Chapter - 2 Lec 3&4

Network and Transport
Layer
Ø Lecture 3:
2.1 Internet transport protocol
ü UDP
ü TCP Computer Networking: A
ü SCTP Top-Down Approach
2.2 Congestion control 8th edition
Jim Kurose, Keith Ross
Pearson, 2020
Chapter 2: Network and Transport layer

ü UDP
ü TCP
ü SCTP
2.2 Congestion control
2.3 Routing Algorithms
2.4 IPv6
Transport Layer: 3-2

Chapter 2: Network and Transport laye

ü UDP
ü TCP Lecture 3
ü SCTP
2.2 Congestion control
2.3 Routing Algorithms
2.4 IPv6

Internet Transport layer: overview
Our goal:
§ understand principles § learn about Internet transport

behind transport layer layer protocols:
services: • UDP: connectionless transport
• TCP: connection-oriented reliable
transport

Transport layer:
§ Transport-layer services
§ Connectionless transport: UDP
§ Connection-oriented transport: TCP

Transport services and protocols
application
transport
§ provide logical communication

network
mobile network
data link
physical
between application processes national or global ISP
running on different hosts
l og
ica
l en
§ transport protocols actions in end
d-e
systems:
nd
local or
tra
• sender: breaks application messages regional ISP
nsp
into segments, passes to network layer
or t
home network content
• receiver: reassembles segments into provider
network
messages, passes to application layer datacenter
application
transport
network
network
§ two transport protocols available to data link

physical
Internet applications enterprise

network
• TCP, UDP
Transport vs. network layer services and protocols
household analogy:
10 kids in A’s house sending
letters to 10 kids in B’s house:
§ hosts = houses
§ processes = kids
§ app messages = letters in
envelopes
§ transport protocol = A and B
who demux to in-house siblings
§ network-layer protocol = postal
service

Transport vs. network layer services and protocols
household analogy:
§transport layer:
10 kids in A’s house sending
communication between letters to 10 kids in B’s house:
processes § hosts = houses
• relies on, enhances, network § processes = kids
layer services § app messages = letters in
envelopes
§network layer: § transport protocol = A and B
communication between who demux to in-house siblings
hosts § network-layer protocol = postal
service

Transport Layer Actions
Sender:
application § is passed an application- app. msg
application
layer message
transport
§ determines segment TTh htransport
app. msg
header fields values
network (IP)
§ creates segment network (IP)
link
§ passes segment to IP link
physical physical

Transport Layer Actions
Receiver:
application § receives segment from IP application
§ checks header values
app. msg
transport § extracts application-layer transport
message
network (IP)
network (IP) § demultiplexes message up
link to application via socket link
physical physical
Th app. msg

Two principal Internet transport protocols
application
transport
§ TCP: Transmission Control Protocol mobile

network
network
data link
physical
• reliable, in-order delivery national or global ISP
• congestion control
l og
ica
• flow control
l en
d-e
• connection setup
nd
local or
§ UDP: User Datagram Protocol
tra
regional ISP
nsp
• unreliable, unordered delivery
or t
home network content
• no-frills extension of “best-effort” IP provider
network datacenter
application
network
§ services not available: transport

network
data link
• delay guarantees physical
• bandwidth guarantees enterprise

network

HTTP server
client
application application
HTTP msg
transport
transport network transport

network link network
link physical link
physical physical

HTTP server
client
HTTP msg
Ht HTTP msg
transport

link physical link
physical physical

HTTP server
client
HTTP msg
Ht HTTP msg
transport
Hnnetwork
Ht HTTP msg
transport transport
link physical link
physical physical

