Chapter 3 Transport Layer

A note on the use of these ppt slides:

Were making these slides freely available to all (faculty, students, readers). Theyre in PowerPoint form so you see the animations; and can add, modify, and delete slides (including this one) and slide content to suit your needs. They obviously represent a lot of work on our part. In return for use, we only ask the following: If you use these slides (e.g., in a class) that you mention their source (after all, wed like people to use our book!) If you post any slides on a www site, that you note that they are adapted from (or perhaps identical to) our slides, and note our copyright of this material.
Thanks and enjoy! JFK/KWR All material copyright 1996-2012 J.F Kurose and K.W. Ross, All Rights Reserved Transport Layer 3-1

Computer Networking: A Top Down Approach

6th edition Jim Kurose, Keith Ross Addison-Wesley March 2012

Chapter 3: Transport Layer

our goals:

understand principles behind transport layer services:

multiplexing, demultiplexing reliable data transfer flow control congestion control

learn about Internet transport layer protocols:

UDP: connectionless transport TCP: connection-oriented reliable transport TCP congestion control

Transport Layer 3-2

Chapter 3 outline
3.1 transport-layer services 3.2 multiplexing and demultiplexing 3.3 connectionless transport: UDP 3.4 connection-oriented transport: TCP
segment structure reliable data transfer flow control connection management

3.5 principles of congestion control 3.6 TCP congestion control

Transport Layer 3-3

Transport services and protocols

provide logical communication between app processes running on different hosts transport protocols run in end systems send side: breaks app messages into segments, passes to network layer rcv side: reassembles segments into messages, passes to app layer more than one transport protocol available to apps Internet: TCP and UDP

application transport network data link physical

application transport network data link physical

Transport Layer 3-4

Transport vs. network layer

network layer: logical communication between hosts transport layer: logical communication between processes

household analogy:
12 kids in Anns house sending letters to 12 kids in Bills house: hosts = houses processes = kids app messages = letters in envelopes transport protocol = Ann and Bill who demux to inhouse siblings network-layer protocol = postal service

relies on, enhances, network layer services

Transport Layer 3-5

Internet transport-layer protocols

reliable, in-order delivery (TCP)

congestion control flow control connection setup

application transport network data link physical network data link physical

network data link physical network data link physical network data link physical network data link physical application transport network data link physical

unreliable, unordered delivery: UDP

no-frills extension of best-effort IP

network data link physical

services not available:

delay guarantees bandwidth guarantees

network data link physical

Transport Layer 3-6

Chapter 3 outline
3.1 transport-layer services 3.2 multiplexing and demultiplexing 3.3 connectionless transport: UDP 3.4 connection-oriented transport: TCP
segment structure reliable data transfer flow control connection management

3.5 principles of congestion control 3.6 TCP congestion control

Transport Layer 3-7

Multiplexing/demultiplexing
multiplexing at sender: handle data from multiple sockets, add transport header (later used for demultiplexing)
application application

demultiplexing at receiver: use header info to deliver received segments to correct socket

P1

P2

application

P3
transport network link physical

transport network

P4
transport network link physical

socket process

link
physical

Transport Layer 3-8

How demultiplexing works

host receives IP datagrams

each datagram has source IP address, destination IP address each datagram carries one transport-layer segment each segment has source, destination port number

32 bits
source port # dest port #

other header fields

host uses IP addresses & port numbers to direct segment to appropriate socket

application data (payload)

TCP/UDP segment format

Transport Layer 3-9

Connectionless demultiplexing

recall: created socket has host-local port #:

DatagramSocket mySocket1 = new DatagramSocket(12534);

recall: when creating datagram to send into UDP socket, must specify
destination IP address destination port #
IP datagrams with same dest. port #, but different source IP addresses and/or source port numbers will be directed to same socket at dest
Transport Layer 3-10

when host receives UDP segment:

checks destination port # in segment directs UDP segment to socket with that port #

Connectionless demux: example

DatagramSocket mySocket2 = new DatagramSocket (9157);
application

DatagramSocket serverSocket = new DatagramSocket (6428);

application

DatagramSocket mySocket1 = new DatagramSocket (5775);

application

P1
transport

P3
transport network link physical source port: 6428 dest port: 9157 source port: ? dest port: ? network link physical

P4
transport network link physical

source port: 9157 dest port: 6428

source port: ? dest port: ?

