End To End Protocols

Understanding the Stack Recall the TCP/IP Internet Architecture

End-to-End Protocols (Week 12)

FTP

HTTP

NV

80

20,21

RTP

4444

UDP

TCP 6

17

IP

Jaringan Komputer NET1

NET2

…

NETn

Fakultas Ilmu Komputer Universitas Indonesia Semester Genap 2003/2004 Versi: 1

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Basic Transport-layer Function

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End-to-end Communication

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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Transport-layer Service Model

Internet Transport-layer Protocols

Transport layer: logical communication between application end-point point.

UDP: connectionless multiplexing/demultiplexing error detection

multiplexing/demultiplexing

Additional services:

TCP: connection oriented

reliable data transfer (guaranteed arrival, no error, inorder) flow control (keep sender from overrunning receiver): good for myself congestion control (keep sender from overrunning network): good for everybody

multiplexing/demultiplexing reliable data transfer flow control congestion control

services not available: delay guarantees bandwidth guarantees

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Port Numbers

How multiplexing/demultiplexing works?

Each port number is a 16-bit number, ranging from 0 to 65535. Port numbers ranging from 0 to 1023 are called wellknown port numbers and are restricted. Port number vs. socket

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??

socket (true destination attached to app. end-point) port number (a mechanism to identify socket)

Analogy PABX system vs Internet:

dest. IP address for routing to the host; IP addresses and port numbers for going to appropriate socket in the dest. host.

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Phone no ≈ Internet address Extension no ≈ Port no

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Simple Demultiplexor (UDP) Unreliable and unordered datagram service Adds multiplexing 0 SrcPort No flow control Checksum Endpoints identified by ports servers have well-known ports see /etc/services on Unix

UDP: User Datagram Protocol What is a connection? 16

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

31 DstPort

UDP is connectionless:

Length

each UDP segment handled independently of others

Data

UDP does multiplexing/demultiplexing simple error detection

Header format Optional checksum

UDP does not do

pseudo header + UDP header + data

reliable data transfer, flow control, congestion control …

Pseudo header consists of:

Protocol no (6 for TCP, 17 for UDP) Source IP Destination IP Length field

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What is good about UDP?

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UDP: more

TCP features may not be needed by some applications, such as? Less overhead:

Often used for streaming multimedia apps loss tolerant rate sensitive

no connection establishment (which can add delay) small segment header no congestion control: UDP can blast away as fast as desired

In general, UDP is also used when TCP features are not important What if you want a subset of features in TCP?

Simple: no connection state at sender, receiver

implemented at applicationlevel flow control and error recovery in many multimedia apps 11

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End-to-End Protocols

TCP Overview

Underlying best-effort network (IP service):

Full duplex Flow control: keep sender from overrunning receiver Congestion control: keep sender from overrunning network

Connection-oriented Byte-stream

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

app writes Bytes TCP sends segments app reads Bytes

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

Application process

Application process

…

…

Write Bytes

TCP Receive buffer

TCP Send buffer

Segment

Read Bytes

Segment

…

Segment

Transmit segments

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Simple Reliability: send/ACK

Reliable Data Transfer

Sender

Our goal: end-to-end solution to achieve reliable data transfer What is reliable data transfer?

Receiver

Timeout

Time

ACK

Timeout (a)

Sender

Receiver

Timeout

Fram e

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

Timeout

Fram

16

ACK

duplication

Receiver Fram e ACK

Fram e ACK

ACK

(b) Versi 1

e

Fram e

Sender

Timeout

end-to-end solution has no control of underlying communication channel, which can be error-prone and lossy

ACK

(c)

Timeout

Why is it difficult?

Receiver Fram e

guaranteed arrival no error in order delivery

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Sender

Fram e

Timeout

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(d)

duplication Versi 1

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Stop-and-Wait

Bandwidth & Latency (Review)

Problem: Overhead ACK: min. 1 RTT, sender stop Example

Kinerja jaringan diukur dalam dua kategori: Bandwidth (throughput): jumlah bits yang dapat ditransfer 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.

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

Latency (delay): berapa lama waktu yang diperlukan untuk mengirimkan “message” dari satu ujung (end) ke ujung lainnya.

Receiver

Length = latency bandwidth

• Ukuran latency adalah satuan waktu. • Misalkan: latency untuk jaringan JKT – SBY: 20 milidetik (oneway). • Pengukuran lain Round-Trip Time (RTT): latency message bolak balik (two way).

Capacity = bandwidth X latency

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Example: Latency Network (Review) A R1

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

Destination

R3

Host A

R4

Host A R1 R2

TRANSP1 Q2

TRANSP1

TRANSP2

PROP1

PROP1

R2 R3

TRANSP3

PROP2

TRANSP2

R1

“Store-and-Forward” at each Router

Host B

TRANSP4

R3

Host B

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Example: Latency Network (Review)

B

R2

Source

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TRANSP3 PROP2

TRANSP4 PROP3

PROP3

PROP4

Actual end to end latency = ∑ (TRANSPi + PROPi + Qi )

PROP4

Minimum end to end latency = ∑ (TRANSPi + PROPi )

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

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)

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.

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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

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*Catatan: Di sini RTT = propagation delay

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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

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Segment Format

Sliding Window Allow multiple outstanding (un-ACKed) Bytes Upper bound on un-ACKed Bytes, called window

0

10

4

16

31

SrcPort

Sender

DstPort

Receiver

SequenceNum Acknowledgment

…

Time

HdrLen

0

Flags

AdvertisedWindow

Checksum

UrgPtr Options (variable)

…

Data

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Segment Format (cont)

Connection Establishment and Termination

Each connection identified with 4-tuple:

Three way handshake

(SrcPort, SrcIPAddr, DsrPort, DstIPAddr)

Sliding window + flow control

Active participant (client)

acknowledgment, SequenceNum, AdvertisedWinow

SYN, Seq

Data(SequenceNum) Sender

Passive participant (server) uence Num =

x

= y, qu e x+1 S = , K ment + AC g d N Y le S w

Receiver

Num ence

Acknowledgment + AdvertisedWindow

o Ackn ACK, Ackno wledg me

Flags SYN, FIN, RESET, PUSH, URG, ACK

Checksum

nt = y

+1

pseudo header + TCP header + data 25

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Flow Control – Credit Allocation

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Sending and Receiving Perspectives

[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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TCP Congestion Control

Receiver vs Network Capacity

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 4th ed. Prentice-Hall: 2003, § 6.5.9. (a) A fast network feeding a low-capacity receiver 29

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Masalah & Penyelesaiannya

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Effective Window Size

Masalah:

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.

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. 31

(b) A slow network feeding a high capacity network

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Slow Start & Threshold

Algoritma Slow Start & Threshold

Mekanisme yg dilakukan ES asal utk memperkirakan kapasitas jaringan. Slow start (Jacobson 1988):

1. 2.

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? 33

3.

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Slow Start & Threshold

Congestion window diberi nilai 1 segmen. Dilakukan slow start sampai congestion window mencapai threshold (pertambahan secara ekponesial). Kemudian congestion window bertambah secara linier, hingga mencapai receiver window atau terjadi timeout. Jika terjadi timeout, threshold diperkecil menjadi ½ dr congestion window terakhir. Kembali ke langkah 1.

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E.g. 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.

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