problem carefully.
There are eight required problems (each worth 12.5 points). You may have with
you a calculator, pencils, blank paper, lucky rabbit's foot, and one 8.5 x 11
inch "formula sheet". On this formula sheet you may have anything you want
(definitions, formulas, old exam answers, etc.) as hand written by you on both
sides of the sheet. Photocopies, scans, or computer generated text are not
allowed on this sheet. Good luck and be sure to show your work!
HINTS: For most (but, not all) of the short answer questions one or two
sentences (not paragraphs!) are all that is needed. For the numerical
problems be sure to show your work and give all formulas used.
PROBLEM #1
Answer the following questions about layered protocol models
a) Sketch the OSI model. For each layer, in one sentence or less, state the
purpose of the layer.
--+--------------+
| Application | - Provides access to user applications
+--------------+
| Presentation | - Provide data independence (translation)
+--------------+
| Session | - Manages end-to-end connections

Fall

+--------------+
| Transport | - Provides reliable end-to-end data transport
+--------------+
| Network | - Maintains point-to-point connections
+--------------+
| Data Link | - Provides reliable point-to-point data transport
+--------------+
| Physical | - Transmission of bit stream (cables, connectors, signal)
+--------------+
--b) Following are 10 functions. Identify which layer of the OSI model each
function is most likely to belong to. It is possible that one or more of the
below cannot be mapped to an OSI layer. If this is the case, explain why.
(1) Recovering lost packets between two directly connected nodes
(2) Recovering lost packets between two nodes separated by multiple hops
(3) Translating between EBCDIC (an old IBM coding of letters) to ASCII
(4) Defining the pin-outs in a connector used to attach to a network cable
(5) Providing an interface to a visual packet monitoring program
(6) Arbitrating between multiple nodes attached to a single medium
(7) Translating between text in messages from Dutch to English
(8) Generating error correction codes for packet error correction
(9) Maintaining connection semantics beteen two directly connected nodes
(10) Finding the shortest path between two nodes separated by multiple hops
--(1) = DLC, (2) = Transport, (3) = Presentation, (4) = Physical,
(5) = Application, (6) = DLC, (7) = Presentation, (8) = NONE (OSI does ARQ),
(9) = Network, and (10) = Transport or Session
--c) Assume a system with fixed-length 250-byte packets where each packet contains
25 bytes of headers and trailers (thus, the data payload is 225 bytes). For
a 600 byte message, what is the overhead (in %) in this system? Hint:
Packets with less payload than 225 bytes are "padded out" to a full length
payload.
--Will need three packets, thus U = 600 / (3 * 250) = 80%.
--d) Describe two fundamentally different ways that standards can be developed.
Give examples of standards groups that use these two different ways.
---

Write the standard, then build it (OSI, ITI, IEEE). Build it, then
standardize is (IETF).
--PROBLEM #2
Answer the following physical layer related questions:
a) Name three types of media used in networks. Give the approximate data
rate achievable on each type of media. Cotton string is not an acceptable
answer for a media type :-).
--Coax (to many hundreds of Mbps), unshielded twisted pair (10's of Mbps
for catagory 3, 150-Mbps for catagory 5), and fiber (terabits per second).
--b) What are some of the parameters of interest for comparing different
types of media? Give at least six parameters.
--Seven parameters are cost per foot, bulk, stength, noise immunity,
how much it radiates, lightening immunity, and how tapable.
--c) What is an advantage of Manchester coding over NRZ coding? What is a
disadvantage (of Manchester over NRZ)?
--It is possible to derive the transmitter clock from a received Manchester
signal thus bit sampling will be correct. Without a synchronized clock,
bit sampling in NRZ signal may be off. However, Manchester requires
twice the bandwidth as NRZ (two "wiggles" per bit and not just one).
--d) What is an RS-232 null modem? How is it built?
--An RS-232 null modem is used to connect two DTE's together. A null modem
is two RS-232 connector wired together. It is built by "criss-crossing"
the transmit and receive wires and also clear-to-send and request-to-send.
--PROBLEM #3

