1. Network Review
Computer Networking: A Top Down Approach , 4th edition. Jim Kurose, Keith Ross Addison-Wesley, July 2007.

2. Network Review Overview: what’s the Internet?
network edge; hosts, access net
network core: packet/circuit switching, Internet structure
performance: loss, delay, throughput
Protocol
TCP and UDP

3. What’s the Internet: “nuts and bolts” viewPC
server
wireless
laptop
cellular
handheld
millions of connected computing devices: hosts = end systems
running network apps
Home network
Institutional network
Mobile network
Global ISP
Regional ISP
communication links
fiber, copper, radio, satellite
transmission rate = bandwidth
wired
links
access
points
routers: forward packets (chunks of data)
router

4. What’s the Internet: “nuts and bolts” view
protocols control sending, receiving of msgs
e.g., TCP, IP, HTTP, Skype, Ethernet
Internet: “network of networks”
loosely hierarchical
public Internet versus private intranet
Internet standards
RFC: Request for comments
IETF: Internet Engineering Task Force
Home network
Institutional network
Mobile network
Global ISP
Regional ISP

5. What’s the Internet: a service view
communication infrastructure enables distributed applications:
Web, VoIP, , games, e-commerce, file sharing
communication services provided to apps:
reliable data delivery from source to destination
“best effort” (unreliable) data delivery

6. A closer look at network structure:
network edge: applications and hosts
access networks, physical media: wired, wireless communication links
network core:
interconnected routers
network of networks

7. The network edge: end systems (hosts): client/server model
run application programs
e.g. Web,
at “edge of network”
peer-peer
client/server
client/server model
client host requests, receives service from always-on server
e.g. Web browser/server; client/server
peer-peer model:
minimal (or no) use of dedicated servers
e.g. Skype, BitTorrenth

8. Network edge: reliable data transfer service
Goal: data transfer between end systems
handshaking: setup (prepare for) data transfer ahead of time
Hello, hello back human protocol
set up “state” in two communicating hosts
TCP - Transmission Control Protocol
Internet’s reliable data transfer service
TCP service [RFC 793]
reliable, in-order byte-stream data transfer
loss: acknowledgements and retransmissions
flow control:
sender won’t overwhelm receiver
congestion control:
senders “slow down sending rate” when network congested

9. Network edge: best effort (unreliable) data transfer service
Goal: data transfer between end systems
same as before!
UDP - User Datagram Protocol [RFC 768]:
connectionless
unreliable data transfer
no flow control
no congestion control
App’s using TCP:
HTTP (Web), FTP (file transfer), Telnet (remote login), SMTP ( )
App’s using UDP:
streaming media, teleconferencing, DNS, Internet telephony

10. Access networks and physical media
Access networks – the physical links that connect and end system to the first router (edge router) on a path from the end system to any other distant end system.
Q: How to connect end systems to edge router?
residential access nets
institutional access networks (school, company)
mobile access networks
Keep in mind:
bandwidth (bits per second) of access network?
shared or dedicated?

11. Residential access: point to point access
Dialup via modem
up to 56Kbps direct access to router (often less)
Can’t surf and phone at same time: can’t be “always on”
DSL: digital subscriber line
deployment: telephone company (typically)
up to 1 Mbps upstream (today typically < 256 kbps)
up to 8 Mbps downstream (today typically < 1 Mbps)
dedicated physical line to telephone central office

12. Residential access: cable modems
HFC: hybrid fiber coax
asymmetric: up to 30Mbps downstream, 2 Mbps upstream
network of cable and fiber attaches homes to ISP router
homes share access to router
deployment: available via cable TV companies

13. Company access: local area networks
company/univ local area network (LAN) connects end system to edge router
Ethernet:
10 Mbs, 100Mbps, 1Gbps, 10Gbps Ethernet
modern configuration: end systems connect into Ethernet switch
LANs: chapter 5

14. Wireless access networks
shared wireless access network connects end system to router
via base station also known as “access point”
wireless LANs:
802.11b/g (WiFi): 11 or 54 Mbps
wider-area wireless access
provided by telco operator
~1Mbps over cellular system (EVDO, HSDPA)
next up (?): WiMAX (10’s Mbps) over wide area
router
base
station
mobile
hosts