Transport Layer 3-11

Connection-oriented demux

TCP socket identified by 4-tuple:

source IP address source port number dest IP address dest port number

server host may support many simultaneous TCP sockets:

each socket identified by its own 4-tuple

demux: receiver uses all four values to direct segment to appropriate socket

web servers have different sockets for each connecting client

non-persistent HTTP will have different socket for each request

Transport Layer 3-12

Connection-oriented demux: example

application application

P4

P5
transport

P6

application

P3
transport network link physical network

P2

P3

transport network link

link
physical

server: IP address B
source IP,port: B,80 dest IP,port: A,9157 source IP,port: A,9157 dest IP, port: B,80

physical

host: IP address A

source IP,port: C,5775 dest IP,port: B,80 source IP,port: C,9157 dest IP,port: B,80

host: IP address C

three segments, all destined to IP address: B, dest port: 80 are demultiplexed to different sockets

Transport Layer 3-13

Connection-oriented demux: example

threaded server
application application application

P3
transport network link physical

P4
transport network

P2

P3

transport network link

link
physical

server: IP address B
source IP,port: B,80 dest IP,port: A,9157 source IP,port: A,9157 dest IP, port: B,80

physical

host: IP address A

source IP,port: C,5775 dest IP,port: B,80 source IP,port: C,9157 dest IP,port: B,80

host: IP address C

Transport Layer 3-14

Chapter 3 outline
3.1 transport-layer services 3.2 multiplexing and demultiplexing 3.3 connectionless transport: UDP 3.4 connection-oriented transport: TCP
segment structure reliable data transfer flow control connection management

3.5 principles of congestion control 3.6 TCP congestion control

Transport Layer 3-15

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

UDP use:
streaming multimedia apps (loss tolerant, rate sensitive) DNS SNMP

reliable transfer over UDP:

add reliability at application layer application-specific error recovery!

Transport Layer 3-16

UDP: segment header

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

why is there a UDP?

application data (payload)

UDP segment format

no connection establishment (which can add delay) simple: no connection state at sender, receiver small header size no congestion control: UDP can blast away as fast as desired
Transport Layer 3-17

UDP checksum
Goal: detect errors (e.g., flipped bits) in transmitted segment

sender:

treat segment contents, including header fields, as sequence of 16-bit integers checksum: addition (ones complement sum) of segment contents sender puts checksum value into UDP checksum field

receiver:

compute checksum of received segment check if computed checksum equals checksum field value: NO - error detected YES - no error detected. But maybe errors nonetheless? More later .
Transport Layer 3-18

Internet checksum: example

example: add two 16-bit integers
1 1 1 1 0 0 1 1 0 0 1 1 0 0 1 1 0 1 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 1 0 1 1 1 0 1 1 1 0 1 1 1 1 0 0 checksum 1 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

Transport Layer 3-19

Chapter 3 outline
3.1 transport-layer services 3.2 multiplexing and demultiplexing 3.3 connectionless transport: UDP 3.4 principles of reliable data transfer 3.5 connection-oriented transport: TCP
segment structure reliable data transfer flow control connection management

3.6 principles of congestion control 3.7 TCP congestion control

Transport Layer 3-20

TCP: Overview

RFCs: 793,1122,1323, 2018, 2581

point-to-point:
one sender, one receiver

full duplex data:

bi-directional data flow in same connection MSS: maximum segment size

reliable, in-order byte steam:

no message boundaries

connection-oriented:
handshaking (exchange of control msgs) inits sender, receiver state before data exchange

pipelined:
TCP congestion and flow control set window size

flow controlled:
sender will not overwhelm receiver
Transport Layer 3-21

TCP segment structure

32 bits URG: urgent data (generally not used) ACK: ACK # valid PSH: push data now (generally not used) RST, SYN, FIN: connection estab (setup, teardown commands) Internet checksum (as in UDP)

source port #

dest port #

sequence number

acknowledgement number
head not UAP R S F len used

counting by bytes of data (not segments!) # bytes rcvr willing to accept

receive window Urg data pointer

checksum

options (variable length)

application data (variable length)