HTTP server
client
transport

link physical link
physical physical
Hn Ht HTTP msg

HTTP server
client
HTTP msg
Ht HTTP msg
transport

Hn Hnetwork
t HTTP msg
link network
link physical link
physical physical

HTTP server
client
HTTP msg
Ht HTTP msg
transport
Htransport
t HTTP msg network transport
link physical link
physical physical

client
HTTP msg
HTTP msg Ht HTTP msg
transport

link physical link
physical physical

Q: how did transport layer know to deliver message to Firefox
browser process rather then Netflix process or Skype process?
client
HTTP msg
HTTP msg Ht HTTP msg
transport

link physical link
physical physical

Chapter 2: outline

UDP: User Datagram Protocol
Why is there a UDP?
§ “no frills,” “bare bones”
Internet transport protocol § no connection
establishment (which can
§ “best effort” service, UDP add RTT delay)
segments may be: § simple: no connection state
• lost at sender, receiver
• delivered out-of-order to app § small header size
§ connectionless: § no congestion control
• no handshaking between UDP § UDP can blast away as fast as
desired!
sender, receiver
§ can function in the face of
• each UDP segment handled congestion
independently of others
UDP: User Datagram Protocol
§ UDP use:
§ streaming multimedia apps (loss tolerant, rate sensitive)
§ DNS
§ SNMP
§ HTTP/3
§ if reliable transfer needed over UDP (e.g., HTTP/3):
§ add needed reliability at application layer
§ add congestion control at application layer

UDP: User Datagram Protocol [RFC 768]

UDP: Transport Layer Actions
SNMP client SNMP server
transport transport
(UDP) (UDP)
network (IP) network (IP)
link link
physical physical


UDP sender actions:
application § is passed an application- SNMP msg
application
layer message
transport § determines UDP segment UDPhtransport
UDP h SNMP msg
(UDP) header fields values (UDP)
network (IP)
§ creates UDP segment network (IP)
link
§ passes segment to IP link
physical physical


UDP receiver actions:
application § receives segment from IP application
§ checks UDP checksum
transport
transport
SNMP msg header value
(UDP) § extracts application-layer (UDP)
h SNMP(IP)
network
UDP msg message network (IP)
§ demultiplexes message up
link to application via socket link
physical physical

UDP segment header
32 bits
source port # dest port #
length checksum
application length, in bytes of

data UDP segment,
(payload) including header
data to/from
UDP segment format application layer

UDP checksum
Goal: detect errors (i.e., flipped bits) in transmitted segment
1st number 2nd number sum
Transmitted: 5 6 11
Received: 4 6 11
receiver-computed
checksum
= sender-computed
checksum (as received)

Internet checksum
Goal: detect errors (i.e., flipped bits) in transmitted segment
sender: receiver:
§ treat contents of UDP § compute checksum of received
segment (including UDP header segment
fields and IP addresses) as
sequence of 16-bit integers § check if computed checksum equals
§ checksum: addition (one’s checksum field value:
complement sum) of segment • not equal - error detected
content • equal - no error detected. But maybe
§ checksum value put into errors nonetheless? More later ….
UDP checksum field
Internet checksum: an example
example: add two 16-bit integers
1 1 1 0 0 1 1 0 0 1 1 0 0 1 1 0
1 1 0 1 0 1 0 1 0 1 0 1 0 1 0 1
wraparound 1 1 0 1 1 1 0 1 1 1 0 1 1 1 0 1 1
sum 1 0 1 1 1 0 1 1 1 0 1 1 1 1 0 0
checksum 0 1 0 0 0 1 0 0 0 1 0 0 0 0 1 1
Note: when adding numbers, a carryout from the most significant bit needs to be
added to the result
* Check out the online interactive exercises for more examples: http://gaia.cs.umass.edu/kurose_ross/interactive/
Internet checksum: weak protection!
example: add two 16-bit integers
0 1
1 1 1 0 0 1 1 0 0 1 1 0 0 1 1 0 1 0
1 1 0 1 0 1 0 1 0 1 0 1 0 1 0 1
wraparound 1 1 0 1 1 1 0 1 1 1 0 1 1 1 0 1 1 Even though
numbers have
sum 1 0 1 1 1 0 1 1 1 0 1 1 1 1 0 0 changed (bit flips),
no change in
checksum 0 1 0 0 0 1 0 0 0 1 0 0 0 0 1 1 checksum!

