3rd Edition: Chapter 3

Chapter 3
Transport Layer
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Computer Networking:
A Top Down Approach
Featuring the Internet,
3rd edition.
Jim Kurose, Keith Ross
Addison-Wesley, July
2004.
Thanks and enjoy! JFK/KWR
All material copyright 1996-2004
J.F Kurose and K.W. Ross, All Rights Reserved
Transport Layer
3-1
邦訳版
インターネット技術のすべて：ト
ップダウンアプローチによる実
践ネットワーク技法第２版
ジェームズ・F・クロセ (著), キ
ース・W・ロス (著), 岡田博美
(翻訳)
出版社: ピアソン・エデュケーシ
ョン (2003/12/25)
ASIN: 4894714949
Transport Layer
3-2
トランスポート層
Chapter 3: Transport Layer
Our goals: 目標
 understand principles
behind transport
layer services:
トランスポート層サービスの
背後にある原理の理解:
 multiplexing/demultipl
exing
 reliable data transfer
 flow control
 congestion control
 learn about transport
layer protocols in the
Internet:
インターネットにおけるトランスポー
ト層について学習:
 UDP: connectionless
transport
 TCP: connection-oriented
transport
 TCP congestion control
Transport Layer
3-3
Chapter 3 outline
 3.1 Transport-layer
services
トランスポート層サービス
 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
3-4
Transport services and protocols
トランスポートサービスとプロトコル
logical communication
（論理的な通信） between app
 provide
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
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
Transport Layer
3-5
Transport vs. network layer
トランスポート層とネットワーク層

network layer: logical
Household analogy:
communication
between hosts
12 kids sending letters to
12 kids
ネットワーク層: ホスト間の論
 processes = kids
理的通信

transport layer: logical
communication
between processes
トランスポート層: プロセス間
の論理的通信
 relies on, enhances,
network layer services
 app messages = letters
in envelopes
 hosts = houses
 transport protocol =
Ann and Bill
 network-layer protocol
= postal service
Transport Layer
3-6
Internet transport-layer protocols
インターネットトランスポート層プロトコル
 reliable, in-order
delivery (TCP)
高信頼，順序保証配送: TCP



congestion control
flow control
connection setup
 unreliable, unordered
delivery: UDP
低信頼，順序非保証配送: 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
network
data link
physical
network
data link
physical
network
data link
physical
application
transport
network
data link
physical
Transport Layer
3-7
Chapter 3 outline
 3.1 Transport-layer
services
 3.2 Multiplexing and
demultiplexing
多重化と逆多重化
 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
3-8
Multiplexing/demultiplexing
多重化／逆多重化
始点ホストにおける多重化:
終点ホストにおける逆多重化:
Multiplexing at send host:
gathering data from multiple
sockets, enveloping data with
header (later used for
demultiplexing)
Demultiplexing at rcv host:
delivering received segments
to correct socket
= socket
application
transport
network
link
= process
P3
P1
P1
application
transport
network
P2
P4
application
transport
network
link
link
physical
host 1
physical
host 2
physical
host 3
Transport Layer
3-9
How demultiplexing works
逆多重化はどのように機能するか
 host receives IP datagrams
ホストはIPデータグラムを受信
 each datagram has source
IP address, destination IP
address
 each datagram carries 1
transport-layer segment
 each segment has source,
destination port number
 host uses IP addresses & port
numbers to direct segment to
appropriate socket
ホストは，適切なソケットにセグメ
ントを向けるためにＩＰアドレスとポ
ート番号を使う
32 bits
source port #
dest port #
other header fields
application
data
(message)
TCP/UDP segment format
Transport Layer 3-10
Connectionless demultiplexing
コネクションレス型の逆多重化
 Create sockets with port
numbers:
ポート番号を付与してソケットを生成：
DatagramSocket mySocket1 = new
DatagramSocket(99111);
DatagramSocket mySocket2 = new
DatagramSocket(99222);
 UDP socket identified by
two-tuple:
(dest IP address, dest port number)
UDP ソケットは
（終点ＩＰアドレス，終点ポート番号）
で識別される
 When host receives UDP
segment:
ホストがUDPセグメントを受信す
ると：


checks destination port
number in segment
directs UDP segment to
socket with that port number
 IP datagrams with different
source IP addresses and/or
source port numbers directed
to same socket
同一ソケットにむけられた異なる
始点ＩＰアドレスあるいはまた異な
る始点ポート番号をもつIPデータ
グラム
Transport Layer
3-11
Connectionless demux (cont)
コネクションレス型の逆多重化（続き）
DatagramSocket serverSocket = new DatagramSocket(6428);
P2
SP: 6428
SP: 6428
DP: 9157
DP: 5775
SP: 9157
client
IP: A
P1
P1
P3
DP: 6428
SP: 5775
server
IP: C
DP: 6428
Client
IP:B
SP provides “return address”
Transport Layer 3-12
Connection-oriented demux
コネクション指向型の逆多重化
 TCP socket identified
by 4-tuple:
TCP ソケットは次の４つの値に
よって識別される:
 source IP address
 source port number
 dest IP address
 dest port number
 recv host uses all four
values to direct
segment to appropriate
socket
終点ホストは，4つの値を使って
セグメントをアプリケーションソケ
ットに向ける
 Server host may support
many simultaneous TCP
sockets:
サーバホストは同時に多数の
TCPソケットをサポートするかもし
れない：
 each socket identified by
its own 4-tuple
 Web servers have
different sockets for
each connecting client
Ｗｅｂサーバは各クライアントに対
して異なるソケットを持つ
 non-persistent HTTP will
have different socket for
each requestTransport Layer 3-13
Connection-oriented demux (cont)
コネクション指向型の逆多重化（続き）
P1
P4
P5
P2
P6
P1P3
SP: 5775
DP: 80
S-IP: B
D-IP:C
SP: 9157
client
IP: A
DP: 80
S-IP: A
D-IP:C
SP: 9157
server
IP: C
DP: 80
S-IP: B
D-IP:C
Client
IP:B
Transport Layer 3-14
Connection-oriented demux:
Threaded Web Server
コネクション指向型の逆多重化：
スレッド化されたWebサーバ
P1
P2
P4
P1P3
SP: 5775
DP: 80
S-IP: B
D-IP:C
SP: 9157
client
IP: A
DP: 80
S-IP: A
D-IP:C
SP: 9157
server
IP: C
DP: 80
S-IP: B
D-IP:C
Client
IP:B
Transport Layer 3-15
Chapter 3 outline
 3.1 Transport-layer
services
 3.2 Multiplexing and
demultiplexing
 3.3 Connectionless
transport: UDP
コネクションレス型トラン
スポート: 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 3-16
UDP: User Datagram Protocol [RFC 768]
 “no frills,” “bare bones”
Internet transport protocol
“余計な機能のない,” “余計な部分を
削った” インターネットトランスポートプ
ロトコル
 “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
Transport Layer 3-17
UDP: more
 often used for streaming
32 bits
multimedia apps
ストリーミングマルチメディアアプリ
Length, in
ケーションによく利用される
bytes of UDP
 loss tolerant
segment,
 rate sensitive
including
 other UDP uses
header
DNS
 SNMP
 reliable transfer over UDP:
add reliability at
application layer
 application-specific
error recovery!
source port #
dest port #
length
checksum