Transport Layer 3-22

TCP seq. numbers, ACKs

sequence numbers: byte stream number of first byte in segments data acknowledgements: seq # of next byte expected from other side cumulative ACK Q: how receiver handles out-of-order segments A: TCP spec doesnt say, - up to implementor
outgoing segment from sender
source port # dest port #

sequence number acknowledgement number rwnd

checksum urg pointer

window size

sender sequence number space

sent ACKed sent, not- usable not yet ACKed but not usable yet sent (inflight)
source port # dest port #

incoming segment to sender

sequence number acknowledgement number rwnd A
checksum urg pointer

Transport Layer 3-23

TCP seq. numbers, ACKs

Host A Host B

User types C

Seq=42, ACK=79, data = C

host ACKs receipt of echoed C

Seq=79, ACK=43, data = C

host ACKs receipt of C, echoes back C

Seq=43, ACK=80

simple telnet scenario

Transport Layer 3-24

TCP round trip time, timeout

Q: how to set TCP timeout value?

Q: how to estimate RTT?

longer than RTT

but RTT varies

too short: premature timeout, unnecessary retransmissions too long: slow reaction to segment loss

SampleRTT: measured time from segment transmission until ACK receipt ignore retransmissions SampleRTT will vary, want estimated RTT smoother average several recent measurements, not just current SampleRTT

Transport Layer 3-25

TCP round trip time, timeout

EstimatedRTT = (1- )*EstimatedRTT + *SampleRTT

exponential weighted moving average influence of past sample decreases exponentially fast typical value: = 0.125
RTT: gaia.cs.umass.edu to fantasia.eurecom.fr
350

RTT: gaia.cs.umass.edu to fantasia.eurecom.fr

RTT (milliseconds)

300

RTT (milliseconds)

250

200

sampleRTT
150

EstimatedRTT
1 8 15 22 29 36 43 50 57 64 71 78 85 92 99 106

100

time (seconds)
SampleRTT

time (seconnds)

Estimated RTT

Transport Layer 3-26

TCP round trip time, timeout

timeout interval: EstimatedRTT plus safety margin

large variation in EstimatedRTT -> larger safety margin

estimate SampleRTT deviation from EstimatedRTT:

DevRTT = (1-)*DevRTT + *|SampleRTT-EstimatedRTT| (typically, = 0.25)

TimeoutInterval = EstimatedRTT + 4*DevRTT

estimated RTT

safety margin

Transport Layer 3-27

Chapter 3 outline
3.1 transport-layer services 3.2 multiplexing and demultiplexing 3.3 connectionless transport: UDP 3.4 connection-oriented transport: TCP
segment structure reliable data transfer flow control connection management

3.5 principles of congestion control 3.6 TCP congestion control

Transport Layer 3-28

TCP reliable data transfer

TCP creates rdt service on top of IPs unreliable service

pipelined segments cumulative acks single retransmission timer

lets initially consider simplified TCP sender:

ignore duplicate acks ignore flow control, congestion control

retransmissions triggered by:

timeout events duplicate acks

Transport Layer 3-29

TCP sender events:

data rcvd from app: create segment with seq # seq # is byte-stream number of first data byte in segment start timer if not already running
think of timer as for oldest unacked segment expiration interval:
TimeOutInterval

timeout: retransmit segment that caused timeout restart timer ack rcvd: if ack acknowledges previously unacked segments
update what is known to be ACKed start timer if there are still unacked segments
Transport Layer 3-30

TCP sender (simplified)

data received from application above create segment, seq. #: NextSeqNum pass segment to IP (i.e., send) NextSeqNum = NextSeqNum + length(data) if (timer currently not running) start timer