Consider an asynchronous transmission system with 1 start bit, 3 data bits,

1 parity bit, and 2 stop bits. Assume Pr[bit error] = 0.05 and bit errors
are independent. Show your work.
a) What is the maximum utilization possible on this link (assuming no errors)?
--U = 3 / (3 + 1 + 1 + 2) = 43%
--b) What is the probability that a received word has no errors?
--(1 - 0.05)^4 = 0.8145
--c) What is the probability that a received word is detected as "good"?
--Sum(i=0, 2, 4) Cmb(4, i) p^i * (1-p)^4-i = 0.828
--d) What is the probability that 10 words in a row are received with no errors?
--((1 - 0.05)^4)^10 = 0.1285
--PROBLEM #4
Consider the use of 1000-bit frames on a 1-Mbps satellite channel with a
270 millisecond delay. What is the maximum link utilization for continuous
flow control (GBN) with a window size of 255 and with Pr[bit error] = 1.0e-6.
--t_fr = 1000 / 1.0e6 = 1 millisec
t_pr = 270 millisec
For N = 255 we have that N*t_fr < 2*t_pr + t_fr, so
p = Pr[frame error] = 1 - (1 - 1.0e-6)^1000 = 0.001
U = (N * t_fr * (1 - p)) / ((1 - p + N*p)(2*t_pr + t_fr)) = 37.7%

--PROBLEM #5
Carefully describe the trade-offs between Go-Back-N (GBN) and Selective
Reject (SR) Automatic Repeat Request (ARQ). Your discussion should make
it clear when GBN is best suited and when Selective Reject is best
suited.
--GBN will likely result in already receive packets being resent. This
is wasteful of network resources. SR will resend only those packets
actually detected as lost. GBN is, however, easier to implement in
both the sender and receiver than is SR. So, the trade-off is in
network resources versus complexity of implementation in the end
nodes (sender and receiver). GBN is best suited when errors are rare
and network resources (bandwidth) is not scarce. SR is best suited
when errors are frequent and/or network resources are scarce.
--PROBLEM #6
Answer the following questions related to circuit switching
a) Sketch a T1 frame and explain why the T1 rate is 1.544-Mbps (show your
calculations).
--++----+----+----//---+----+
|| 1 | 2 |
| 24 | <--- 1 bit SDEL followed be 24 slots of
++----+----+----//---+----+
8 bits each for 193 bits total
Each slot must run at 64-kbps (for PCM voice), so 8000 frames per second,
or 193 bits/frame * 8000 frames/sec = 1.544 Mbps
--b) Outline how the Erlang-B formula could be used to determine how many
phone lines would be needed to provision an office of 100 workers.
Clearly, each worker will have a telephone, but how many phone lines
should be leased between the office and the phone company switching
center? Hint: It is not sufficient to simply state "Use the Erlang
formula to determine how many phone lines are needed"... you must be
much more detailed! Carefully state any and all assumptions that
must be made. Consider only incoming calls to the office.

--We need to know the rate of arrival of phone calls and the average
duration of a call. We could measure these or use past experience.
We will assume that arriving calls are independent and Poisson. We
should establish a Pr[block] that is acceptable to us (this is a
business decision). Say, 99.999% of all calls should be connected
(and then Pr[block] = 0.001%). Then we solve the Pr[block] for
decreasing n (starting with n = 100) until we arrive at an
unacceptable Pr[block]. This is then the number of phone lines we
should lease.
--PROBLEM #7
Answer the following questions related to packet switching (PS):
a) Describe datagram packet switching.
--Packet switching where each packet is invidually routed at each
intermediate node.
--b) Describe virtual circuit packet switching
--Packet switching where an path is established and all packet in
a connection travel on this same path.
--c) What is one significant advantage of datagram PS compared to Virtual
Circuit PS?
--Datagram PS can tolerate network failures (e.g., down links or
intermediate nodes) better than can VC PS.
--d) What is one significant advantage of virtual circuit compared to
Datagram PS?
--VC PS can deny a connection if the network resources are not there to
adequately support additional traffic.
---

PROBLEM #8
Carefully and precisely describe the IEEE 802.3 MAC protocol (CSMA/CD plus BEB)
for shared-medium LANs. Describe all relevant algorithms, parameters (such as
minimum and maximum frame sizes), and constraints.
--CSMA/CD is Carrier Sense Multiple Access / Collision Detection and enables the
sharing of a copper medium among multiple hosts. A 1-persistent CSMA
algorithm is implemented as follows (for when a host has a frame to transmit):
1) If medium is idle then transmit
2) If medium is busy then wait for medium to go idle and then transmit
3) If detect a collision then xmit a brief jam signal, stop transmitting,
wait a random period of time, then (1)
The "wait a random period of time" is implemented with Binary Exponential
Backoff (BEB) as:
while (attempts < 16)
k = min(attempts, 10)
r = rand(0, 2^k)
delay = r * slot_time
A slot_time is the time it takes to transmit 64 bytes. This time determines
the maximum span of an Ethernet (i.e., tpr <= tfr for frame of 64 bytes and
tpr being the end-to-end propagation time). For 100-Mbps this results in a
maximum span of slightly over 200 meters. The maximum frame size in Ethernet
is 1500 bytes (and the minimum is 64 bytes).
-----