Application
data
(message)
UDP segment format
Transport Layer 3-18
UDP checksum UDPチェックサム
Goal: detect “errors” (e.g., flipped bits) in transmitted
segment
目的: 転送セグメント中の“誤り”を検出する（例えば，ビット反転など）
Sender: 始点ホスト
 treat segment contents
as sequence of 16-bit
integers
 checksum: addition (1’s
complement sum) of
segment contents
 sender puts checksum
value into UDP checksum
field
Receiver: 終点ホスト
 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 3-19
Internet Checksum Example
インターネットチェックサムの例
 Note 注意

When adding numbers, a carryout from the
most significant bit needs to be added to the
result
加算結果の最上位のキャリーアウトは結果に加算されなければ
ならない
 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
Transport Layer 3-20
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
高信頼データ転送の原理
 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 3-21
Principles of Reliable data transfer
高信頼データ転送の原理
 important in app., transport, link layers
 top-10 list of important networking topics!
 characteristics of unreliable channel will determine
complexity of reliable data transfer protocol (rdt)
Transport Layer 3-22
Reliable data transfer: getting started
高信頼データ転送：はじめに
rdt_send(): called from above,
(e.g., by app.). Passed data to
deliver to receiver upper layer
send
side
udt_send(): called by rdt,
to transfer packet over
unreliable channel to receiver
deliver_data(): called by
rdt to deliver data to upper
receive
side
rdt_rcv(): called when packet
arrives on rcv-side of channel
Transport Layer 3-23
Reliable data transfer: getting started
高信頼データ転送：はじめに
We’ll:以後
 incrementally develop sender, receiver sides of
reliable data transfer protocol (rdt)
高信頼転送プロトコル(rdt)の始点ホスト，終点ホストを順に発
展させる
 consider only unidirectional data transfer
 but control info will flow on both directions!
 use finite state machines (FSM) to specify
sender, receiver
state: when in this
“state” next state
uniquely determined
by next event
state
1
event causing state transition
actions taken on state transition
event
actions
state
2
Transport Layer 3-24
Rdt1.0: reliable transfer over a reliable channel
高信頼チャネルを介した高信頼転送
 underlying channel perfectly reliable
下層チャネルは完全に信頼できる
 no bit errors
 no loss of packets
 separate FSMs for sender, receiver:
始点ホストと終点ホストは分離:
 sender sends data into underlying channel
 receiver read data from underlying channel
Wait for
call from
above
rdt_send(data)
packet = make_pkt(data)
udt_send(packet)
sender
Wait for
call from
below
rdt_rcv(packet)
extract (packet,data)
deliver_data(data)
receiver
Transport Layer 3-25
Rdt2.0: channel with bit errors
ビットエラーのあるチャンネル
 underlying channel may flip bits in packet
下層チャネルでパケット内のビット反転が発生しうる
 checksum to detect bit errors

the question: how to recover from errors:
誤りからどのように回復するか:

acknowledgements (ACKs): receiver explicitly tells sender

negative acknowledgements (NAKs): receiver explicitly

that pkt received OK
tells sender that pkt had errors
sender retransmits pkt on receipt of NAK
 new mechanisms in rdt2.0 (beyond rdt1.0):
 error detection
 receiver feedback: control msgs (ACK,NAK) rcvr->sender
Transport Layer 3-26
rdt2.0: FSM specification FSMの記述
rdt_send(data)
snkpkt = make_pkt(data, checksum)
udt_send(sndpkt)
rdt_rcv(rcvpkt) &&
isNAK(rcvpkt)
Wait for
Wait for
call from
ACK or
udt_send(sndpkt)
above
NAK
rdt_rcv(rcvpkt) && isACK(rcvpkt)
L
sender
receiver
rdt_rcv(rcvpkt) &&
corrupt(rcvpkt)
udt_send(NAK)
Wait for
call from
below
rdt_rcv(rcvpkt) &&
notcorrupt(rcvpkt)
extract(rcvpkt,data)
deliver_data(data)
udt_send(ACK)
Transport Layer 3-27
rdt2.0: operation with no errors
rdt2.0: 誤りがない場合
rdt_send(data)
snkpkt = make_pkt(data, checksum)
udt_send(sndpkt)
rdt_rcv(rcvpkt) &&
isNAK(rcvpkt)
Wait for
Wait for
call from
ACK or
udt_send(sndpkt)
above
NAK
rdt_rcv(rcvpkt) && isACK(rcvpkt)
L
rdt_rcv(rcvpkt) &&
corrupt(rcvpkt)
udt_send(NAK)
Wait for
call from
below
rdt_rcv(rcvpkt) &&
notcorrupt(rcvpkt)
extract(rcvpkt,data)
deliver_data(data)
udt_send(ACK)
Transport Layer 3-28
rdt2.0: error scenario
rdt_send(data)
snkpkt = make_pkt(data, checksum)
udt_send(sndpkt)
rdt_rcv(rcvpkt) &&
isNAK(rcvpkt)
Wait for
Wait for
call from
ACK or
udt_send(sndpkt)
above
NAK
rdt_rcv(rcvpkt) && isACK(rcvpkt)
L
エラーがある場合
rdt_rcv(rcvpkt) &&
corrupt(rcvpkt)
udt_send(NAK)
Wait for
call from
below
rdt_rcv(rcvpkt) &&
notcorrupt(rcvpkt)
extract(rcvpkt,data)
deliver_data(data)
udt_send(ACK)
Transport Layer 3-29
rdt2.0 has a fatal flaw!
rdt2.0 の致命的欠陥！
What happens if
ACK/NAK corrupted?
ACK/NAK が壊れたらどうする？
 sender doesn’t know what
happened at receiver!
 can’t just retransmit:
possible duplicate
Handling duplicates:
重複の取り扱い:
 sender retransmits current
pkt if ACK/NAK garbled
 sender adds sequence
number to each pkt
 receiver discards (doesn’t
deliver up) duplicate pkt
stop and wait
Sender sends one packet,
then waits for receiver
response
Transport Layer 3-30
rdt2.1: sender, handles garbled ACK/NAKs
rdt2.1: 始点ホストでの ACK/NAKs 誤りの扱い
rdt_send(data)
sndpkt = make_pkt(0, data, checksum)
udt_send(sndpkt)
rdt_rcv(rcvpkt) &&
( corrupt(rcvpkt) ||
Wait
for
Wait for
isNAK(rcvpkt) )
ACK or
call 0 from
udt_send(sndpkt)
NAK 0
above
rdt_rcv(rcvpkt)
&& notcorrupt(rcvpkt)
&& isACK(rcvpkt)
rdt_rcv(rcvpkt)
&& notcorrupt(rcvpkt)
&& isACK(rcvpkt)
L
rdt_rcv(rcvpkt) &&
( corrupt(rcvpkt) ||
isNAK(rcvpkt) )
udt_send(sndpkt)
L
Wait for
ACK or
NAK 1
Wait for
call 1 from
above
rdt_send(data)
sndpkt = make_pkt(1, data, checksum)
udt_send(sndpkt)
Transport Layer 3-31
rdt2.1: receiver, handles garbled ACK/NAKs
rdt2.1: 始点ホストでの ACK/NAKs 誤りの扱い
rdt_rcv(rcvpkt) && notcorrupt(rcvpkt)
&& has_seq0(rcvpkt)
rdt_rcv(rcvpkt) && (corrupt(rcvpkt)
extract(rcvpkt,data)
deliver_data(data)
sndpkt = make_pkt(ACK, chksum)
udt_send(sndpkt)
rdt_rcv(rcvpkt) && (corrupt(rcvpkt)
sndpkt = make_pkt(NAK, chksum)
udt_send(sndpkt)
rdt_rcv(rcvpkt) &&
not corrupt(rcvpkt) &&
has_seq1(rcvpkt)
sndpkt = make_pkt(ACK, chksum)
udt_send(sndpkt)
sndpkt = make_pkt(NAK, chksum)
udt_send(sndpkt)
Wait for
0 from
below
Wait for
1 from
below
rdt_rcv(rcvpkt) && notcorrupt(rcvpkt)
&& has_seq1(rcvpkt)
rdt_rcv(rcvpkt) &&
not corrupt(rcvpkt) &&
has_seq0(rcvpkt)
sndpkt = make_pkt(ACK, chksum)
udt_send(sndpkt)
extract(rcvpkt,data)
deliver_data(data)
sndpkt = make_pkt(ACK, chksum)
udt_send(sndpkt)
Transport Layer 3-32
rdt2.1: discussion
Sender: 始点ホスト
 seq # added to pkt
パケットにシーケンス番号を付与
 two seq. #’s (0,1) will
suffice. Why?
二つのシーケンス番号(0,1) で十
分．なぜか？
 must check if received
ACK/NAK corrupted
 twice as many states

state must “remember”
whether “current” pkt
has 0 or 1 seq. #
Receiver: 終点ホスト
 must check if received
packet is duplicate

state indicates whether
0 or 1 is expected pkt
seq #
 note: receiver can
not
know if its last
ACK/NAK received OK
at sender
Transport Layer 3-33
rdt2.2: a NAK-free protocol
rdt2.2: NAK フリープロトコル
 same functionality as rdt2.1, using ACKs only
ACKのみを使って rdt2.1 と同一機能を実現
 instead of NAK, receiver sends ACK for last pkt
received OK
NAKの代わりに，終点ホストは最後に正しく受信されたパケットに対す
る ACK を送信
 receiver must explicitly include seq # of pkt being ACKed
 duplicate ACK at sender results in same action as
NAK: retransmit current pkt
始点ホストは，重複ACKに対して NAK と同様に対処する：パケットの
再送
Transport Layer 3-34
rdt2.2: sender, receiver fragments
rdt2.2: 始点ホスト，終点ホストの FSM（一部）
rdt_send(data)
sndpkt = make_pkt(0, data, checksum)
udt_send(sndpkt)
rdt_rcv(rcvpkt) &&
( corrupt(rcvpkt) ||
Wait for
Wait for
isACK(rcvpkt,1) )
ACK
call 0 from
0
udt_send(sndpkt)
above
sender FSM
fragment
rdt_rcv(rcvpkt) &&
(corrupt(rcvpkt) ||
has_seq1(rcvpkt))
udt_send(sndpkt)
Wait for
0 from
below
rdt_rcv(rcvpkt)
&& notcorrupt(rcvpkt)
&& isACK(rcvpkt,0)
receiver FSM
fragment
L
rdt_rcv(rcvpkt) && notcorrupt(rcvpkt)
&& has_seq1(rcvpkt)
extract(rcvpkt,data)
deliver_data(data)
sndpkt = make_pkt(ACK1, chksum)
udt_send(sndpkt)
Transport Layer 3-35
rdt3.0: channels with errors and loss
rdt3.0: 誤りとロスがあるチャネル
New assumption:
新しい仮定：
underlying channel can
also lose packets (data
or ACKs)
下層チャネルはパケット(data,
ACK)を損失する可能性がある
 checksum, seq. #, ACKs,
retransmissions will be
of help, but not enough
Approach: sender waits
“reasonable” amount of
time for ACK
アプローチ: 始点ホストは，“適切な”
時間 ACK を待つ
 retransmits if no ACK
received in this time
 if pkt (or ACK) just delayed
(not lost):
 retransmission will be
duplicate, but use of seq.
#’s already handles this
 receiver must specify seq
# of pkt being ACKed
 requires countdown timer
Transport Layer 3-36
rdt3.0 sender 始点ホスト
rdt_send(data)
sndpkt = make_pkt(0, data, checksum)
udt_send(sndpkt)
start_timer
rdt_rcv(rcvpkt)
L
rdt_rcv(rcvpkt)
&& notcorrupt(rcvpkt)
&& isACK(rcvpkt,1)
rdt_rcv(rcvpkt) &&
( corrupt(rcvpkt) ||
isACK(rcvpkt,0) )
timeout
udt_send(sndpkt)
start_timer
rdt_rcv(rcvpkt)
&& notcorrupt(rcvpkt)
&& isACK(rcvpkt,0)
stop_timer
stop_timer
timeout
udt_send(sndpkt)
start_timer
L
Wait
for
ACK0
Wait for
call 0from
above
L
rdt_rcv(rcvpkt) &&
( corrupt(rcvpkt) ||
isACK(rcvpkt,1) )
Wait
for
ACK1
Wait for
call 1 from
above
rdt_send(data)
rdt_rcv(rcvpkt)
L
sndpkt = make_pkt(1, data, checksum)
udt_send(sndpkt)
start_timer
Transport Layer 3-37
rdt3.0 in action 動作
Transport Layer 3-38
rdt3.0 in action 動作
Transport Layer 3-39
Performance of rdt3.0
rdt3.0 の性能
 rdt3.0 works, but performance stinks
rdt3.0 は動作するが，性能は悪い
 example: 1 Gbps link, 15 ms e-e prop. delay, 1KB packet:
Ttransmit =
U