L
NextSeqNum = InitialSeqNum SendBase = InitialSeqNum

wait for event

timeout
retransmit not-yet-acked segment with smallest seq. # start timer

ACK received, with ACK field value y

if (y > SendBase) { SendBase = y /* SendBase1: last cumulatively ACKed byte */ if (there are currently not-yet-acked segments) start timer else stop timer }

Transport Layer 3-31

TCP: retransmission scenarios

Host A Host B Host A Host B

SendBase=92 Seq=92, 8 bytes of data timeout ACK=100 timeout Seq=92, 8 bytes of data Seq=100, 20 bytes of data

ACK=100 ACK=120 Seq=92, 8 bytes of data SendBase=100 ACK=100 SendBase=120 ACK=120 SendBase=120 Seq=92, 8 bytes of data

lost ACK scenario

premature timeout
Transport Layer 3-32

TCP: retransmission scenarios

Host A Host B

Seq=92, 8 bytes of data Seq=100, 20 bytes of data timeout

ACK=100

ACK=120

Seq=120, 15 bytes of data

cumulative ACK
Transport Layer 3-33

TCP ACK generation

event at receiver
arrival of in-order segment with expected seq #. All data up to expected seq # already ACKed arrival of in-order segment with expected seq #. One other segment has ACK pending arrival of out-of-order segment higher-than-expect seq. # . Gap detected arrival of segment that partially or completely fills gap

[RFC 1122, RFC 2581]

TCP receiver action

delayed ACK. Wait up to 500ms for next segment. If no next segment, send ACK immediately send single cumulative ACK, ACKing both in-order segments

immediately send duplicate ACK, indicating seq. # of next expected byte

immediate send ACK, provided that segment starts at lower end of gap
Transport Layer 3-34

TCP fast retransmit

time-out period often relatively long:

long delay before resending lost packet

TCP fast retransmit

detect lost segments via duplicate ACKs.

sender often sends many segments backto-back if segment is lost, there will likely be many duplicate ACKs.

if sender receives 3 ACKs for same data

( ( triple triple duplicate duplicate ACKs ACKs ), ),

resend unacked segment with smallest seq #

likely that unacked segment lost, so dont wait for timeout

Transport Layer 3-35

TCP fast retransmit

Host A

Host B

Seq=92, 8 bytes of data Seq=100, 20 bytes of data

X
ACK=100 timeout ACK=100 ACK=100 ACK=100 Seq=100, 20 bytes of data

fast retransmit after sender receipt of triple duplicate ACK

Transport Layer 3-36

Chapter 3 outline
3.1 transport-layer services 3.2 multiplexing and demultiplexing 3.3 connectionless transport: UDP 3.4 connection-oriented transport: TCP
segment structure reliable data transfer flow control connection management

3.5 principles of congestion control 3.6 TCP congestion control

Transport Layer 3-37

TCP flow control

application may remove data from TCP socket buffers .
slower than TCP receiver is delivering (sender is sending) application process application TCP socket receiver buffers OS

TCP code

receiver controls sender, so sender wont overflow receivers buffer by transmitting too much, too fast

flow control

IP code

from sender

receiver protocol stack

Transport Layer 3-38

TCP flow control

receiver advertises free buffer space by including rwnd value in TCP header of receiver-to-sender segments
RcvBuffer size set via socket options (typical default is 4096 bytes) many operating systems autoadjust RcvBuffer

to application process

RcvBuffer rwnd

buffered data free buffer space

sender limits amount of unacked (in-flight) data to receivers rwnd value guarantees receive buffer will not overflow

TCP segment payloads

receiver-side buffering

Transport Layer 3-39

Chapter 3 outline
3.1 transport-layer services 3.2 multiplexing and demultiplexing 3.3 connectionless transport: UDP 3.4 connection-oriented transport: TCP
segment structure reliable data transfer flow control connection management

3.5 principles of congestion control 3.6 TCP congestion control

Transport Layer 3-40

Connection Management
before exchanging data, sender/receiver handshake:

agree to establish connection (each knowing the other willing to establish connection) agree on connection parameters
application
connection state: ESTAB connection variables: seq # client-to-server server-to-client rcvBuffer size at server,client

application
connection state: ESTAB connection Variables: seq # client-to-server server-to-client rcvBuffer size at server,client

network

network

Socket clientSocket = newSocket("hostname","port number");

