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2017 session 1 TELE3118: Network Technologies Week 9: Transport Layer Basics, UDP
Slides have been adapted from: Computer Networking: A Top Down Approach, 7th (global) edition. Jim Kurose, Keith Ross. Pearson, April All material copyright J.F Kurose and K.W. Ross, All Rights Reserved. Computer Networks, 5th edition. Andrew S. Tanenbaum, David J. Wetherall, Pearson, 2010. Transport Layer
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Transport services and protocols
application transport network data link physical provide logical communication between app 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 logical end-end transport application transport network data link physical Transport Layer
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Transport vs. network layer
network layer: logical communication between hosts transport layer: logical communication between processes relies on, enhances, network layer services household analogy: 12 kids in Ann’s house sending letters to 12 kids in Bill’s house: hosts = houses processes = kids app messages = letters in envelopes transport protocol = Ann and Bill who demux to in-house siblings network-layer protocol = postal service Transport Layer
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Internet transport-layer protocols
application transport network data link physical reliable, in-order delivery (TCP) congestion control flow control connection setup unreliable, unordered delivery: UDP no-frills extension of “best-effort” IP services not available: delay guarantees bandwidth guarantees network data link physical network data link physical network data link physical logical end-end transport network data link physical network data link physical network data link physical network data link physical application transport network data link physical Transport Layer
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Multiplexing/demultiplexing
handle data from multiple sockets, add transport header (later used for demultiplexing) multiplexing at sender: use header info to deliver received segments to correct socket demultiplexing at receiver: application application P1 P2 application socket P3 transport P4 process transport network transport link network network physical link link physical physical Transport Layer
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How demultiplexing works
host receives IP datagrams each datagram has source IP address, destination IP address each datagram carries one transport-layer segment each segment has source, destination port number host uses IP addresses & port numbers to direct segment to appropriate socket 32 bits source port # dest port # other header fields application data (payload) TCP/UDP segment format Transport Layer
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Connectionless demultiplexing
recall: when creating datagram to send into UDP socket, must specify destination IP address destination port # recall: created socket has host-local port #: DatagramSocket mySocket = new DatagramSocket(12534); when host receives UDP segment: checks destination port # in segment directs UDP segment to socket with that port # IP datagrams with same dest. port #, but different source IP addresses and/or source port numbers will be directed to same socket at dest Transport Layer
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Connectionless demux: example
DatagramSocket serverSocket = new DatagramSocket (6428); DatagramSocket mySocket2 = new DatagramSocket (9157); DatagramSocket mySocket1 = new DatagramSocket (5775); application application application P1 P3 P4 transport transport transport network network network link link link physical physical physical source port: 6428 dest port: 9157 source port: ? dest port: ? source port: 9157 dest port: 6428 source port: ? dest port: ? Transport Layer
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Connection-oriented demux
TCP socket identified by 4-tuple: source IP address source port number dest IP address dest port number demux: receiver uses all four values to direct segment to appropriate socket server host may support many simultaneous TCP sockets: each socket identified by its own 4-tuple web servers have different sockets for each connecting client non-persistent HTTP will have different socket for each request Transport Layer
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Connection-oriented demux: example
application application application P4 P5 P6 P3 P2 P3 transport transport transport network network network link link link physical physical physical server: IP address B source IP,port: B,80 dest IP,port: A,9157 host: IP address C host: IP address A source IP,port: C,5775 dest IP,port: B,80 source IP,port: A,9157 dest IP, port: B,80 source IP,port: C,9157 dest IP,port: B,80 three segments, all destined to IP address: B, dest port: 80 are demultiplexed to different sockets Transport Layer
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Connection-oriented demux: example
threaded server application application application P4 P3 P2 P3 transport transport transport network network network link link link physical physical physical server: IP address B source IP,port: B,80 dest IP,port: A,9157 host: IP address C host: IP address A source IP,port: C,5775 dest IP,port: B,80 source IP,port: A,9157 dest IP, port: B,80 source IP,port: C,9157 dest IP,port: B,80 Transport Layer
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UDP: User Datagram Protocol [RFC 768]
“no frills,” “bare bones” Internet transport protocol “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 UDP use: streaming multimedia apps (loss tolerant, rate sensitive) DNS SNMP reliable transfer over UDP: add reliability at application layer application-specific error recovery! Transport Layer
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UDP: segment header why is there a UDP?
