End-to-End Protocols (Week 12)

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Semester Genap 2003/2004 Versi: 1

Jaringan Komputer

Fakultas Ilmu Komputer Universitas Indonesia

End-to-End Protocols

(Week 12)

2 Versi 1

Understanding the Stack

Recall the TCP/IP Internet Architecture

…

FTP HTTP NV RTP

TCP UDP

IP

NET1 NET2 NETn

6 17

80

20,21 4444

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Basic Transport-layer Function

Network layer: end-to-end logical

communication between hosts

Transport layer (rely on network layer): logical

communication between application-level comm.

end-points

Multiple application-level end-points can reside in one host

Application-level end-points can be a Web browser/server,

a FTP client/server, etc

Transport layer: end-to-end implementation

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5 Versi 1

Transport-layer Service Model

Transport layer: logical communication between

application end-point point.

multiplexing/demultiplexing

Additional services:

reliable data transfer (guaranteed arrival, no error,

in-order)

flow control (keep sender from overrunning receiver):

good for myself

congestion control (keep sender from overrunning

network): good for everybody

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Internet Transport-layer Protocols

UDP: connectionless

multiplexing/demultiplexing

error detection

TCP: connection oriented

multiplexing/demultiplexing

reliable data transfer

flow control

congestion control

services not available:

delay guarantees

bandwidth guarantees

How multiplexing/demultiplexing works?

using port numbers

each IP datagram has source IP address, destination IP address

each IP datagram carries a transport-layer segment each segment has source, destination port number port number??

dest. IP address for routing to the host; IP addresses and

port numbers for going to appropriate socket in the dest.

host.

Port Numbers

Each port number is a 16-bit number, ranging from 0

to 65535.

Port numbers ranging from 0 to 1023 are called

well-known port numbers and are restricted.

Port number vs. socket

socket (true destination attached to app. end-point)

port number (a mechanism to identify socket)

Analogy PABX system vs Internet:

Phone no

≈ Internet address

Extension no

≈ Port no

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Simple Demultiplexor (UDP)

Unreliable and unordered datagram service

Adds multiplexing

No flow control

Endpoints identified by ports

servers have well-known ports see /etc/services on Unix

Header format

Optional checksum

pseudo header + UDP header + data

Pseudo header consists of:

Protocol no (6 for TCP, 17 for UDP) Source IP Destination IP Length field SrcPort DstPort Checksum Length Data 0 16 31 10 Versi 1

UDP: User Datagram Protocol

What is a connection?

a group of segments between the same pair of comm. endpoints allow for shared resources, provide services more efficiently

UDP is connectionless:

each UDP segment handled independently of others

UDP does

multiplexing/demultiplexing simple error detection

UDP does not do

reliable data transfer, flow control, congestion control …

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What is good about UDP?

TCP features may not be needed by some

applications, such as?

Less overhead:

no connection establishment (which can add delay)

small segment header

no congestion control: UDP can blast away as fast as

desired

Simple: no connection state at sender, receiver

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UDP: more

Often used for streaming

multimedia apps

loss tolerant rate sensitive

In general, UDP is also

used when TCP features

are not important

What if you want a subset

of features in TCP?

implemented at application-level

flow control and error recovery in many multimedia apps

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End-to-End Protocols

Underlying best-effort network (IP service):

drop messages re-orders messages

delivers duplicate copies of a given message limits messages to some finite size

delivers messages after an arbitrarily long delay

Common end-to-end services:

guarantee message delivery

deliver messages in the same order they are sent deliver at most one copy of each message support arbitrarily large messages support synchronization

allow the receiver to flow control the sender support multiple application processes on each host

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TCP Overview

Connection-oriented Byte-stream

app writes Bytes TCP sends segments app reads Bytes

Application process Write Bytes TCP Send buffer

Segment Segment Segment Transmit segments Application process Read Bytes TCP Receive buffer … … … Full duplex

Flow control: keep sender from overrunning receiver

Congestion control: keep sender from overrunning network

Reliable Data Transfer

Our goal: end-to-end solution to achieve reliable

data transfer

What is reliable data transfer?

guaranteed arrival

no error

in order delivery

Why is it difficult?

end-to-end solution has no control of underlying

communication channel, which can be error-prone

and lossy

Where is it used in computer networks?

reliable data link service on top of unreliable physical

layer

reliable transport service on top of unreliable IP

Simple Reliability: send/ACK

Sender Receiver Frame ACK T im eout Ti m e Sender Receiver Frame ACK T im eout Frame ACK T im eout Sender Receiver Frame ACK T im eout Frame ACK T im eout Sender Receiver Frame T im eout Frame ACK T im eout (a) (c) (b) (d) duplication duplication

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Stop-and-Wait

Problem: Overhead ACK: min. 1 RTT, sender

stop

Example

Mak. bit yang dapat dikirimkan: BW x latency.