Socket connectionSocket = welcomeSocket.accept(); Transport Layer 3-41

Agreeing to establish a connection

2-way handshake:

Lets talk ESTAB OK ESTAB

Q: will 2-way handshake always work in network?

choose x ESTAB

req_conn(x)
acc_conn(x) ESTAB

variable delays retransmitted messages (e.g. req_conn(x)) due to message loss message reordering cant see other side

Transport Layer 3-42

Agreeing to establish a connection

2-way handshake failure scenarios:
choose x

choose x retransmit req_conn(x) ESTAB

req_conn(x)

req_conn(x) ESTAB acc_conn(x) data(x+1)

ESTAB
acc_conn(x) retransmit req_conn(x) ESTAB req_conn(x)
connection x completes

retransmit data(x+1)
server forgets x ESTAB client terminates
connection x completes

accept data(x+1)

client terminates

req_conn(x) data(x+1)

server forgets x
ESTAB accept data(x+1)

half open connection! (no client!)

Transport Layer 3-43

TCP 3-way handshake

client state
LISTEN
choose init seq num, x send TCP SYN msg

server state
LISTEN

SYNSENT

SYNbit=1, Seq=x

choose init seq num, y send TCP SYNACK SYN RCVD msg, acking SYN

received SYNACK(x) indicates server is live; ESTAB send ACK for SYNACK; this segment may contain client-to-server data

SYNbit=1, Seq=y ACKbit=1; ACKnum=x+1

ACKbit=1, ACKnum=y+1
received ACK(y) indicates client is live

ESTAB

Transport Layer 3-44

TCP 3-way handshake: FSM

closed
Socket connectionSocket = welcomeSocket.accept();

L
SYN(x)
SYNACK(seq=y,ACKnum=x+1) create new socket for communication back to client

Socket clientSocket = newSocket("hostname","port number");

listen

SYN(seq=x)

SYN rcvd ESTAB

SYN sent
SYNACK(seq=y,ACKnum=x+1)

ACK(ACKnum=y+1)

ACK(ACKnum=y+1)

L
Transport Layer 3-45

TCP: closing a 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

Transport Layer 3-46

TCP: closing a connection

client state
ESTAB
clientSocket.close()

server state
ESTAB
can no longer send but can receive data wait for server close

FIN_WAIT_1

FINbit=1, seq=x CLOSE_WAIT ACKbit=1; ACKnum=x+1

can still send data

FIN_WAIT_2

FINbit=1, seq=y TIMED_WAIT

timed wait for 2*max segment lifetime

LAST_ACK
can no longer send data

ACKbit=1; ACKnum=y+1 CLOSED

CLOSED
Transport Layer 3-47

Chapter 3 outline
3.1 transport-layer services 3.2 multiplexing and demultiplexing 3.3 connectionless transport: UDP 3.4 connection-oriented transport: TCP
segment structure reliable data transfer flow control connection management

3.5 principles of congestion control 3.6 TCP congestion control

Transport Layer 3-48

Principles of congestion control

congestion:

informally: too many sources sending too much data too fast for network to handle different from flow control! manifestations: lost packets (buffer overflow at routers) long delays (queueing in router buffers) a top-10 problem!

Transport Layer 3-49

Causes/costs of congestion: scenario 1

two senders, two receivers one router, infinite buffers output link capacity: R no retransmission

original data: lin

Host A

throughput:

lout

unlimited shared output link buffers

Host B

R/2

lin R/2 maximum per-connection throughput: R/2

lin R/2 large delays as arrival rate, lin, approaches capacity

Transport Layer 3-50

delay

lout

Causes/costs of congestion: scenario 2

one router, finite buffers sender retransmission of timed-out packet

application-layer input = application-layer output: lin = lout transport-layer input includes retransmissions : l lin in
lin : original data l'in: original data, plus
retransmitted data Host A

lout

Host B

finite shared output link buffers

Transport Layer 3-51

Causes/costs of congestion: scenario 2

idealization: perfect knowledge sender sends only when router buffers available
R/2

lout
lin

R/2

copy

lin : original data l'in: original data, plus

retransmitted data A

lout

free buffer space!

