End-to-End Protocols (Week 12)

Transkripsi

1 Understanding the Stack Recall the TCP/IP Internet Architecture End-to-End Protocols (Week 12) FTP HTTP NV RTP 80 20, TCP UDP 6 17 Jaringan Komputer IP NET 1 NET 2 NET n Fakultas Ilmu Komputer Universitas Indonesia Semester Genap 2003/2004 Versi: 1 2 Versi 1 Basic Transport-layer Function 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 3 Versi 1 4 Versi 1

2 Transport-layer Service Model Transport layer: logical communication between application end-point point. multiplexing/demultiplexing Additional services: 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 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 5 Versi 1 6 Versi 1 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 Port numbers ranging from 0 to 1023 are called wellknown 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 7 Versi 1 8 Versi 1

3 Simple Demultiplexor (UDP) Unreliable and unordered datagram service Adds multiplexing No flow control SrcPort Checksum 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 Data DstPort Length 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 9 Versi 1 10 Versi 1 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 applicationlevel flow control and error recovery in many multimedia apps 11 Versi 1 12 Versi 1

4 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 Connection-oriented Byte-stream app writes Bytes TCP sends segments app reads Bytes Application process TCP Send buffer Write Bytes Segment Segment Segment Transmit segments Full duplex Flow control: keep sender from overrunning receiver Congestion control: keep sender from overrunning network Application process TCP Receive buffer Read Bytes 13 Versi 1 14 Versi 1 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 15 Versi 1 Simple Reliability: send/ Time Timeout Sender Receiver Sender Receiver Timeout Timeout (b) (d) 16 Versi 1 Timeout Timeout Sender Receiver Sender Receiver (a) (c) Timeout Timeout duplication duplication

5 Stop-and-Wait Problem: Overhead : 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 bandwidth Length = latency Capacity = bandwidth X latency Bandwidth & Latency (Review) 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. 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 (oneway). Pengukuran lain Round-Trip Time (RTT): latency message bolak balik (two way). 17 Versi 1 18 Versi 1 Example: Latency Network (Review) A B R2 Source R1 R3 Destination R4 Host A Host B TRANSP1 Store-and-Forward at each Router TRANSP2 R1 PROP1 TRANSP3 R2 PROP2 TRANSP4 R3 PROP3 PROP4 Minimum end to end latency = ( TRANSP + PROP) i i i Example: Latency Network (Review) Kemungkinan output link sedang digunakan, maka paket harus antri (queued) di dalam buffer => delay antrian Host A R1 R2 R3 Host B TRANSP1 PROP1 Q2 TRANSP2 TRANSP3 PROP2 TRANSP4 Actual end to end latency = ( TRANSP + PROP + Q ) PROP3 i PROP4 i i i 19 Versi 1 20 Versi 1

6 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 * 10 6 ) s] - Latency = 0.25 s s = 5.71 second - *Catatan: Di sini RTT = propagation delay 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 = second. - *Catatan: Di sini RTT = propagation delay 21 Versi 1 22 Versi 1 Sliding Window Allow multiple outstanding (un-ed) Bytes Upper bound on un-ed Bytes, called window Sender Receiver Segment Format SrcPort DstPort SequenceNum Acknowledgment HdrLen 0 Flags AdvertisedWindow Time Checksum Options (variable) Data UrgPtr 23 Versi 1 24 Versi 1

7 Segment Format (cont) Each connection identified with 4-tuple: (SrcPort, SrcIPAddr, DsrPort, DstIPAddr) Sliding window + flow control acknowledgment, SequenceNum, AdvertisedWinow Sender Data (SequenceNum) Acknowledgment + AdvertisedWindow Flags SYN, FIN, RESET, PUSH, URG, Checksum pseudo header + TCP header + data Receiver Connection Establishment and Termination Active participant (client) Three way handshake SYN, SequenceNum = x SYN +, SequenceNum = y, Acknowledgment = x + 1, Acknowledgment = y + 1 Passive participant (server) 25 Versi 1 26 Versi 1 Flow Control Credit Allocation [STAL00] Stalling W., Data and Computer Communications 6 th ed, Prentice-Hall:2000, 17.1 Sending and Receiving Perspectives 1 segment = 200 octets Initial W = 7 segments (1400 octets) 27 Versi 1 28 Versi 1

8 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 4 th ed. Prentice-Hall: 2003, (a) A fast network feeding a low-capacity receiver (b) A slow network feeding a high capacity network 29 Versi 1 30 Versi 1 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. 31 Versi 1 32 Versi 1

9 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 dr segmen yg telah dikirimkan sebelum timeout. Bagaimana jika congestion window mencapai receiver window? 33 Versi 1 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 Versi 1 Slow Start & Threshold 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. 35 Versi 1 36 Versi 1