FINAL EXAM FOR INTRODUCTION TO COMPUTER NETWORKS

Fall 2002
>>> SOLUTIONS <<<
Welcome to the Final Exam in Introduction to Computer Networks. Read each
problem carefully. There are eleven required problems (each worth 9 points and
you get 1 point for putting your name on the cover sheet) and one extra credit

problem worth 10 points. You may have with you a calculator, pencils, blank
paper, lucky rabbit's foot, and one 8.5 x 11 inch "formula sheet". On this
formula sheet you may have anything you want (definitions, formulas, old exam
answers, etc.) as hand written by you on both sides of the sheet. Photocopies,
scans, or computer generated text are not allowed on this sheet. It is strongly
recommended that you answer each problem on a separate sheet of paper. Good
luck and be sure to show your work!
PROBLEM #1
Answer the following questions about layered protocol models.
a) Describe the layered concept. Be precise and concise.
--Layers perform unique and specific tasks and have knowledge of adjacent
layers only. Use services from below and provide services above. Independent
of implementation.
--b) Precisely define "protocol" and "interface". Give an example of each (a
protocol and an interface).
--Protocol = complete set of rules for information interchange between same level
layers at different sites. Example = TCP/IP.
Interface = same, except for between consecutive layers at the same site.
Example = sockets interface.
--c) Consider a three layer protocol. The layers from top to bottom are
application, transport, and network. All layers use headers and the
transport layer also uses a trailer. Sketch a packet showing headers and
trailers for this protocol.
--+-------+-------+-------+---------------------------+-------+
| Net | Trans | Appl | Payload or user data | Trans |
+-------+-------+-------+---------------------------+-------+
--d) Assume a protocol that uses variable length packets of maximum payload size
250 bytes. In addition to the payload there are 25 bytes of header and 12
bytes of trailer in each packet. For a 600 byte message, what is the

overhead (in %) for this protocol?

--600 / 250 = 2 r 100 so three packets are needed. The total overhead is three
sets of headers and trailers for 600 bytes of user data or 111 / 711 = 15.6%
overhead.
--PROBLEM #2
Answer the following questions about the physical layer.
a) What are some of the key parameters of interest for guided media. List at
least five parameters.
--Noise immunity, low radiation, physical strength, physical size, and cost.
--b) What is the difference between UTP category 3 and UTP category 5? Be precise
(it is not sufficient to say that one is "faster" than the other).
--UTP-5 has more twists per inch than UTP-3 resulting in less radiation. Hence,
two pairs of UTP-5 can carry up to about 100-Mbps data rate where UTP-3 can
carry a maximum data rate of about 16-Mbps.
--c) If a received signal is 50% the power level of a transmitted signal, what is
the loss in dB?
--Ndb = 10*log10(P2/P1) = 10*log10(0.50) = -3db
--d) If a digital signal is sent as +5V for a "1" and 0V for a "0" can the
receiver recover the sender's clock rate? Explain why or why not.
--A long sequence of 1 (or 0) bits would not have any transitions thus making
it impossible for the receiver to recover a clock. Transitions are needed, at
the rate (or some known multiple) of the sending clock, to be able to recover a
signal.
---

PROBLEM #3
Assume a 100-Mbps link of 10,000-meters in length with 5 nanoseconds per meter
propagation delay. Assume constant length 400-byte data frames, 64-byte ACK
frames, 10 microsecond of processing delay for each data frame, and 5
microseconds of processing time for each ACK. The sender always has data to
send. Solve for link utilization (U) between a sender and a receiver assuming a
stop-and-wait protocol. Show your work!
--t_pr = 10,000 * 5e-9 = 50 us
t_fr = 8 * 400 / 100e6 = 32 us
t_ack = 8 * 64 / 100e6 = 5.12 us
We note that the processing delays are not negligible compared to these values
so we must include them in our calculation of U...
U = t_fr / (t_pr + t_fr + t_proc + t_ack + t_proc) = 21%
If we neglected t_ack and t_proc our result would have been U = 24% which is
more than 10% "off" from the real result confirming that to ignore these values
would not have been correct to do.
--PROBLEM #4
Answer the following questions about circuit and packet switching.
a) Describe the fundamental difference between circuit and packet switching.
--Circuit switching assigns a dedicated channel for two end-points. Packet
switching shares channels.
--b) What are the two basic types of circuit switches? Give an example of each
--Space switch (e.g., crossbar) and time switch (e.g. Time-Slot Interchange).
--c) What is virtual circuit packet switching? What is datagram packet switching?
What are the key advantages of each?
---