Host B

finite shared output link buffers

Transport Layer 3-52

Causes/costs of congestion: scenario 2

Idealization: known loss
packets can be lost, dropped at router due to full buffers sender only resends if packet known to be lost
copy

lin : original data l'in: original data, plus

retransmitted data A

lout

no buffer space!

Host B
Transport Layer 3-53

Causes/costs of congestion: scenario 2

Idealization: known loss
packets can be lost, dropped at router due to full buffers sender only resends if packet known to be lost
R/2 when sending at R/2, some packets are retransmissions but asymptotic goodput is still R/2 (why?)

lout

lin

R/2

lin : original data l'in: original data, plus

retransmitted data A

lout

free buffer space!

Host B
Transport Layer 3-54

Causes/costs of congestion: scenario 2

Realistic: duplicates

packets can be lost, dropped at router due to full buffers sender times out prematurely, sending two copies, both of which are delivered
timeout copy

R/2 when sending at R/2, some packets are retransmissions including duplicated that are delivered!

lout

lin

R/2

lin l'in
A

lout

free buffer space!

Host B
Transport Layer 3-55

Causes/costs of congestion: scenario 2

Realistic: duplicates

packets can be lost, dropped at router due to full buffers sender times out prematurely, sending two copies, both of which are delivered

R/2 when sending at R/2, some packets are retransmissions including duplicated that are delivered!

lout

lin

R/2

costs of congestion:

more work (retrans) for given goodput unneeded retransmissions: link carries multiple copies of pkt decreasing goodput

Transport Layer 3-56

Causes/costs of congestion: scenario 3

four senders multihop paths timeout/retransmit

Host A

Q: what happens as lin and lin

increase ? A: as red lin increases, all arriving blue pkts at upper queue are dropped, blue throughput g 0
lout
Host B

lin : original data l'in: original data, plus

retransmitted data finite shared output link buffers

Host D Host C

Transport Layer 3-57

Causes/costs of congestion: scenario 3

C/2

lout

lin

C/2

another cost of congestion: when packet dropped, any upstream transmission capacity used for that packet was wasted!
Transport Layer 3-58

Approaches towards congestion control

two broad approaches towards congestion control:

end-end congestion control:

network-assisted congestion control:

no explicit feedback from network congestion inferred from end-system observed loss, delay approach taken by TCP

routers provide feedback to end systems single bit indicating congestion (SNA, DECbit, TCP/IP ECN, ATM) explicit rate for sender to send at
Transport Layer 3-59

Chapter 3 outline
3.1 transport-layer services 3.2 multiplexing and demultiplexing 3.3 connectionless transport: UDP 3.4 connection-oriented transport: TCP
segment structure reliable data transfer flow control connection management

3.5 principles of congestion control 3.6 TCP congestion control

Transport Layer 3-60

TCP congestion control: additive increase

multiplicative decrease

approach: sender increases transmission rate (window size), probing for usable bandwidth, until loss occurs additive increase: increase cwnd by 1 MSS every RTT until loss detected multiplicative decrease: cut cwnd in half after loss
cwnd: TCP sender congestion window size

additively increase window size . until loss occurs (then cut window in half)

AIMD saw tooth behavior: probing for bandwidth

time
Transport Layer 3-61

TCP Congestion Control: details

sender sequence number space
cwnd

last byte ACKed

sender limits transmission:

LastByteSent< cwnd LastByteAcked

sent, notyet ACKed (inflight)

last byte sent

TCP sending rate: roughly: send cwnd bytes, wait RTT for ACKS, then send more bytes
rate
~ ~

cwnd
RTT

bytes/sec

cwnd is dynamic, function of perceived network congestion

Transport Layer 3-62

TCP Slow Start

when connection begins, increase rate exponentially until first loss event:
initially cwnd = 1 MSS double cwnd every RTT done by incrementing cwnd for every ACK received