length, in bytes of UDP segment, including header 32 bits source port # dest port # length checksum why is there a UDP? no connection establishment (which can add delay) simple: no connection state at sender, receiver small header size no congestion control: UDP can blast away as fast as desired application data (payload) UDP segment format Transport Layer
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Reliable data transfer: principles
sender receiver sender receiver send pkt0 send pkt0 pkt0 pkt0 rcv pkt0 rcv pkt0 send ack0 send ack0 ack0 ack0 rcv ack0 rcv ack0 send pkt1 pkt1 send pkt1 pkt1 X loss rcv pkt1 ack1 send ack1 rcv ack1 send pkt0 pkt2 timeout resend pkt1 rcv pkt2 ack2 send ack2 pkt1 rcv pkt1 ack1 send ack1 rcv ack1 send pkt0 pkt2 (a) no loss rcv pkt2 ack2 send ack2 (b) packet loss Transport Layer
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rdt: Stop-and-wait operation
sender receiver first packet bit transmitted, t = 0 last packet bit transmitted, t = L / R first packet bit arrives RTT last packet bit arrives, send ACK ACK arrives, send next packet, t = RTT + L / R Stop-and-wait is correct, but performance stinks e.g.: 1 Gbps link, 15 ms prop. delay, 8000 bit packet: Transport Layer
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rdt with pipelining 9-packet pipelining increases
sender receiver first packet bit transmitted, t = 0 last bit transmitted, t = L / R first packet bit arrives RTT last packet bit arrives, send ACK last bit of 2nd packet arrives, send ACK last bit of 3rd packet arrives, send ACK ACK arrives, send next packet, t = RTT + L / R 9-packet pipelining increases utilization by a factor of 3! Transport Layer
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Pipelined protocols pipelining: sender allows multiple, “in-flight”, yet-to-be-acknowledged pkts range of sequence numbers must be increased buffering at sender and/or receiver two generic forms of pipelined protocols: go-Back-N, selective repeat Transport Layer
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Pipelined protocols: overview
Go-back-N: sender can have up to N unacked packets in pipeline receiver only sends cumulative ack doesn’t ack packet if there’s a gap sender has timer for oldest unacked packet when timer expires, retransmit all unacked packets Selective Repeat: sender can have up to N unack’ed packets in pipeline rcvr sends individual ack for each packet sender maintains timer for each unacked packet when timer expires, retransmit only that unacked packet Transport Layer
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Go-Back-N: sender k-bit seq # in pkt header
“window” of up to N, consecutive unack’ed pkts allowed ACK(n): ACKs all pkts up to, including seq # n - “cumulative ACK” may receive duplicate ACKs (see receiver) timer for oldest in-flight pkt timeout(n): retransmit packet n and all higher seq # pkts in window Transport Layer
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Selective repeat receiver individually acknowledges all correctly received pkts buffers pkts, as needed, for eventual in-order delivery to upper layer sender only resends pkts for which ACK not received sender timer for each unACKed pkt sender window N consecutive seq #’s limits seq #s of sent, unACKed pkts Transport Layer
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Selective repeat: sender, receiver windows
Transport Layer
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TCP: Overview RFCs: 793,1122,1323, 2018, 2581 point-to-point:
full duplex data: bi-directional data flow in same connection MSS: maximum segment size connection-oriented: handshaking (exchange of control msgs) inits sender, receiver state before data exchange flow controlled: sender will not overwhelm receiver point-to-point: one sender, one receiver reliable, in-order byte steam: no “message boundaries” pipelined: TCP congestion and flow control set window size Transport Layer
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TCP segment structure source port # dest port # sequence number
32 bits URG: urgent data (generally not used) counting by bytes of data (not segments!) source port # dest port # sequence number ACK: ACK # valid acknowledgement number head len not used PSH: push data now (generally not used) U A P R S F receive window # bytes rcvr willing to accept checksum Urg data pointer RST, SYN, FIN: connection estab (setup, teardown commands) options (variable length) application data (variable length) Internet checksum (as in UDP) Transport Layer
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TCP seq. numbers, ACKs sequence numbers:
source port # dest port # sequence number acknowledgement number checksum rwnd urg pointer outgoing segment from sender sequence numbers: byte stream “number” of first byte in segment’s data acknowledgements: 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 window size N sender sequence number space source port # dest port # sequence number acknowledgement number checksum rwnd urg pointer incoming segment to sender sent ACKed sent, not-yet ACKed (“in-flight”) usable but not yet sent not usable A Transport Layer
