Recall the TCPIP Internet Architecture

  Semester Genap 2003/2004

  End-to-End Protocols (Week 12)

Jaringan Komputer Fakultas Ilmu Komputer Universitas Indonesia

  Understanding the Stack

Recall the TCP/IP Internet Architecture

  … FTP HTTP NV RTP TCP UDP

  IP NET ₁ NET ₂ NET _n ^{6 17 80 20,21 4444} 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 End-to-end Communication

  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 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: ≈ Internet address

  Phone no ≈ Port no

  Extension no 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 16 31} 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 … 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 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 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 TCP Overview Full duplex Connection-oriented Byte-stream Flow control: keep sender from overrunning receiver app writes Bytes TCP sends segments Congestion control: keep sender from overrunning app reads Bytes network

  Application process Application process _{Bytes Bytes Write Read Send buffer Receive buffer TCP} … _TCP ^… _{Segment Segment Segment Transmit segments}

… 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 imeout Ti m e} ^{Sender Receiver} Frame _ACK

_T

_imeout

Frame _ACK

_T

_imeout

  Sender Receiver Frame _{ACK T}

_imeout

Frame _{ACK T}

_imeout

^{Sender Receiver} Frame _{T imeout} Frame _{ACK T imeout} ^{(a) (c)}

  (b) ^(d) duplication duplication Stop-and-Wait Problem: Overhead ACK: min. 1 RTT, sender stop Example Mak. bit yang dapat dikirimkan: BW x latency.

  BW = Bandwidth

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

  Length = latency bandwidth

  Capacity = bandwidth X latency 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). Example: Latency Network (Review) A

  B R2 Source

  Destination R1 R3 _TRANSP1 R4

  Host A _TRANSP2 “Store-and-Forward” at each Router R1 _{PROP1 TRANSP 3} R2 _{PROP2 TRANSP 4} R3 _PROP3

  Host B _{PROP 4} = ∑

• ( TRANSP PROP ) Minimum end to end latency _{i i i}

Example: Latency Network (Review) Kemungkinan output link sedang digunakan, maka paket harus antri (queued) di dalam buffer => delay antrian _TRANSP

  1 Host A _{Q 2 TRANSP}

₂

R1 _{PROP 1 TRANSP 3} R2 _{PROP 2 TRANSP 4} R3 _{PROP 3} Host B _{PROP 4} ( TRANSP PROP Q )

=

+ + Actual end to end latency _{i i i}

  ∑ _i 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. Badwidth 1.5 Mbps, dan paket data dikirim

  a) 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] ₆ ^- Latency = [200ms] + [50 ms] + [(1000*1024*8)/(1.5 * 10 ) s] ^- Latency = 0.25 s + 5.46 s = 5.71 second

• *Catatan: Di sini RTT = propagation delay

  

kontinyu, tapi setiap satu paket dikirimkan sender harus

menunggu 1 RTT, kemudian mengirim paket berikutnya.

_-

  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.

  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 T ime

  

……

Segment Format

  Options (variable) Data Checksum SrcPort DstPort

  HdrLen Flags UrgPtr AdvertisedWindow SequenceNum

  Acknowledgment

  4

  10

  16

  31 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 Connection Establishment and Termination Active participant (client)

  Passive participant (server) SYN, Se quenceN um = x

• 1
• AC K, Se quen ceNu m =

  SYN

  y ,

  ACK, Ack nowledgm ent = y + 1

  Ackn owle dgme nt = x

  Three way handshake Flow Control – Credit Allocation _th [STAL00] Stalling W., Data and Computer Communications 6 ed, Prentice-Hall:2000, § 17.1 _{1 segment = 200 octets}

  Initial W = 7 segments (1400 octets) Sending and Receiving Perspectives

  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 th

  Networks 4 ed. Prentice-Hall: 2003, § 6.5.9. 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. 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? 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. 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. E.g. Slow Start & Threshold