Host A

Host B

summary: initial rate is slow but ramps up exponentially fast

RTT

time

Transport Layer 3-63

TCP: detecting, reacting to loss

loss indicated by timeout:

cwnd set to 1 MSS; window then grows exponentially (as in slow start) to threshold, then grows linearly loss indicated by 3 duplicate ACKs: TCP RENO dup ACKs indicate network capable of delivering some segments cwnd is cut in half window then grows linearly TCP Tahoe always sets cwnd to 1 (timeout or 3

duplicate acks)

Transport Layer 3-64

TCP: switching from slow start to CA

Q: when should the exponential increase switch to linear? A: when cwnd gets to 1/2 of its value before timeout.

Implementation:

variable ssthresh on loss event, ssthresh is set to 1/2 of cwnd just before loss event
Transport Layer 3-65

Summary: TCP Congestion Control

duplicate ACK dupACKcount++ L cwnd = 1 MSS ssthresh = 64 KB dupACKcount = 0

New ACK!
new ACK cwnd = cwnd+MSS dupACKcount = 0 transmit new segment(s), as allowed cwnd > ssthresh L timeout ssthresh = cwnd/2 cwnd = 1 MSS dupACKcount = 0 retransmit missing segment

new ACK cwnd = cwnd + MSS (MSS/cwnd) dupACKcount = 0 transmit new segment(s), as allowed

New ACK!

slow start

congestion avoidance
duplicate ACK dupACKcount++

timeout ssthresh = cwnd/2 cwnd = 1 MSS dupACKcount = 0 retransmit missing segment

dupACKcount == 3 ssthresh= cwnd/2 cwnd = ssthresh + 3 retransmit missing segment

timeout ssthresh = cwnd/2 cwnd = 1 dupACKcount = 0 retransmit missing segment

New ACK!
New ACK cwnd = ssthresh dupACKcount = 0 dupACKcount == 3 ssthresh= cwnd/2 cwnd = ssthresh + 3 retransmit missing segment

fast recovery
duplicate ACK

cwnd = cwnd + MSS transmit new segment(s), as allowed

Transport Layer 3-66

TCP throughput

avg. TCP thruput as function of window size, RTT?

ignore slow start, assume always data to send

W: window size (measured in bytes) where loss occurs

avg. window size (# in-flight bytes) is W avg. thruput is 3/4W per RTT
avg TCP thruput =
W

3 W bytes/sec 4 RTT

W/2

Transport Layer 3-67

TCP Futures: TCP over long, fat pipes

example: 1500 byte segments, 100ms RTT, want 10 Gbps throughput requires W = 83,333 in-flight segments throughput in terms of segment loss probability, L
[Mathis 1997]:

. MSS 1.22 TCP throughput = RTT L to achieve 10 Gbps throughput, need a loss rate of L = 210-10 a very small loss rate!

new versions of TCP for high-speed

Transport Layer 3-68

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

TCP connection 2

bottleneck router capacity R

Transport Layer 3-69

Why is TCP fair?

two competing sessions:

additive increase gives slope of 1, as throughout increases multiplicative decrease decreases throughput proportionally
R
equal bandwidth share

loss: decrease window by factor of 2 congestion avoidance: additive increase loss: decrease window by factor of 2 congestion avoidance: additive increase

Connection 1 throughput

R
Transport Layer 3-70

Fairness (more)
Fairness and UDP multimedia apps often do not use TCP Fairness, parallel TCP connections application can open do not want rate multiple parallel throttled by congestion connections between two control hosts instead use UDP: web browsers do this send audio/video at e.g., link of rate R with 9 constant rate, tolerate packet loss existing connections:
new app asks for 1 TCP, gets rate R/10 new app asks for 11 TCPs, gets R/2
Transport Layer 3-71

Chapter 3: summary

principles behind transport layer services: multiplexing, demultiplexing reliable data transfer flow control congestion control instantiation, implementation in the Internet
UDP TCP

next: leaving the network edge (application, transport layers) into the network core

Transport Layer 3-72