L (packet length in bits)
8kb/pkt
=
= 8 microsec
R (transmission rate, bps)
10**9 b/sec
sender
=
L/R
RTT + L / R
=
.008
30.008
= 0.00027
microsec
onds
U sender: utilization – fraction of time sender busy sending
1KB pkt every 30 msec -> 33kB/sec thruput over 1 Gbps link
network protocol limits use of physical resources!
Transport Layer 3-40
rdt3.0: stop-and-wait operation
rdt3.0: stop-and-wait の動作
sender
receiver
first packet bit transmitted, t = 0
last packet bit transmitted, t = L / R
first packet bit arrives
last packet bit arrives, send ACK
RTT
ACK arrives, send next
packet, t = RTT + L / R
U
sender
=
L/R
RTT + L / R
=
.008
30.008
= 0.00027
microsec
onds
Transport Layer 3-41
Pipelined protocols パイプラインプロトコル
Pipelining: sender allows multiple, “in-flight”, yet-tobe-acknowledged pkts
パイプライン: 始点ホストは，ACK を待つことなく複数のパケットを送
信できる
 range of sequence numbers must be increased
 buffering at sender and/or receiver
 Two generic forms of pipelined protocols:
selective repeat
go-Back-N,
Transport Layer 3-42
Pipelining: increased utilization
パイプライン: 利用率の改善
sender
receiver
first packet bit transmitted, t = 0
last bit transmitted, t = L / R
first packet bit arrives
last packet bit arrives, send ACK
last bit of 2nd packet arrives, send ACK
last bit of 3rd packet arrives, send ACK
RTT
ACK arrives, send next
packet, t = RTT + L / R
Increase utilization
by a factor of 3!
U
sender
=
3*L/R
RTT + L / R
=
.024
30.008
= 0.0008
microsecon
ds
Transport Layer 3-43
Go-Back-N
Sender: 始点ホスト
 k-bit seq # in pkt header パケットヘッダ内に k ビットのシーケンス番号
 “window” of up to N, consecutive unack’ed pkts allowed
 ACK(n): ACKs all pkts up to, including seq # n - “cumulative ACK”
may deceive duplicate ACKs (see receiver)
 timer for each in-flight pkt
 timeout(n): retransmit pkt n and all higher seq # pkts in window

Transport Layer 3-44
GBN: sender extended FSM
GBN: 始点ホストの拡張
rdt_send(data)
L
base=1
nextseqnum=1
if (nextseqnum < base+N) {
sndpkt[nextseqnum] = make_pkt(nextseqnum,data,chksum)
udt_send(sndpkt[nextseqnum])
if (base == nextseqnum)
start_timer
nextseqnum++
}
else
refuse_data(data)
Wait
rdt_rcv(rcvpkt)
&& corrupt(rcvpkt)
timeout
start_timer
udt_send(sndpkt[base])
udt_send(sndpkt[base+1])
…
udt_send(sndpkt[nextseqnum-1])
rdt_rcv(rcvpkt) &&
notcorrupt(rcvpkt)
base = getacknum(rcvpkt)+1
If (base == nextseqnum)
stop_timer
else
start_timer
Transport Layer 3-45
GBN: receiver extended FSM
GBN: 終点ホストの拡張 FSM
default
udt_send(sndpkt)
L
Wait
expectedseqnum=1
sndpkt =
make_pkt(expectedseqnum,ACK,chksum)
rdt_rcv(rcvpkt)
&& notcurrupt(rcvpkt)
&& hasseqnum(rcvpkt,expectedseqnum)
extract(rcvpkt,data)
deliver_data(data)
sndpkt = make_pkt(expectedseqnum,ACK,chksum)
udt_send(sndpkt)
expectedseqnum++
ACK-only: always send ACK for correctly-received pkt
with highest in-order seq #
ACKのみ: 正しく受信したパケットに対して，順序どおりに正しく受信された最大シー
ケンス番号に対してACK を送信


may generate duplicate ACKs
need only remember expectedseqnum
 out-of-order pkt:
 discard (don’t buffer) -> no receiver buffering!
 Re-ACK pkt with highest in-order seq #
Transport Layer
3-46
GBN in
action
動作
Transport Layer 3-47
Selective Repeat 選択的再送
 receiver
individually acknowledges all correctly
received pkts
終点ホストは，正しく受信された個々のパケットに ACK を送信する
 buffers pkts, as needed, for eventual in-order delivery
to upper layer
 sender only resends pkts for which ACK not
received
始点ホストは，ACK 未受信のパケットのみを再送
 sender timer for each unACKed pkt
 sender window
 N consecutive seq #’s
 again limits seq #s of sent, unACKed pkts
Transport Layer 3-48
Selective repeat: sender, receiver windows
選択的再送: 始点ホスト，終点ホストのウインドウ
Transport Layer 3-49
Selective repeat 選択的再送
sender
data from above :
receiver
pkt n in [rcvbase, rcvbase+N-1]
上位層からのデータ :
 if next available seq # in
window, send pkt
 out-of-order: buffer
timeout(n):
 resend pkt n, restart timer
ACK(n) in [sendbase,sendbase+N]:
 mark pkt n as received
 if n smallest unACKed pkt,
advance window base to
next unACKed seq #
 send ACK(n)
 in-order: deliver (also
deliver buffered, in-order
pkts), advance window to
next not-yet-received pkt
pkt n in
[rcvbase-N,rcvbase-1]
 ACK(n)
otherwise:
 ignore
Transport Layer 3-50
Selective repeat in action 選択的再送：動作
Transport Layer 3-51
Selective repeat:
dilemma ジレンマ
Example:
 seq #’s: 0, 1, 2, 3
 window size=3
 receiver sees no
difference in two
scenarios!
 incorrectly passes
duplicate data as new
in (a)
Q: what relationship
between seq # size
and window size?
Transport Layer 3-52
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
コネクション指向型トランス
ポート: TCP