In VC PS a path is established as part of a connection. In DG PS packets are

individually routed. Advantage of VC PS is the ability to reject connections
if network load is high. Advantage of DG PS is robustness in face of
intermediate node failures.
--d) What is the Erlang formula used for? Be specific.
--To determine the blocking probability (i.e., that no servers are available to
service a request) given N servers for a given rate of arrival of requests.
--PROBLEM #5
Carefully and precisely describe the IEEE 802.3 MAC protocol (CSMA/CD plus BEB)
for shared-medium LANs. Describe all relevant algorithms, parameters (such as
minimum and maximum frame sizes), and constraints.
--CSMA/CD is Carrier Sense Multiple Access / Collision Detection and enables the
sharing of a copper medium among multiple hosts. A 1-persistent CSMA
algorithm is implemented as follows (for when a host has a frame to transmit):
1) If medium is idle then transmit
2) If medium is busy then wait for medium to go idle and then transmit
3) If detect a collision then xmit a brief jam signal, stop transmitting,
wait a random period of time, then (1)
The "wait a random period of time" is implemented with Binary Exponential
Backoff (BEB) as:
while (attempts < 16)
k = min(attempts, 10)
r = rand(0, 2^k)
delay = r * slot_time
A slot_time is the time it takes to transmit 64 bytes. This time determines
the maximum span of an Ethernet (i.e., tpr <= tfr for frame of 64 bytes and
tpr being the end-to-end propagation time). For 100-Mbps this results in a
maximum span of slightly over 200 meters. The maximum frame size in Ethernet
is 1500 bytes (and the minimum is 64 bytes).
---

PROBLEM #6
Give the bridge tables for the two bridges in the network below. Assume
that all stations have communicated with all other stations. The letter
(e.g., "A") signifies a MAC address of "A" for a station at this position.
Note that one of the bridges is a three-port bridge.
|
A --+
|
|
+--------------------------------+
B --+ |
|
|
|
|
|
+-- Bridge 1 --+
|
|
|
|
+-- C
|
|
|
+-- E
+-- Bridge 2 --+
|
|
|
|
+-- D +-- F
|
|
--Define the port numbers on bridge 1 as 0, 1, 2 in clockwise order staring
from port 0 on the left. The ports on bridge 2 are 0 and 1 (0 is left).
Bridge 1
Bridge 2
--------------------MAC Direc
MAC Direc
A 0
A 0
B 0
B 0
C 2
C 0
D 2
D 1
E 1
E 0
F 1
F 0
----------------------PROBLEM #7
Suppose that you have been assigned IP address 132.128.0.0 and have 9 networks
in your organization that you wish to remain distinct. Give the subnet mask.
Give the subnet addresses of the 9 distinct networks.
---

Will need 4 bits to hold 9 values. So, the subnet mask is:
255.255.240.0 (note that 240 = 11110000b)
Subnet #1 = 132.128.16.0
Subnet #2 = 132.128.32.0
Subnet #3 = 132.128.48.0
Subnet #4 = 132.128.64.0
Subnet #5 = 132.128.80.0
Subnet #6 = 132.128.96.0
Subnet #7 = 132.128.112.0
Subnet #8 = 132.128.128.0
Subnet #9 = 132.128.144.0
--PROBLEM #8
Below is a captured TCP/IP packet. On the last page of this exam are packet
header templates. Decode the packet for the following fields:
- MAC source and destination addresses
- IP source and destination addresses (in standard dotted-decimal format)
- TCP source and destination port numbers
- Type of TCP segment (e.g., SYN, FIN, ACK, etc.)
- TCP window size
00 60 CF 20 | 2B 8B 00 60 | CF 20 2B 7F | 08 00 45 00 [.`. +..`. +..E.]
00 30 C4 B2 | 40 00 80 06 | 2A 08 83 F7 | 02 10 83 F7 [.0..@...*.......]
02 0F 0B 0B | 00 50 50 20 | 0D 16 00 00 | 00 00 70 02 [.....PP ......p.]
40 00 CE 80 | 00 00 02 04 | 05 B4 01 01 | 04 02
[@.............]
--MAC source = 0060CF202B7Fh
MAC dest
= 0060CF202B8Bh
IP source
= 131.247.2.16
IP dest
= 131.247.2.15
TCP source port = 2827
TCP dest port = 80
TCP type
= SYN
TCP window size = 16384
--PROBLEM #9