15. Characteristics of selected wireless link standards
200
802.11n 54
802.11a,g
802.11a,g point-to-point
data
5-11
802.11b
(WiMAX) 4
3G cellular
enhanced
UMTS/WCDMA-HSPDA, CDMA2000-1xEVDO
Data rate (Mbps) 1
802.15
.384
UMTS/WCDMA, CDMA2000 3G
.056
IS-95, CDMA, GSM 2G
Indoor
10-30m
Outdoor
50-200m
Mid-range
outdoor
200m – 4 Km
Long-range
outdoor
5Km – 20 Km

16. Home networks Typical home network components: DSL or cable modem
router/firewall/NAT
Ethernet
wireless access
point
wireless
laptops
to/from
cable
headend
cable
modem
router/
firewall
wireless
access
point
Ethernet

17. The Network Core mesh of interconnected routers
the fundamental question: how is data transferred through net?
circuit switching: dedicated circuit per call: telephone net
packet-switching: data sent thru net in discrete “chunks”

18. Network Core: Packet Switching
each end-end data stream divided into packets
user A, B packets share network resources
each packet uses full link bandwidth
resources used as needed
resource contention:
aggregate resource demand can exceed amount available
congestion: packets queue, wait for link use
store and forward: packets move one hop at a time
Node receives complete packet before forwarding
Bandwidth division into “pieces”
Dedicated allocation
Resource reservation

19. Packet Switching: Statistical Multiplexing
100 Mb/s
Ethernet C A
statistical multiplexing
1.5 Mb/s B
queue of packets
waiting for output
link D E
Sequence of A & B packets does not have fixed pattern, bandwidth shared on demand  statistical multiplexing.
TDM: each host gets same slot in revolving TDM frame.

20. Packet-switching: store-and-forwardL R R R
takes L/R seconds to transmit (push out) packet of L bits on to link at R bps
store and forward: entire packet must arrive at router before it can be transmitted on next link
delay = 3L/R (assuming zero propagation delay)
Example:
L = 7.5 Mbits
R = 1.5 Mbps
transmission delay = 15 sec
more on delay shortly …

21. How do loss and delay occur?
packets queue in router buffers
packet arrival rate to link exceeds output link capacity
packets queue, wait for turn
packet being transmitted (delay) A
free (available) buffers: arriving packets
dropped (loss) if no free buffers
packets queueing (delay) B

22. Four sources of packet delay
1. nodal processing:
check bit errors
determine output link
2. queueing
time waiting at output link for transmission
depends on congestion level of router A B
propagation
transmission
nodal
processing
queueing

23. Delay in packet-switched networks
3. Transmission delay:
R=link bandwidth (bps)
L=packet length (bits)
time to send bits into link = L/R
4. Propagation delay:
d = length of physical link
s = propagation speed in medium (~2x108 m/sec)
propagation delay = d/s
Note: s and R are very different quantities! A B
propagation
transmission
nodal
processing
queueing

24. Nodal delay dproc = processing delay dqueue = queuing delay
typically a few microsecs or less
dqueue = queuing delay
depends on congestion
dtrans = transmission delay
= L/R, significant for low-speed links
dprop = propagation delay
a few microsecs to hundreds of msecs

25. Queueing delay (revisited)
R=link bandwidth (bps)
L=packet length (bits)
a=average packet arrival rate
traffic intensity = La/R
La/R ~ 0: average queueing delay small
La/R -> 1: delays become large
La/R > 1: more “work” arriving than can be serviced, average delay infinite!