segment structure
reliable data transfer
flow control
connection management
 3.6 Principles of
congestion control
 3.7 TCP congestion
control
Transport Layer 3-53
TCP: Overview
 point-to-point:
 one sender, one receiver
 reliable, in-order
steam:
byte
高信頼，順序保証，バイトストリーム

no “message boundaries”
 pipelined:
 TCP congestion and flow
control set window size

send & receive buffers
送信＆受信バッファ
socket
door
application
writes data
application
reads data
TCP
send buffer
TCP
receive buffer
segment
RFCs: 793, 1122, 1323, 2018, 2581
 full duplex data:
全二重データ:
 bi-directional data flow
in same connection
 MSS: maximum segment
size
 connection-oriented:
コネクション指向型:
 handshaking (exchange
of control msgs) init’s
sender, receiver state
before data exchange
 flow controlled:
フロー制御:
socket
 sender will not
door
overwhelm receiver
Transport Layer 3-54
TCP segment structure
TCP セグメント構造
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)
32 bits
source port #
dest port #
sequence number
acknowledgement number
head not
UA P R S F
len used
checksum
Receive window
Urg data pnter
Options (variable length)
counting
by bytes
of data
(not segments!)
# bytes
rcvr willing
to accept
application
data
(variable length)
Transport Layer 3-55
TCP seq. #’s and ACKs
TCP シークエンス番号とACK
Seq. #’s: シークエンス番号
 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
User
types
‘C’
Host B
host ACKs
receipt of
‘C’, echoes
back ‘C’
host ACKs
receipt
of echoed
‘C’
simple telnet scenario
time
Transport Layer 3-56
TCP Round Trip Time and Timeout
TCP ラウンドトリップ時間とタイムアウト
Q: how to set TCP
timeout value?
TCPはどのようにタイムア
ウト時間を設定するのか
 longer than RTT

but RTT varies
 too short: premature
timeout
 unnecessary
retransmissions
 too long: slow reaction
to segment loss
Q: how to estimate RTT?
RTTをどのように見積もるか?
 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 3-57
TCP Round Trip Time and Timeout
TCP ラウンドトリップ時間とタイムアウト
EstimatedRTT = (1- )*EstimatedRTT + *SampleRTT
 Exponential weighted moving average
指数加重移動平均
 influence of past sample decreases exponentially fast
過去のサンプルの影響は指数的に減少
 typical value:  = 0.125
典型的な値:  = 0.125
Transport Layer 3-58
Example RTT estimation:
RTTの推定の例
RTT: gaia.cs.umass.edu to fantasia.eurecom.fr
350
RTT (milliseconds)
300
250
200
150
100
1
8
15
22
29
36
43
50
57
64
71
78
85
92
99
106
time (seconnds)
SampleRTT
Estimated RTT
Transport Layer 3-59
TCP Round Trip Time and Timeout
TCP ラウンドトリップ時間とタイムアウト
Setting the timeout タイムアウトの設定
 EstimtedRTT plus “safety margin”

large variation in EstimatedRTT -> larger safety margin
 first estimate of how much SampleRTT deviates from
EstimatedRTT:
DevRTT = (1-)*DevRTT +
*|SampleRTT-EstimatedRTT|
(typically,  = 0.25)
Then set timeout interval:
TimeoutInterval = EstimatedRTT + 4*DevRTT
Transport Layer 3-60
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
高信頼データ転送
flow control
connection management
 3.6 Principles of
congestion control
 3.7 TCP congestion
control
Transport Layer 3-61
TCP reliable data transfer
TCP高信頼データ転送
 TCP creates rdt
service on top of IP’s
unreliable service
TCPはIPの低信頼サービスの
上に高信頼のサービスを作る
 Pipelined segments
パイプラインで繋がれたセグメント
 Cumulative acks
累積的なAck（認証）
 TCP uses single
retransmission timer
TCPは１つの再送タイマーを使
用
 Retransmissions are
triggered by:
再送は以下の場合に引き起こ
される：
 timeout events
 duplicate acks
 Initially consider
simplified TCP sender:
まず単純化されたTCP送信者
を考える
 ignore duplicate acks
 ignore flow control,
congestion control
Transport Layer 3-62
TCP sender events:
TCP送信者イベント：
data rcvd from app:
アプリから届いたデータ：
 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: タイムアウト
 retransmit segment
that caused timeout
 restart timer
Ack rcvd: 届いたAck
 If acknowledges
previously unacked
segments