Summary: UDP
§ “no frills” protocol:
• segments may be lost, delivered out of order
• best effort service: “send and hope for the best”
§ UDP has its plusses:
• no setup/handshaking needed (no RTT incurred)
• can function when network service is compromised
• helps with reliability (checksum)
§ build additional functionality on top of UDP in application layer
(e.g., HTTP/3)
Chapter 2: outline

• segment structure
• flow control
• connection management

TCP: overview RFCs: 793,1122, 2018, 5681, 7323
§ point-to-point: § cumulative ACKs

• one sender, one receiver § pipelining:
§ reliable, in-order byte • TCP congestion and flow control
stream: set window size
• no “message boundaries" § connection-oriented:
§ full duplex data: • handshaking (exchange of control
• bi-directional data flow in messages) initializes sender,
same connection receiver state before data exchange
• MSS: maximum segment size § flow controlled:
• sender will not overwhelm receiver

TCP segment structure
32 bits
source port # dest port # segment seq #: counting

ACK: seq # of next expected sequence number bytes of data into bytestream
byte; A bit: this is an ACK (not segments!)
acknowledgement number
length (of TCP header) head not
len used C EUAP R SF receive window flow control: # bytes
Internet checksum checksum Urg data pointer receiver willing to accept
options (variable length)

C, E: congestion notification
TCP options
application data sent by
RST, SYN, FIN: connection data application into
management (variable length) TCP socket

TCP sequence numbers, ACKs
outgoing segment from sender
Sequence numbers: source port # dest port #
sequence number
• byte stream “number” of acknowledgement number
first byte in segment’s data

rwnd
checksum urg pointer
window size
Acknowledgements: N
• seq # of next byte expected

from other side sender sequence number space
• cumulative ACK sent sent, not- usable not

ACKed yet ACKed but not usable
Q: how receiver handles out-of- (“in-flight”) yet sent
order segments outgoing segment from receiver
• A: TCP spec doesn’t say, - up

source port # dest port #
sequence number
to implementor acknowledgement number

A rwnd
checksum urg pointer
TCP sequence numbers, ACKs
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
simple telnet scenario

TCP round trip time, timeout
Q: how to set TCP timeout Q: how to estimate RTT?
value? § SampleRTT:measured time
§ longer than RTT, but RTT varies! from segment transmission until
ACK receipt
§ too short: premature timeout,
• ignore retransmissions
unnecessary retransmissions
§ SampleRTT will vary, want
§ too long: slow reaction to estimated RTT “smoother”
segment loss • average several recent measurements,
not just current SampleRTT

EstimatedRTT = (1- )*EstimatedRTT + *SampleRTT
§ exponential weighted moving average (EWMA)
§ influence of past sample decreases exponentially fast
§ typical value:  = 0.125 (RFC 6298)
RTT: gaia.cs.umass.edu to fantasia.eurecom.fr
RTT (milliseconds)
sampleRTT
EstimatedRTT
time (seconds) Transport Layer: 3-43

§ timeout interval: EstimatedRTT plus “safety margin”
• large variation in EstimatedRTT: want a larger safety margin
TimeoutInterval = EstimatedRTT + 4*DevRTT
estimated RTT “safety margin”
§ DevRTT: EWMA of SampleRTT deviation from EstimatedRTT:

DevRTT = (1-)*DevRTT + *|SampleRTT-EstimatedRTT|
(typically,  = 0.25)
* Check out the online interactive exercises for more examples: http://gaia.cs.umass.edu/kurose_ross/interactive/
TCP Sender (simplified)
event: data received from event: timeout
application § retransmit segment that
caused timeout
§ create segment with seq #
§ restart timer
§ seq # is byte-stream number
of first data byte in segment
event: ACK received
§ start timer if not already
running § if ACK acknowledges
previously unACKed segments
• think of timer as for oldest
unACKed segment • update what is known to be
ACKed
• expiration interval:
TimeOutInterval • start timer if there are still
unACKed segments
TCP Receiver: ACK generation [RFC 5681]
Event at receiver TCP receiver action
arrival of in-order segment with delayed ACK. Wait up to 500ms
expected seq #. All data up to for next segment. If no next segment,
expected seq # already ACKed send ACK
arrival of in-order segment with immediately send single cumulative

expected seq #. One other ACK, ACKing both in-order segments
segment has ACK pending
arrival of out-of-order segment immediately send duplicate ACK,

higher-than-expect seq. # . indicating seq. # of next expected byte
Gap detected
arrival of segment that immediate send ACK, provided that

partially or completely fills gap segment starts at lower end of gap

TCP: retransmission scenarios
Host A Host B Host A Host B
SendBase=92
Seq=92, 8 bytes of data Seq=92, 8 bytes of data
Seq=100, 20 bytes of data

timeout
timeout
ACK=100
X
ACK=100
ACK=120
Seq=92, 8 bytes of data Seq=92, 8

SendBase=100 bytes of data send cumulative
SendBase=120 ACK for 120
ACK=100
ACK=120
SendBase=120
lost ACK scenario premature timeout