26. “Real” Internet delays and routes
What do “real” Internet delay & loss look like?
Traceroute program: provides delay measurement from source to router along end-end Internet path towards destination. For all i:
sends three packets that will reach router i on path towards destination
router i will return packets to sender
sender times interval between transmission and reply.
3 probes
3 probes
3 probes

27. “Real” Internet delays and routes
traceroute: gaia.cs.umass.edu to
Three delay measurements from
gaia.cs.umass.edu to cs-gw.cs.umass.edu
1 cs-gw ( ) 1 ms 1 ms 2 ms
2 border1-rt-fa5-1-0.gw.umass.edu ( ) 1 ms 1 ms 2 ms
3 cht-vbns.gw.umass.edu ( ) 6 ms 5 ms 5 ms
4 jn1-at wor.vbns.net ( ) 16 ms 11 ms 13 ms
5 jn1-so wae.vbns.net ( ) 21 ms 18 ms 18 ms
6 abilene-vbns.abilene.ucaid.edu ( ) 22 ms 18 ms 22 ms
7 nycm-wash.abilene.ucaid.edu ( ) 22 ms 22 ms 22 ms
( ) 104 ms 109 ms 106 ms
9 de2-1.de1.de.geant.net ( ) 109 ms 102 ms 104 ms
10 de.fr1.fr.geant.net ( ) 113 ms 121 ms 114 ms
11 renater-gw.fr1.fr.geant.net ( ) 112 ms 114 ms 112 ms
12 nio-n2.cssi.renater.fr ( ) 111 ms 114 ms 116 ms
13 nice.cssi.renater.fr ( ) 123 ms 125 ms 124 ms
14 r3t2-nice.cssi.renater.fr ( ) 126 ms 126 ms 124 ms
15 eurecom-valbonne.r3t2.ft.net ( ) 135 ms 128 ms 133 ms
( ) 126 ms 128 ms 126 ms
17 * * *
18 * * *
19 fantasia.eurecom.fr ( ) 132 ms 128 ms 136 ms
trans-oceanic
link
* means no response (probe lost, router not replying)

28. Packet loss
queue (aka buffer) preceding link in buffer has finite capacity
packet arriving to full queue dropped (aka lost)
lost packet may be retransmitted by previous node, by source end system, or not at all
buffer
(waiting area)
packet being transmitted A B
packet arriving to
full buffer is lost

29. Throughput
throughput: rate (bits/time unit) at which bits transferred between sender/receiver
instantaneous: rate at given point in time
average: rate over long(er) period of time
pipe that can carry
fluid at rate
Rc bits/sec)
pipe that can carry
fluid at rate
Rs bits/sec)
link capacity
Rs bits/sec
link capacity
Rc bits/sec
server sends bits
(fluid) into pipe
server, with
file of F bits
to send to client

30. Throughput (more) Rs < Rc What is average end-end throughput?
Rc bits/sec
Rs bits/sec
Rs > Rc What is average end-end throughput?
Rs bits/sec
Rc bits/sec
link on end-end path that constrains end-end throughput
bottleneck link

31. Throughput: Internet scenarioRs
per-connection end-end throughput: min(Rc,Rs,R/10)
in practice: Rc or Rs is often bottleneck Rs Rs R Rc Rc Rc
10 connections (fairly) share backbone bottleneck link R bits/sec

32. Protocol “Layers” Networks are complex! many “pieces”: hosts routers
links of various media
applications
protocols
hardware, software

33. Why layering? Dealing with complex systems:
explicit structure allows identification, relationship of complex system’s pieces
layered reference model for discussion
modularization eases maintenance, updating of system
change of implementation of layer’s service transparent to rest of system
e.g., change in gate procedure doesn’t affect rest of system
layering considered harmful?

34. Internet protocol stack
application: supporting network applications
FTP, SMTP, HTTP
transport: process-process data transfer
TCP, UDP
network: routing of datagrams from source to destination
IP, routing protocols
link: data transfer between neighboring network elements
PPP, Ethernet
physical: bits “on the wire”
application
transport
network
link
physical

35. ISO/OSI reference model
presentation: allow applications to interpret meaning of data, e.g., encryption, compression, machine-specific conventions
session: synchronization, checkpointing, recovery of data exchange
Internet stack “missing” these layers!
these services, if needed, must be implemented in application
needed?
application
presentation
session
transport
network
link
physical