update what is known to
be acked
start timer if there are
outstanding segments
Transport Layer 3-63
NextSeqNum = InitialSeqNum
SendBase = InitialSeqNum
loop (forever) {
switch(event)
event: data received from application above
create TCP segment with sequence number NextSeqNum
if (timer currently not running)
start timer
pass segment to IP
NextSeqNum = NextSeqNum + length(data)
event: timer timeout
retransmit not-yet-acknowledged segment with
smallest sequence number
start timer
event: ACK received, with ACK field value of y
if (y > SendBase) {
SendBase = y
if (there are currently not-yet-acknowledged segments)
start timer
}
} /* end of loop forever */
TCP
sender
(simplified)
Comment:
• SendBase-1: last
cumulatively
ack’ed byte
Example:
• SendBase-1 = 71;
y= 73, so the rcvr
wants 73+ ;
y > SendBase, so
that new data is
acked
Transport Layer 3-64
TCP: retransmission scenarios
TCP: 再送シナリオ
Host A
X
loss
Sendbase
= 100
SendBase
= 120
SendBase
= 100
time
SendBase
= 120
lost ACK scenario
Host B
Seq=92 timeout
Host B
Seq=92 timeout
timeout
Host A
time
premature timeout
Transport Layer 3-65
TCP retransmission scenarios (more)
TCP再送シナリオ（続き）
timeout
Host A
Host B
X
loss
SendBase
= 120
time
Cumulative ACK scenario
Transport Layer 3-66
TCP ACK generation
TCP ACK生成
[RFC 1122, RFC 2581]
Event at Receiver
TCP Receiver action
Arrival of in-order segment with
expected seq #. All data up to
expected seq # already ACKed
Delayed ACK. Wait up to 500ms
for next segment. If no next segment,
send ACK
Arrival of in-order segment with
expected seq #. One other
segment has ACK pending
Immediately send single cumulative
ACK, ACKing both in-order segments
Arrival of out-of-order segment
higher-than-expect seq. # .
Gap detected
Immediately send duplicate ACK,
indicating seq. # of next expected byte
Arrival of segment that
partially or completely fills gap
Immediate send ACK, provided that
segment startsat lower end of gap
Transport Layer 3-67
Fast Retransmit 高速再送
 Time-out period often
relatively long:
タイムアウト間隔は，相対的に
長い：
 long delay before
resending lost packet
 Detect lost segments
via duplicate ACKs.
重複ACKによるセグメントロス
の検出
 Sender often sends
many segments back-toback
 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:
始点ホストが同一データに対す
る ACK を３つ受信した場合，
ACKが対象とするセグメントが
失われた後のセグメントに対す
るものと推定する:
 fast retransmit: resend
segment before timer
expires
Transport Layer 3-68
Fast retransmit algorithm:
高速再送アルゴリズム：
event: ACK received, with ACK field value of y
if (y > SendBase) {
SendBase = y
if (there are currently not-yet-acknowledged segments)
start timer
}
else {
increment count of dup ACKs received for y
if (count of dup ACKs received for y = 3) {
resend segment with sequence number y
}
a duplicate ACK for
already ACKed segment
fast retransmit
Transport Layer 3-69
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
フロー制御
connection management
 3.6 Principles of
congestion control
 3.7 TCP congestion
control
Transport Layer 3-70
TCP Flow Control
TCPフロー制御
 receive side of TCP
connection has a receive
buffer:
TCPコネクションの終点ホストは受
信バッファをもつ：
flow control
sender won’t overflow
receiver’s buffer by
transmitting too much,
too fast
 speed-matching service:
 app process may be slow at
matching the send rate to
the receiving app’s drain
rate
スピード制御サービス：アプリ
ケーション読み出し速度に送
信速度を調整する
reading from buffer
アプリプロセスは，バッファからの
読み出しに遅れるかもしれない
Transport Layer 3-71
TCP Flow control: how it works
TCPフロー制御：どう動作するか
 Rcvr advertises spare
(Suppose TCP receiver
discards out-of-order
segments)
 spare room in buffer
room by including value
of RcvWindow in
segments
 Sender limits unACKed
data to RcvWindow

guarantees receive
buffer doesn’t overflow
= RcvWindow
= RcvBuffer-[LastByteRcvd LastByteRead]
Transport Layer 3-72
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
コネクション管理
 3.6 Principles of
congestion control
 3.7 TCP congestion
control
Transport Layer 3-73
TCP Connection Management コネクション管理
Recall: TCP sender, receiver
establish “connection”
before exchanging data
segments
 initialize TCP variables:
 seq. #s
 buffers, flow control
info (e.g. RcvWindow)
 client: connection initiator
Socket clientSocket = new
Socket("hostname","port
number");

server: contacted by client
Socket connectionSocket =
welcomeSocket.accept();
Three way handshake:
３ステップでの接続設定
Step 1: client host sends TCP
SYN segment to server
 specifies initial seq #
 no data
Step 2: server host receives
SYN, replies with SYNACK
segment
server allocates buffers
 specifies server initial
seq. #
Step 3: client receives SYNACK,
replies with ACK segment,
which may contain data

Transport Layer 3-74
TCP Connection Management (cont.)
TCPコネクション管理（続き）
Closing a connection:
コネクションの終了:
client closes socket:
clientSocket.close();
client
server
close
Step 1: client end system sends
close
TCP FIN control segment to
server
replies with ACK. Closes
connection, sends FIN.
timed wait
Step 2: server receives FIN,
closed
Transport Layer 3-75
TCP Connection Management (cont.)
TCPコネクション管理（続き）
Step 3: client receives FIN,
replies with ACK.

client
server
closing
Enters “timed wait” will respond with ACK
to received FINs
closing
Step 4: server, receives
Note: with small
modification, can handle
simultaneous FINs.
timed wait
ACK. Connection closed.
closed
closed
Transport Layer 3-76
TCP Connection Management (cont)
TCPコネクション管理（続き）
TCP server
lifecycle
TCP client
lifecycle
Transport Layer 3-77
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
輻輳制御の原理
 3.7 TCP congestion
control
Transport Layer 3-78
Principles of Congestion Control
輻輳制御の原理
Congestion: 輻輳
 informally: “too many sources sending too much
data too fast for network to handle”
簡単にいうと: “多くのソースが大量のデータをネットワークが扱うこと
ができる速度より速く送信している”
 different from flow control!
フロー制御とは異なる！
 manifestations:
lost packets (buffer overflow at routers)
 long delays (queueing in router buffers)
 a top-10 problem!

Transport Layer 3-79
Causes/costs of congestion: scenario 1
輻輳の原因/コスト：シナリオ１
Host A
 two senders, two
receivers
 one router,
infinite buffers
 no retransmission
Host B
lout
lin : original data
unlimited shared
output link buffers
 large delays
when congested
 maximum
achievable
throughput
Transport Layer 3-80
Causes/costs of congestion: scenario 2
輻輳の原因/コスト：シナリオ２
 one router,
finite buffers
 sender retransmission of lost packet
Host A
Host B
lin : original
data
l'in : original data, plus
retransmitted data
lout
finite shared output
link buffers
Transport Layer 3-81
Causes/costs of congestion: scenario 2
輻輳の原因/コスト：シナリオ２
 always: l = l
(goodput)
out
in
 “perfect” retransmission only when loss:
l > lout
in
 retransmission of delayed (not lost) packet makes
(than perfect case) for same
R/2
l
in
lout
R/2
larger
R/2
lin
a.
R/2
lout
lout
lout
R/3
lin
b.
R/2
R/4
lin
R/2
c.
“costs” of congestion:
 more work (retrans) for given “goodput”
 unneeded retransmissions: link carries multiple copies of pkt
Transport Layer 3-82
Causes/costs of congestion: scenario 3
輻輳の原因/コスト：シナリオ３
 four senders
Q: what happens as l
in
and l increase ?
 multihop paths
 timeout/retransmit
in
Host A
lin : original data
lout
l'in : original data, plus
retransmitted data
finite shared output
link buffers
Host B
Transport Layer 3-83
Causes/costs of congestion: scenario 3
輻輳の原因/コスト：シナリオ３
H
o
s
t
A
l
o
u
t
H
o
s
t
B
Another “cost” of congestion:
 when packet dropped, any “upstream transmission
capacity used for that packet was wasted!
Transport Layer 3-84
Approaches towards congestion control
輻輳制御へのアプローチ
Two broad approaches towards congestion control:
輻輳制御のための２つの基本的アプローチ:
Network-assisted congestion
End-end congestion
control:
control:
エンド間輻輳制御:
 no explicit feedback from
network
 congestion inferred from
end-system observed loss,
delay
 approach taken by TCP
ネットワーク支援型輻輳制御:
 routers provide feedback to
end systems
 single bit indicating
congestion (SNA, DECbit,
TCP/IP ECN, ATM)
 explicit rate sender should
send at
Transport Layer 3-85
Case study: ATM ABR congestion control
ケーススタディ: ATM ABR 輻輳制御
ABR: available bit rate:
 “elastic service”
 if sender’s path
“underloaded”:
 sender should use
available bandwidth
 if sender’s path congested:
 sender throttled to
minimum guaranteed rate
RM (resource management)
cell:
資源管理 (RM: resource
management)セル:
 sent by sender, interspersed
with data cells
 bits in RM cell set by switches
(“network-assisted”)
 NI bit: no increase in rate
(mild congestion)
 CI bit: congestion
indication
 RM cells returned to sender by
receiver, with bits intact
Transport Layer 3-86
Case study: ATM ABR congestion control
ケーススタディ: ATM ABR 輻輳制御
 two-byte ER (explicit rate) field in RM cell
 congested switch may lower ER value in cell
 sender’ send rate thus minimum supportable rate on path
 EFCI bit in data cells: set to 1 in congested switch
 if data cell preceding RM cell has EFCI set, sender sets CI
bit in returned RM cell
Transport Layer 3-87
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
TCP 輻輳制御
Transport Layer 3-88
TCP Congestion Control
TCP 輻輳制御
 end-end control (no network
assistance)
 sender limits transmission:
LastByteSent-LastByteAcked
 CongWin
 Roughly,
rate =
CongWin
Bytes/sec
RTT
 CongWin is dynamic, function
of perceived network
congestion
How does sender perceive
congestion?
始点ホストはどうやって輻輳を知
覚するのか?
 loss event = timeout or 3
duplicate acks
 TCP sender reduces rate
(CongWin) after loss event
three mechanisms:



AIMD
slow start
conservative after timeout
events
Transport Layer 3-89
TCP AIMD
multiplicative decrease:
乗算的減少:
cut CongWin in half
after loss event
congestion
window
24 Kbytes
additive increase:
加算的増加:
increase CongWin by
1 MSS every RTT in
the absence of loss
events: probing
16 Kbytes
8 Kbytes
time
Long-lived TCP connection
Transport Layer 3-90
TCP Slow Start
TCP スロースタート
 When connection begins,
CongWin = 1 MSS


Example: MSS = 500
bytes & RTT = 200 msec
initial rate = 20 kbps
 available bandwidth may
be >> MSS/RTT

 When connection begins,
increase rate
exponentially fast until
first loss event
コネクション開始時は，最初のロ
スイベントまで，レートを指数関
数的に増加させることが望ましい
desirable to quickly ramp
up to respectable rate
Transport Layer 3-91
TCP Slow Start (more)
TCP スロースタート（続き）
 When connection begins,
コネクション開始時，最初のロスイ
ベントまでレートを指数関数的に
増加：
 double CongWin every
RTT
 done by incrementing
CongWin for every ACK
received
Host B
RTT
increase rate exponentially
until first loss event:
Host A
 Summary: initial rate is
slow but ramps up
exponentially fast
まとめると…: 初期レートは小さい
が，指数関数的に増加
time
Transport Layer 3-92
Refinement 改善点
Philosophy:
 After 3 dup ACKs:
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
 CongWin
• 3 dup ACKs indicates
network capable of
delivering some segments
• timeout before 3 dup
ACKs is “more alarming”
Transport Layer 3-93
Refinement (more)
改善（続き）
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
Transport Layer 3-94
Summary: TCP Congestion Control
まとめ: TCP 輻輳制御
 When CongWin is below Threshold, sender in slow-start
phase, window grows exponentially.
CongWin が Threshold 以下のとき，始点ホストはスロースタートフェー
ズに入り，ウインドウを指数的に増加させる
 When CongWin is above Threshold, sender is in congestion-
avoidance phase, window grows linearly.
CongWin が Threshold を超えると，始点ホストは，輻輳回避フェーズに
入り，ウインドウを線形的に増加させる
 When a triple duplicate ACK occurs, Threshold set to
CongWin/2 and CongWin set to Threshold.
３重ACK受信後，Threshold を CongWin/2 に設定し，CongWin を閾
値に設定する
 When timeout occurs, Threshold set to CongWin/2 and
CongWin is set to 1 MSS.
タイムアウト発生後は，閾値を CongWin/2 に設定し，CongWin を 1 MSS
に設定する
Transport Layer
3-95
TCP sender congestion control
TCP始点ホスト輻輳管理
State
Event
TCP Sender Action
Commentary
Slow Start
(SS)
ACK receipt
for previously
unacked
data
CongWin = CongWin + MSS,
If (CongWin > Threshold)
set state to “Congestion
Avoidance”
Resulting in a doubling of
CongWin every RTT
Congestion
Avoidance
(CA)
ACK receipt
for previously
unacked
data
CongWin = CongWin+MSS *
(MSS/CongWin)
Additive increase, resulting
in increase of CongWin by
1 MSS every RTT
SS or CA
Loss event
detected by
triple
duplicate
ACK
Threshold = CongWin/2,
CongWin = Threshold,
Set state to “Congestion
Avoidance”
Fast recovery,
implementing multiplicative
decrease. CongWin will not
drop below 1 MSS.
SS or CA
Timeout
Threshold = CongWin/2,
CongWin = 1 MSS,
Set state to “Slow Start”
Enter slow start
SS or CA
Duplicate
ACK
Increment duplicate ACK count
for segment being acked
CongWin and Threshold not
changed
Transport Layer 3-96
TCP throughput
TCPスループット
 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
Transport Layer 3-97
TCP Futures TCPの将来
 Example: 1500 byte segments, 100ms RTT, want 10
Gbps throughput
 Requires window size W = 83,333 in-flight
segments
 Throughput in terms of loss rate:
1.22  MSS
RTT L
 ➜ L = 2·10-10
Wow
 New versions of TCP for high-speed needed!
Transport Layer 3-98
TCP Fairness TCPの公平性
Fairness goal: if K TCP sessions share same
bottleneck link of bandwidth R, each should have
average rate of R/K
公平性の目的: K本のTCPセッションが帯域 R の同一ボトルネックリン
クを共有している場合，各セッションは平均 R/Kの速度を持つべき
である
TCP connection 1
TCP
connection 2
bottleneck
router
capacity R
Transport Layer 3-99
Why is TCP fair?
TCP はなぜ公平？
Two competing sessions: ２本の競合セッション:
 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 3-100
Fairness (more) 公平性（続き）
Fairness and UDP
公平性と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
Fairness and parallel TCP
connections
公平性と並列TCPコネクション
 nothing prevents app from
opening parallel cnctions
between 2 hosts.
 Web browsers do this
 Example: link of rate R
supporting 9 cnctions;