BW = Bandwidth

1.5Mbps link x 45ms latency = 67.5Kb ≈ 8KB

1KB Byte setiap 90ms => 1/16 utilisasi BW (link)

Sender Receiver

bandw

idt

h

Length = latency

Capacity = bandwidth X latency

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Bandwidth & Latency (Review)

Kinerja jaringan diukur dalam dua kategori:

Bandwidth

(throughput): jumlah bits yang dapat

di-transfer dalam satu periode waktu

• Misalkan: 1 Mbits/detik => 1 Mbps, berarti dapat mengirimkan data

1 juta bit setiap detik;

• Bandwidth 1 Mbps, diperlukan waktu 1 mikro-detik untuk

mengirimkan 1 bit.

Latency

(delay): berapa lama waktu yang diperlukan

untuk mengirimkan “message” dari satu ujung (end) ke

ujung lainnya.

• Ukuran latency adalah satuan waktu.

• Misalkan: latency untuk jaringan JKT – SBY: 20 milidetik

(one-way).

• Pengukuran lain Round-Trip Time (RTT): latency message bolak

balik (two way).

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Example: Latency Network (Review)

Host A

Host B

R1 R2 R3 A R1 R2 R4 R3 B TRANSP1 TRANSP2 TRANSP3 TRANSP4 PROP1 PROP2 PROP3 PROP4 Source Destination

“Store-and-Forward” at each Router

(

i i

)

i

TRANSP PROP

=

∑

+

Minimum end to end latency

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Example: Latency Network (Review)

Host A

Host B

R1 R2 R3 TRANSP1 TRANSP2 TRANSP3 TRANSP4 PROP1 PROP2 PROP3 PROP4

(

i i i

)

i

TRANSP PROP Q

=

∑

+

Actual end to end latency

Q2

Kemungkinan output link sedang digunakan, maka paket harus antri (queued) di dalam buffer => delay antrian

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E.g. : Exercise 1.5 (Page 61) - Review

**Hitung waktu transfer 1000 KB file, asumsi *RTT=100ms,**

ukuran paket 1KB data, dan diperlukan 2 RTT untuk

handshaking awal.

a)

Badwidth 1.5 Mbps, dan paket data dikirim

secara kontinyu

(tidak terputus)

- Gunakan rumus latency dan perhitungkan semua faktor yang

memberikan kontribusi terjadinya delay dari sender ke receiver.

- Latency = [handshaking] + waktu propagasi [paket 1, one way] +

waktu transmisi

- Latency = [2 * RTT] + [RTT/2] + [BesarData/Bandwidth]

- Latency = [200ms] + [50 ms] + [1000KB/1.5Mbps]

- Latency = [200ms] + [50 ms] + [(1000*1024*8)/(1.5 * 106) s]

- Latency = 0.25 s + 5.46 s = 5.71 second

-

*Catatan: Di sini RTT = propagation delay

22 Versi 1

E.g. : Exercise 1.5 (Page 61) - Review

**Hitung waktu transfer 1000 KB file, asumsi *RTT=100ms,**

ukuran paket 1KB data, dan diperlukan 2 RTT untuk

handshaking awal.

b) Badwidth 1.5 Mbps, dan paket data tidak dikirim secara

kontinyu, tapi setiap satu paket dikirimkan sender harus

menunggu 1 RTT, kemudian mengirim paket berikutnya.

- Dengan cara ini terdapat overhead 1 RTT pada paket kedua, ketiga,

dst sampai paket ke-1000; paket pertama tidak perlu menunggu sehingga total delay dari 1000 paket tsb adalah 999 RTT.