TCP: retransmission scenarios
Host A Host B

ACK=100
X
ACK=120
cumulative ACK covers

for earlier lost ACK

TCP fast retransmit
Host A Host B
TCP fast retransmit
if sender receives 3 additional
ACKs for same data (“triple Seq=92
Seq=1
, 8 byte
s of dat
a
duplicate ACKs”), resend unACKed 00, 20
bytes
of dat
segment with smallest seq #
a
X
§ likely that unACKed segment lost,
100
so don’t wait for timeout ACK=
=100
timeout
A CK
=100
ACK
K=100
Receipt of three duplicate ACKs AC
indicates 3 segments received Seq=100, 20 bytes of data
after a missing segment – lost

segment is likely. So retransmit!
TCP flow control
application
Q: What happens if network Application removing
process
layer delivers data faster than data from TCP socket
buffers
application layer removes TCP socket
data from socket buffers? receiver buffers
TCP
code
Network layer
delivering IP datagram
payload into TCP
IP
socket buffers
code
from sender
receiver protocol stack

TCP flow control
application
process
buffers
TCP
code
Network layer
delivering IP datagram
payload into TCP
IP
socket buffers
code
from sender

TCP flow control
application
process
buffers
TCP
code
receive window
flow control: # bytes
receiver willing to accept IP
code
from sender

TCP flow control
application
process
buffers
TCP
flow control code
receiver controls sender, so

sender won’t overflow IP
receiver’s buffer by code
transmitting too much, too fast

from sender

TCP flow control
§ TCP receiver “advertises” free buffer
space in rwnd field in TCP header to application process
• RcvBuffer size set via socket
options (typical default is 4096 bytes) RcvBuffer buffered data
• many operating systems autoadjust
rwnd
RcvBuffer free buffer space
§ sender limits amount of unACKed

(“in-flight”) data to received rwnd TCP segment payloads
§ guarantees receive buffer will not TCP receiver-side buffering

overflow

TCP flow control
flow control: # bytes receiver willing to accept
§ TCP receiver “advertises” free buffer

space in rwnd field in TCP header
• RcvBuffer size set via socket
receive window
options (typical default is 4096 bytes)
• many operating systems autoadjust
RcvBuffer
§ sender limits amount of unACKed
(“in-flight”) data to received rwnd
§ guarantees receive buffer will not
overflow
TCP segment format

TCP connection management
before exchanging data, sender/receiver “handshake”:
§ agree to establish connection (each knowing the other willing to establish connection)
§ agree on connection parameters (e.g., starting seq #s)
connection state: ESTAB connection state: ESTAB

connection variables: connection Variables:
seq # client-to-server seq # client-to-server
server-to-client server-to-client
rcvBuffer size rcvBuffer size
at server,client at server,client
network network
Socket clientSocket = Socket connectionSocket =

newSocket("hostname","port number"); welcomeSocket.accept();
Agreeing to establish a connection
2-way handshake:
Q: will 2-way handshake always

Let’s talk work in network?
ESTAB
ESTAB
OK § variable delays
§ retransmitted messages (e.g.
req_conn(x)) due to message loss
§ message reordering
choose x § can’t “see” other side
req_conn(x)
ESTAB
acc_conn(x)
ESTAB

2-way handshake scenarios
choose x
req_conn(x)
ESTAB
acc_conn(x)
ESTAB
data(x+1) accept
ACK(x+1) data(x+1)
connection
x completes
No problem!