36. Encapsulation source destination application transport network link
message M
application
transport
network
link
physical
segment Ht M Ht
datagram Ht Hn M Hn
frame Ht Hn Hl M
link
physical
switch
destination
network
link
physical Ht Hn M Ht Hn Hl M M
application
transport
network
link
physical Ht Hn M Ht M Ht Hn M
router Ht Hn Hl M

37. Internet apps: application, transport protocols
layer protocol
SMTP [RFC 2821]
Telnet [RFC 854]
HTTP [RFC 2616]
FTP [RFC 959]
proprietary
(e.g. RealNetworks)
(e.g., Vonage,Dialpad)
Underlying
transport protocol
TCP
TCP or UDP
typically UDP
Application
remote terminal access
Web
file transfer
streaming multimedia
Internet telephony

38. Internet transport-layer protocols
reliable, in-order delivery (TCP)
congestion control
flow control
connection setup
unreliable, unordered delivery: UDP
no-frills extension of “best-effort” IP
services not available:
delay guarantees
bandwidth guarantees
application
transport
network
data link
physical
network
data link
physical
network
data link
physical
logical end-end transport
network
data link
physical
network
data link
physical
network
data link
physical
network
data link
physical
application
transport
network
data link
physical

39. UDP: User Datagram Protocol [RFC 768]
“no frills,” “bare bones” Internet transport protocol
“best effort” service, UDP segments may be:
lost
delivered out of order to app
connectionless:
no handshaking between UDP sender, receiver
each UDP segment handled independently of others
Why is there a UDP?
no connection establishment (which can add delay)
simple: no connection state at sender, receiver
small segment header
no congestion control: UDP can blast away as fast as desired

40. UDP: more other UDP uses often used for streaming multimedia apps
loss tolerant
rate sensitive
other UDP uses
DNS
SNMP
reliable transfer over UDP: add reliability at application layer
application-specific error recovery!
32 bits
source port #
dest port #
Length, in
bytes of UDP
segment,
including
header
length
checksum
Application
data
(message)
UDP segment format

41. TCP: Overview RFCs: 793, 1122, 1323, 2018, 2581 point-to-point:
one sender, one receiver
reliable, in-order byte steam:
no “message boundaries”
pipelined:
TCP congestion and flow control set window size
send & receive buffers
full duplex data:
bi-directional data flow in same connection
MSS: maximum segment size
connection-oriented:
handshaking (exchange of control msgs) init’s sender, receiver state before data exchange
flow controlled:
sender will not overwhelm receiver

42. TCP segment structure source port # dest port # application data
32 bits
application
data
(variable length)
sequence number
acknowledgement number
Receive window
Urg data pnter
checksum F S R P A U
head
len
not
used
Options (variable length)
URG: urgent data
(generally not used)
counting
by bytes
of data
(not segments!)
ACK: ACK #
valid
PSH: push data now
(generally not used)
# bytes
rcvr willing
to accept
RST, SYN, FIN:
connection estab
(setup, teardown
commands)
Internet
checksum
(as in UDP)

43. simple telnet scenario
TCP seq. #’s and ACKs
Seq. #’s:
byte stream “number” of first byte in segment’s data
ACKs:
seq # of next byte expected from other side
cumulative ACK
Q: how receiver handles out-of-order segments
A: TCP spec doesn’t say, - up to implementor
Host A
Host B
User
types
‘C’
Seq=42, ACK=79, data = ‘C’
host ACKs
receipt of
‘C’, echoes
back ‘C’
Seq=79, ACK=43, data = ‘C’
host ACKs
receipt
of echoed
‘C’
Seq=43, ACK=80
time
simple telnet scenario

44. TCP: retransmission scenarios
Host A
Seq=92, 8 bytes data
ACK=100
loss
timeout
lost ACK scenario
Host B X
time
Host A
Host B
Seq=92 timeout
Seq=92, 8 bytes data
Seq=100, 20 bytes data
ACK=100
ACK=120
Sendbase
= 100
Seq=92, 8 bytes data
SendBase
= 120
Seq=92 timeout
ACK=120
SendBase
= 100
SendBase
= 120
premature timeout
time