- Latency = [handshaking] + waktu propagasi [paket 1, one way] +

waktu transmisi + [total delay overhead menunggu]

- Latency = 5.71 s + [999 * RTT]

- Latency = 105.61 second.

-

*Catatan: Di sini RTT = propagation delay

Sliding Window

Allow multiple outstanding (un-ACKed) Bytes

Upper bound on un-ACKed Bytes, called window

Sender Receiver Ti m e

……

Segment Format

Options (variable) Data Checksum SrcPort DstPort HdrLen 0 Flags UrgPtr AdvertisedWindow SequenceNum Acknowledgment 0 4 10 16 31

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Segment Format (cont)

Each connection identified with 4-tuple:

(SrcPort, SrcIPAddr, DsrPort, DstIPAddr)

Sliding window + flow control

acknowledgment, SequenceNum, AdvertisedWinow

Flags

SYN, FIN, RESET, PUSH, URG, ACK

Checksum

pseudo header + TCP header + data Sender Data (SequenceNum) Acknowledgment + AdvertisedWindow Receiver 26 Versi 1

Connection Establishment and Termination

Active participant

(client) Passive participant(server)

SYN, Seq_uenceNum

= x SYN + ACK , Sequence Num= y, ACK, Ackn owledgmen_{t = y + 1} Acknowledg ment = x + 1

Three way handshake

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Flow Control – Credit Allocation

[STAL00] Stalling W., Data and Computer Communications 6th_{ed, Prentice-Hall:2000, § 17.1}

1 segment = 200 octets Initial W = 7 segments (1400 octets)

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29 Versi 1

TCP Congestion Control

Yang lebih berperan mengendalikan

kemacetan adl lapisan transport (transport

layer).

Kemacetan dpt dikendalikan jika data rate

dikurangi, dan hal tsb merupakan porsi tugas

lapisan transport.

[TAN03] Tanenbaum, A.S., Computer

Networks 4

th

_{ed. Prentice-Hall: 2003,}

§ 6.5.9.

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Receiver vs Network Capacity

(a) A fast network feeding a low-capacity receiver (b) A slow network feeding a high capacity network

Masalah & Penyelesaiannya

Masalah:

Apakah TCP congestion control cukup jika hanya

mengandalkan ukuran jendela (window size) yg

ditentukan oleh End System (ES) tujuan? Lihat

slide 31.

Bagaimana dgn internal congestion pd slide

sebelum ini?

Penyelesaiannya:

Selain receiver window, perlu juga congestion

window.

Effective Window Size

Ukuran jendela yg aman menurut ES asal.

Min(receiver window size, congestion

window size).

Jika ES tujuan menyanggupi ukuran

jendela 8KB, tetapi ES asal mengetahui

kapasitas jaringan hanya 4KB -> ES asal

memilih jendela berukuran 4KB.

Jika ES tujuan menyanggupi ukuran

jendela 8KB, dan ES asal mengetahui

kapasitas jaringan 32 KB -> ES asal

memilih jendela berukuran 8KB.

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Slow Start & Threshold

Mekanisme yg dilakukan ES asal utk

memperkirakan kapasitas jaringan.

Slow start (Jacobson 1988):

Congestion window bertambah besar secara

eksponensial, sampai terjadi timeout atau

receiver window tercapai.

Penambahan congestion window terjadi jika ES

asal menerima ACK dr segmen yg telah

dikirimkan sebelum timeout.

Bagaimana jika congestion window mencapai

receiver window?

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Algoritma Slow Start & Threshold

1.

Congestion window diberi nilai 1 segmen.

2.

Dilakukan slow start sampai congestion

window mencapai threshold (pertambahan

secara ekponesial).

3.

Kemudian congestion window bertambah

secara linier, hingga mencapai receiver window

atau terjadi timeout.

4.

Jika terjadi timeout, threshold diperkecil

menjadi ½ dr congestion window terakhir.

Kembali ke langkah 1.

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Slow Start & Threshold

Dalam suatu koneksi TCP, ES tujuan dpt

mengubah ukuran receiver window. Lihat

slide 31.

ICMP Source quench akan dilaporkan ke

TCP & dianggap sbg timeout.

Timer management sangat penting &

ditentukan secara statistik.

36 Versi 1