choose x
req_conn(x)
ESTAB
retransmit acc_conn(x)
req_conn(x)
ESTAB
req_conn(x)
connection
client x completes server
terminates forgets x
ESTAB
acc_conn(x)
Problem: half open
connection! (no client)
choose x
req_conn(x)
ESTAB
retransmit acc_conn(x)
req_conn(x)
ESTAB
data(x+1) accept
data(x+1)
retransmit
data(x+1)
connection
x completes server
client
terminates forgets x
req_conn(x)
ESTAB
data(x+1) accept
data(x+1)
Problem: dup data
accepted!
TCP 3-way handshake
Server state
serverSocket = socket(AF_INET,SOCK_STREAM)
Client state serverSocket.bind((‘’,serverPort))
serverSocket.listen(1)
clientSocket = socket(AF_INET, SOCK_STREAM) connectionSocket, addr = serverSocket.accept()
LISTEN
clientSocket.connect((serverName,serverPort)) LISTEN
choose init seq num, x
send TCP SYN msg
SYNSENT SYNbit=1, Seq=x
choose init seq num, y
send TCP SYNACK
msg, acking SYN SYN RCVD
SYNbit=1, Seq=y
ACKbit=1; ACKnum=x+1
received SYNACK(x)
ESTAB indicates server is live;
send ACK for SYNACK;
this segment may contain ACKbit=1, ACKnum=y+1
client-to-server data
received ACK(y)
indicates client is live
ESTAB

Closing a TCP connection
§ client, server each close their side of connection
• send TCP segment with FIN bit = 1
§ respond to received FIN with ACK
• on receiving FIN, ACK can be combined with own FIN
§ simultaneous FIN exchanges can be handled

Ø Reading and Note take Assignment
§ Page 263 - 279 of book
“computer networking
A top-down Approach”

Stream Control Transmission Protocol (SCTP)
ØSuitable transport protocol for signaling traffic (SS7) over IP
– may be used for traditional Internet services such as those based on HTTP
and SIP
– sometimes called the “Next Generation TCP”
– Internet standard RFC2960
Ø Service model
– unicast transport protocol
– provides reliable, message-oriented data delivery
• SCTP sends “complete” messages whereas TCP is byte oriented and
does not preserve any structure in the byte stream
• reliable = lost/corrupted messages are retransmitted

Ø Other features
– functionality is TCP-like with modifications due to message oriented
principle
• TCP-like sliding window, flow control and congestion control
• uses selective acknowledgements (SACKs) to report out of
sequence data
– multistreaming: data stream can be partitioned into multiple streams,
each controlled independently
– multihoming: for redundancy, an SCTP endpoint can be associated with
multiple IP addresses

��
Congestion Control vs. Flow Control
�Congestion Control has to due with making sure the subnets are able to
carry the offered traffic. Thus it is a global issue involving the behavior of
• all the hosts,
• all the routers,
• the store-and forwarding processing within the routers, and
• all the other factors that tend to diminish the carrying capacity of the
subnet.
� Flow Control relates to point-to-point traffic between a given sender a given

receiver. Its job is to make sure that a fast sender cannot continually transmit
data faster than the receiver is able to absorb it.
• It frequently involves some direct feedback from the receiver to the
sender to tell the sender how things are doing at the other end.
Congestion Control vs. Flow Control Example
� Flow Control Problem: Consider a fiber optic network with a capacity of

1000 Gbps on which a supercomputer is trying to transfer a file to a personal
computer at 1 Gbps. Although there is not congestion the supercomputer
has to frequently stop in order to allow PC to catch up.
� Congestion Control Problem: Consider a store-and-forward network with

1 Mbps lines and 1000 large computers, half of which are trying to transfer
files at 100 kbps to the other half creating a traffic of 500x100 kbps = 50000
kbps = 50 Mbps. The problem here is that this traffic exceeds the capacity of
what the network can handle.
Virtual circuit subnets
§ Dynamic approach: act when needed
§ Admission control
• No new virtual circuits when congestion is signalled
• e.g. telephone network
§ Route new virtual circuits around problem areas
Virtual circuit subnets
§ Dynamic approach
§ Admission control
§ Route new virtual circuits around problem areas
§ Negotiation when virtual circuit is set up