45. TCP retransmission scenarios (more)
Host A
Seq=92, 8 bytes data
ACK=100
loss
timeout
Cumulative ACK scenario
Host B X
Seq=100, 20 bytes data
ACK=120
time
SendBase
= 120

46. Fast Retransmit Time-out period often relatively long:
long delay before resending lost packet
Detect lost segments via duplicate ACKs.
Sender often sends many segments back-to-back
If segment is lost, there will likely be many duplicate ACKs.
If sender receives 3 ACKs for the same data, it supposes that segment after ACKed data was lost:
fast retransmit: resend segment before timer expires

47. TCP congestion control: additive increase, multiplicative decrease
Approach: increase transmission rate (window size), probing for usable bandwidth, until loss occurs
additive increase: increase CongWin by 1 MSS every RTT until loss detected
multiplicative decrease: cut CongWin in half after loss
Saw tooth
behavior: probing
for bandwidth
congestion window size
time

48. TCP Congestion Control: details
sender limits transmission:
LastByteSent-LastByteAcked
 CongWin
Roughly,
CongWin is dynamic, function of perceived network congestion
How does sender perceive congestion?
loss event = timeout or 3 duplicate acks
TCP sender reduces rate (CongWin) after loss event
three mechanisms:
AIMD
slow start
conservative after timeout events
rate =
CongWin
RTT
Bytes/sec

49. TCP Slow Start When connection begins, CongWin = 1 MSS
Example: MSS = 500 bytes & RTT = 200 msec
initial rate = 20 kbps
available bandwidth may be >> MSS/RTT
desirable to quickly ramp up to respectable rate
When connection begins, increase rate exponentially fast until first loss event

50. TCP Slow Start (more)
When connection begins, increase rate exponentially until first loss event:
double CongWin every RTT
done by incrementing CongWin for every ACK received
Summary: initial rate is slow but ramps up exponentially fast
Host A
Host B
one segment
RTT
two segments
four segments
time

51. Refinement Implementation:
Q: When should the exponential increase switch to linear?
A: When CongWin gets to 1/2 of its value before timeout.
Implementation:
Variable Threshold
At loss event, Threshold is set to 1/2 of CongWin just before loss event

52. Refinement: inferring loss
After 3 dup ACKs:
CongWin is cut in half
window then grows linearly
But after timeout event:
CongWin instead set to 1 MSS;
window then grows exponentially
to a threshold, then grows linearly
Philosophy:
3 dup ACKs indicates network capable of delivering some segments
timeout indicates a “more alarming” congestion scenario

53. Summary: TCP Congestion Control
When CongWin is below Threshold, sender in slow-start phase, window grows exponentially.
When CongWin is above Threshold, sender is in congestion-avoidance phase, window grows linearly.
When a triple duplicate ACK occurs, Threshold set to CongWin/2 and CongWin set to Threshold.
When timeout occurs, Threshold set to CongWin/2 and CongWin is set to 1 MSS.

54. TCP throughput
What’s the average throughout of TCP as a function of window size and RTT?
Ignore slow start
Let W be the window size when loss occurs.
When window is W, throughput is W/RTT
Just after loss, window drops to W/2, throughput to W/2RTT.
Average throughout: .75 W/RTT

55. TCP Throughput In the case of loss,
Average throughput of a TCP connection =
L – loss rate
RTT – round trip time
MSS – maximum segment size

56. TCP Fairness
Fairness goal: if K TCP sessions share same bottleneck link of bandwidth R, each should have average rate of R/K
TCP connection 1
bottleneck
router
capacity R
TCP
connection 2

57. Fairness (more) Fairness and parallel TCP connections Fairness and UDP
nothing prevents app from opening parallel connections between 2 hosts.
Web browsers do this
Example: link of rate R supporting 9 connections;
new app asks for 1 TCP, gets rate R/10
new app asks for 11 TCPs, gets R/2 !
Fairness and UDP
Multimedia apps often do not use TCP
do not want rate throttled by congestion control
Instead use UDP:
pump audio/video at constant rate, tolerate packet loss
Research area: TCP friendly