Answer the following questions about ISDN, frame relay, and BISDN.
a) What were the original goals of ISDN? Of frame relay? Of BISDN?
--ISDN = end-to-end digital voice telephony
Frame relay = frame-based, billed by usage interconnection of LANs
BISDN = video transport (also voice and data)
--b) Sketch a graph of offered load versus network throughput (where throughput is
"good" or useful carried traffic). Show the ideal throughput behavior. Show
actual, with and without flow control, behavior. Identify the points for
congestion avoidance and congestion recovery in the no flow control curve.
--Goodput |
|
Capacity| - - - -/----------------------------------- Ideal
|
/ +++++++++++++++++++++++++++++ Control
|
/ + ........
| / +A
B
| / +.
.
A = congestion avoidance
| /+.
.
B = congestion recovery
| /+ .
. No control
| /+.
.
|/.
.
+----------------------------------------------offered load
--c) Sketch the BISDN protocol model.
--See textbook, Figure 11.1 on page 349.
--PROBLEM #10
Answer the following questions about BISDN and ATM.
a) ATM is intended to be the "perfect transport". Carefully describe the
desired characteristics of a perfect transport.

--* Suitable for LAN and WAN

* Suitable for carrying real-time traffic (data, voice, and video)
* Speed scalable
* Easy to switch
* Exploit cheap h/w and cheap/plentiful bandwidth
--b) Sketch an ATM cell and identify the header fields.
--See textbook, Figure 11.4 on page 355.
--c) Briefly describe ATM AAL1 and AAL5. What are they for and (very briefly)
how do the work.
--AAL1 supports CBR and includes timing information (e.g., for clock extraction
at the receiver.
AAL5 support data packets and includes segmentation and reassembly, and error
detection via a CRC in an added packet (PDU) trailer.
--d) Briefly describe the ATM UNI. What is it for and (very briefly) how does
it work.
--The ATM User Network Interface describes attachment to an ATM switch and
deals with connection types (permanent or switched virtual circuit) and
congestion control (e.g., ABR).
--PROBLEM #11
Answer the following questions about MPLS and VPNs
a) What was the original purpose (or goal) of MPLS? What is the present
purpose?
--The original purpose of MPLS was to overcome an IP routing table look-up
bottleneck by using ATM switching to carry IP traffic. This look-up
bottleneck has been "fixed", however MPLS is today useful for traffic

engineering (being able to precisely control paths for traffic in a network).

--b) What is the goal of a VPN?
--The goal of a VPN is to provide private communications on a public network
--c) Why not use a physically private network instead of a VPN?
--A physically private network (e.g., such as leased T1's between corporate
sites) is usually more expensive than just using the Internet.
--d) What are the four basic functions of a VPN?
--Authentication
Access Control
Data Security
Tunneling
--EXTRA CREDIT:
Having a feel for performance, or "the numbers", is very important. Answer the
following:
a) For a Gigabit Ethernet the minimum packet length is 512 bytes, for 100-Mbps
the minimum packet length is 64 bytes. Compute media (or wire) speed for
Gigabit and 100-Mbps Ethernet for minimum length packets. What conclusion
can you draw?
--1.0e9 / (8 * (512 + 12 + 8)) = 234,962 pkts/sec for Gigabit
1.0e8 / (8 * (64 + 12 + 8)) = 148,819 pkts/sec for 100-Mbps
The "8" is for preamble and the "12" is for interframe gap.
The media speed for Gigabit is less than double that of 100-Mbps for minimum
size packets. To get the full gain of Gigabit, you must use large payloads.

--b) Approximately, what is the propagation delay from here to Disney World?
--It is about 70 miles (or about 370,000 feet) from here to DW. At light
propagation of about 1 ft per nanosecond, the propagation delay is about 0.37
milliseconds (less than 1/2 millisecond).
--c) Given a bit error probability of 1e-5 (bit errors are assumed to be
independent), what is the probability of a 1000 byte packet being error
free? How about a 100 byte packet?
--Pr[no error for 1000 bytes] = (1 - 1e-5)^(8*1000) = 0.92
Pr[no error for 100 bytes] = (1 - 1e-5)^(8*100) = 0.99
---