• About kind of traffic + service desired
• Resource reservation in subnet
• Line capacity
• Buffers in routers
èNo congestion
èUnused resources
General techniques
§ Approaches: reduce traffic by
• Requesting senders
local measurements + info packets to senders
+ hope for ….
• Throwing away packets

local measurements + local actions to reduce load
Source based approach
§ Warning bit
• Output line in warning state
• Warning bit set in header
• Destination copies bit into next ack
• Source cuts back traffic
• Algorithm at source
• As long as warning bits arrive: reduce traffic
• Less warning bits: increase traffic
• Problems
• voluntary action of host!
• correct source selected?
• Used in
• DecNet
• Frame relay
§ Choke packet
• In case of overload: router sends choke packet to host causing the overload
• Host receiving choke packet
• reduces traffic to the specified destination
• ignores choke packets for a fixed interval
• new choke packets during next listening interval?
• Yes: reduce traffic
• No: increase traffic
• Problems:
• voluntary action of host!
• correct host selected?
§ Choke packets:
• Example showing
slow reaction
• Solution: Hop-by-
Hop choke packets
81
§ Hop-by-Hop choke
packets
• Have choke packet
take effect at every
hop
• Problem: more
buffers needed in
routers
82
Load shedding
§ Throw away packets that cannot be handled!!
§ Packet selection?
§ Random early detection
• Discard packets before all buffer space is exhausted
• Routers maintain running average of queue lengths
• Select at random a packet
• Inform source?
• Send choke packet? è more load!!
• No reporting
• When does it work?
• Source slows down when packets are lost
Congestion: jitter control
§Important for audio and video applications?
• not delay
• variance of delay
Congestion: jitter control
§ Jitter = variation in packet delay
§ Compute feasible mean value for delay
• compute expected transit time for each hop
• router checks to see if packet is
• behind
• ahead schedule
• behind: forward packet asap
• ahead: hold back packet to get it on schedule again
§ Buffering? Depends on characteristics:
• Video on demand: ok
• Videoconferencing: nok
ØLecture 4
Routing Algorithms/protocols

Chapter - 2 Lec 3&4

Uploaded by

Copyright:

Available Formats

Chapter - 2 Lec 3&4

Uploaded by

Document Information

Original Title

Copyright

Available Formats

Share this document

Share or Embed Document

Sharing Options

Did you find this document useful?

Is this content inappropriate?

Copyright:

Available Formats

Chapter - 2 Lec 3&4

Uploaded by

Copyright:

Available Formats

Network and Transport

2.1 Internet transport protocol

Transport Layer: 3-2

2.1 Internet transport protocol

Transport Layer: 3-3

§ understand principles § learn about Internet transport

Transport Layer: 3-4

Transport Layer: 3-5

§ provide logical communication

between application processes national or global ISP

running on different hosts

§ two transport protocols available to data link

Internet applications enterprise

Transport Layer: 3-7

Transport Layer: 3-8

Transport Layer: 3-9

Transport Layer: 3-10

§ TCP: Transmission Control Protocol mobile

• reliable, in-order delivery national or global ISP

§ services not available: transport

• delay guarantees physical

• bandwidth guarantees enterprise

Transport Layer: 3-11

transport network transport

Transport Layer: 3-12

transport network transport

Transport Layer: 3-13

Transport Layer: 3-14

transport network transport

Transport Layer: 3-15

transport network transport

Transport Layer: 3-16

Transport Layer: 3-17

transport network transport

Transport Layer: 3-18

transport network transport

Transport Layer: 3-19

Transport Layer: 3-20

Transport Layer: 3-22

Transport Layer: 3-23

SNMP client SNMP server

network (IP) network (IP)

Transport Layer: 3-24

SNMP client SNMP server

(UDP) header fields values (UDP)

Transport Layer: 3-25

SNMP client SNMP server

Transport Layer: 3-26

application length, in bytes of

Transport Layer: 3-27

Transport Layer: 3-28

Transport Layer: 3-31

§ Connectionless transport: UDP

§ Connection-oriented transport: TCP

Transport Layer: 3-33

§ point-to-point: § cumulative ACKs

Transport Layer: 3-34

source port # dest port # segment seq #: counting

options (variable length)