new app asks for 1 TCP, gets
rate R/10
new app asks for 11 TCPs,
gets R/2 !
Transport Layer 3-101
Delay modeling 遅延モデル
Q: How long does it take to
receive an object from a Web
server after sending a
request?
Ignoring congestion, delay is
influenced by:
輻輳を無視すると，遅延は以下に影響され
る:
 TCP connection establishment
 data transmission delay
 slow start
Notation, assumptions:
記号と仮定:
 Assume one link between
client and server of rate R
 S: MSS (bits)
 O: object size (bits)
 no retransmissions (no loss,
no corruption)
Window size:
 First assume: fixed
congestion window, W
segments
 Then dynamic window,
modeling slow start
Transport Layer 3-102
Fixed congestion window (1)
固定輻輳ウインドウ (1)
First case:
WS/R > RTT + S/R: ACK for
first segment in window
returns before window’s
worth of data sent
ケース１:
WS/R > RTT + S/R: ウインドウ内のデ
ータを全て送信する前にウインドウ
内の第１セグメントに対する ACK が
返ってくる
delay = 2RTT + O/R
Transport Layer 3-103
Fixed congestion window (2)
Second case:
 WS/R < RTT + S/R: wait
for ACK after sending
window’s worth of data
sent
ケース２:
 WS/R < RTT + S/R: ウイン
ドウ内のデータを送信した後
，ACKが返ってくるのを待つ
delay = 2RTT + O/R
+ (K-1)[S/R + RTT - WS/R]
Transport Layer 3-104
TCP Delay Modeling: Slow Start (1)
TCP 遅延モデル: スロースタート (1)
Now suppose window grows according to slow start
ウインドウがスロースタートに従って増加すると仮定
Will show that the delay for one object is:
Latency  2 RTT 
O
S
S

 P  RTT    (2 P  1)
R
R
R

where P is the number of times TCP idles at server:
P  min{Q, K  1}
- where Q is the number of times the server idles
if the object were of infinite size.
- and K is the number of windows that cover the object.
Transport Layer 3-105
TCP Delay Modeling: Slow Start (2)
TCP 遅延モデル: スロースタート (2)
Delay components:
遅延要素:
• 2 RTT for connection
estab and request
• O/R to transmit
object
• time server idles due
to slow start
initiate TCP
connection
request
object
first window
= S/R
RTT
third window
= 4S/R
Server idles:
P = min{K-1,Q} times
Example:
• O/S = 15 segments
• K = 4 windows
•Q=2
• P = min{K-1,Q} = 2
Server idles P=2 times
second window
= 2S/R
fourth window
= 8S/R
complete
transmission
object
delivered
time at
client
time at
server
Transport Layer 3-106
TCP Delay Modeling (3) TCP 遅延モデル (3)
S
 RTT  timefrom when server startstosend segment
R
untilserver receivesacknowledgement
initiate TCP
connection
2k 1
S
 time to transmit the kth window
R

request
object
S
k 1 S 

RTT

2
R
  idle timeafter thekth window
R


first window
= S/R
RTT
second window
= 2S/R
third window
= 4S/R
P
O
delay   2 RTT   idleTim ep
R
p 1
P
O
S
S
  2 RTT   [  RTT  2 k 1 ]
R
R
k 1 R
O
S
S
  2 RTT  P[ RTT  ]  (2 P  1)
R
R
R
fourth window
= 8S/R
complete
transmission
object
delivered
time at
client
time at
server
Transport Layer 3-107
TCP Delay Modeling (4)
TCP 遅延モデル (４)
Recall K = number of windows that cover object
K = オブジェクト全てを送信するのに必要なウインドウ回数
How do we calculate K ?
K  min{k : 20 S  21 S    2 k 1 S  O}
 min{k : 20  21    2 k 1  O / S}
O
 min{k : 2  1  }
S
O
 min{k : k  log2 (  1)}
S
O


 log2 (  1)
S


k
Calculation of Q, number of idles for infinite-size object,
is similar (see HW).
Transport Layer 3-108
HTTP Modeling HTTP モデリング
 Assume Web page consists of:
Web ページが以下から構成されると仮定:


1 base HTML page (of size O bits)
M images (each of size O bits)
 Non-persistent HTTP: 非継続型 HTTP:


M+1 TCP connections in series
Response time = (M+1)O/R + (M+1)2RTT + sum of idle
times
 Persistent HTTP: 継続型 HTTP:



2 RTT to request and receive base HTML file
1 RTT to request and receive M images
Response time = (M+1)O/R + 3RTT + sum of idle times
 Non-persistent HTTP with X parallel connections




Suppose M/X integer.
1 TCP connection for base file
M/X sets of parallel connections for images.
Response time = (M+1)O/R + (M/X + 1)2RTT + sum of idle
times
Transport Layer 3-109
HTTP Response time (in seconds)
HTTP 応答時間 (秒単位)
RTT = 100 msec, O = 5 Kbytes, M=10 and X=5
20
18
16
14
12
10
8
6
4
2
0
non-persistent
persistent
parallel nonpersistent
28
100
1
10
Kbps Kbps Mbps Mbps
For low bandwidth, connection & response time dominated by
transmission time.
Persistent connections only give minor improvement over parallel
connections.
Transport Layer 3-110
HTTP Response time (in seconds)
HTTP 応答時間 (秒単位)
RTT =1 sec, O = 5 Kbytes, M=10 and X=5
70
60
50
non-persistent
40
persistent
30
20
parallel nonpersistent
10
0
28
100
1
10
Kbps Kbps Mbps Mbps
For larger RTT, response time dominated by TCP establishment
& slow start delays. Persistent connections now give important
improvement: particularly in high delaybandwidth networks.
Transport Layer 3-111
Chapter 3: Summary まとめ
 principles behind transport
layer services:
トランスポート層サービスの背後に
ある原理:
 multiplexing,
demultiplexing
 reliable data transfer
 flow control
 congestion control
 instantiation and
implementation in the
Internet
インターネットにおける事例と実装
UDP
 TCP
Next:
 leaving the network
“edge” (application,
transport layers)
 into the network
“core”

Transport Layer 3-112

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