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TCP seq. numbers, ACKs simple telnet scenario Host A Host B User types
Seq=42, ACK=79, data = ‘C’ host ACKs receipt of ‘C’, echoes back ‘C’ Seq=79, ACK=43, data = ‘C’ host ACKs receipt of echoed ‘C’ Seq=43, ACK=80 simple telnet scenario Transport Layer
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TCP round trip time, timeout
Q: how to set TCP timeout value? longer than RTT but RTT varies too short: premature timeout, unnecessary retransmissions too long: slow reaction to segment loss Q: how to estimate RTT? 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
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TCP round trip time, timeout
EstimatedRTT = (1- )*EstimatedRTT + *SampleRTT exponential weighted moving average influence of past sample decreases exponentially fast typical value: = 0.125 RTT: gaia.cs.umass.edu to fantasia.eurecom.fr RTT (milliseconds) sampleRTT EstimatedRTT time (seconds) Transport Layer
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TCP round trip time, timeout
timeout interval: EstimatedRTT plus “safety margin” large variation in EstimatedRTT -> larger safety margin estimate SampleRTT deviation from EstimatedRTT: DevRTT = (1-)*DevRTT + *|SampleRTT-EstimatedRTT| (typically, = 0.25) TimeoutInterval = EstimatedRTT + 4*DevRTT estimated RTT “safety margin” * Check out the online interactive exercises for more examples: Transport Layer
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TCP reliable data transfer
TCP creates rdt service on top of IP’s unreliable service pipelined segments cumulative acks single retransmission timer retransmissions triggered by: timeout events duplicate acks let’s initially consider simplified TCP sender: ignore duplicate acks ignore flow control, congestion control Transport Layer
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TCP sender events: timeout: data rcvd from app:
retransmit segment that caused timeout restart timer ack rcvd: if ack acknowledges previously unacked segments update what is known to be ACKed start timer if there are still unacked segments data rcvd from app: 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 Transport Layer
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TCP sender (simplified)
create segment, seq. #: NextSeqNum pass segment to IP (i.e., “send”) NextSeqNum = NextSeqNum + length(data) if (timer currently not running) start timer data received from application above L wait for event NextSeqNum = InitialSeqNum SendBase = InitialSeqNum retransmit not-yet-acked segment with smallest seq. # start timer timeout if (y > SendBase) { SendBase = y /* SendBase–1: last cumulatively ACKed byte */ if (there are currently not-yet-acked segments) start timer else stop timer } ACK received, with ACK field value y Transport Layer
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TCP: retransmission scenarios
Host A Host B Host A Host B SendBase=92 Seq=92, 8 bytes of data Seq=92, 8 bytes of data Seq=100, 20 bytes of data timeout timeout ACK=100 X ACK=100 ACK=120 Seq=92, 8 bytes of data Seq=92, 8 bytes of data SendBase=100 SendBase=120 ACK=100 ACK=120 SendBase=120 lost ACK scenario premature timeout Transport Layer
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TCP: retransmission scenarios
Host A Host B timeout Seq=92, 8 bytes of data Seq=100, 20 bytes of data ACK=100 X ACK=120 Seq=120, 15 bytes of data cumulative ACK Transport Layer
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TCP ACK generation [RFC 1122, RFC 2581]
event at receiver arrival of in-order segment with expected seq #. All data up to expected seq # already ACKed expected seq #. One other segment has ACK pending arrival of out-of-order segment higher-than-expect seq. # . Gap detected arrival of segment that partially or completely fills gap TCP receiver action delayed ACK. Wait up to 500ms for next segment. If no next segment, send ACK immediately send single cumulative ACK, ACKing both in-order segments immediately send duplicate ACK, indicating seq. # of next expected byte immediate send ACK, provided that segment starts at lower end of gap Transport Layer
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TCP fast retransmit if sender receives 3 ACKs for same data
time-out period often relatively long: long delay before resending lost packet detect lost segments via duplicate ACKs. sender often sends many segments back-to-back if segment is lost, there will likely be many duplicate ACKs. TCP fast retransmit if sender receives 3 ACKs for same data (“triple duplicate ACKs”), resend unacked segment with smallest seq # likely that unacked segment lost, so don’t wait for timeout (“triple duplicate ACKs”), Transport Layer
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TCP fast retransmit X fast retransmit after sender
Host A Host B timeout Seq=92, 8 bytes of data Seq=100, 20 bytes of data X ACK=100 ACK=100 ACK=100 ACK=100 Seq=100, 20 bytes of data fast retransmit after sender receipt of triple duplicate ACK Transport Layer
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