Chap.

3 Transport Layer
Goal : study principle of providing comm services to app processes and implementation issues in the Internet protocols, TCP and UDP Contents Relationship bw transport and net layers extending net layer’s delivery service to a delivery service bw two app-layer processes, by covering UDP Principles of reliable data transfer and TCP Principles of congestion control and TCP’s congestion control

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Chap.3 Transport Layer
Introduction and Transport-Layer Services Relationship Between Transport and Network Layers Overview of the Transport Layer in the Internet Multiplexing and Demultiplexing Connectionless Transport: UDP Principle of Reliable Data Transfer Connection-Oriented Transport: TCP Principles of Congestion Control TCP Congestion Control

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Overview of Transport-layer
provide logical comm bw app processes running on diff hosts transport protocols run in end systems sending side: converts msgs from app process into transport-layer pkts (segments in Internet term), passes them to net layer (possibly) break app msgs into small chunks, and add headers receiving side: processes segments from net layer, making them available to app more than one transport protocol available to apps Internet: TCP and UDP

3-3

Relationship bw Transport and Network layers
transport layer provides logical comm bw processes, whereas net layer provides logical comm bw hosts Household analogy kids in one household (A) write letters to kids in another household (B) Ann in A and Bill in B collect/distribute mail from/to other kids analogies letters in envelopes ~ app messages kids ~ processes houses ~ hosts Ann and Bill ~ transport protocol not involved in delivering mail bw mail centers Susan-Harvey, substituting Ann-Bill, may provide diff service services (e.g., delay and bw guarantees) clearly constrained by the service the postal service provides certain service (e.g., reliable, secure) can be offered even when postal service doesn’t offer the corresponding service postal service ~ net layer protocol
3-4

Overview of Transport-layer in the Internet
IP (Internet Protocol) provides best-effort delivery service makes “best-effort” to deliver segments, but no guarantees : no guarantee on orderly delivery, integrity of data in segments ⇒ unreliable service User Datagram Protocol (UDP) : provides an unreliable connectionless service, no-frills extension of IP service transport-layer multiplexing and demultiplexing : extend IP’s host-to-host delivery to process-to-process delivery integrity checking by including error-detection fields in segment header Transmission Control Protocol (TCP) : provides a reliable connection-oriented service with several additional services to app reliable data transfer : correct and in-order delivery by using flow control and error control (seq #, ack, timers) connection setup congestion control
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Chap.3 Transport Layer
Introduction and Transport-Layer Services Multiplexing and Demultiplexing Connectionless Transport: UDP Principle of Reliable Data Transfer Connection-Oriented Transport: TCP Principles of Congestion Control TCP Congestion Control

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Multiplexing and Demultiplexing
a process can have one or more sockets; each socket having a unique id multiplexing at sending host : Ann’s job in household analogy gathering data chunks at sources from diff sockets encapsulating each chunk with header info to create segments passing segments to net layer demultiplexing at receiving host : Bill’s job in household analogy delivering data in a seg to the correct socket

3-7

How Demultiplexing Works
host receives IP datagrams each datagram has src and dst IP addrs each datagram carries a transport-layer seg each seg has src and dst port #s well-known port #s : reserved for well-known app protocols, ranging 0 ~ 1023 : HTTP(80), FTP(21), SMTP(25) , DNS(53) other #s : can be used for user apps IP addrs and port #s used to direct seg to appropriate socket

3-8

Connectionless Multiplexing and Demultiplexing
creating UDP socket DatagramSocket mySocket1 = new DatagramSocket(); transport layer automatically assigns a port # to the socket, in the range 1024~65535 not currently used by other UDP ports DatagramSocket mySocket2 = new DatagramSocket(19157); app assigns a specific port # 19157 to the UDP socket typically, the port # in the client side is automatically assigned, whereas the server side assigns a specific port # When a host receives UDP seg, it checks dst port # in the seg and directs the seg to the socket with that port # UDP socket identified by 2-tuple : (dst IP addr, dst port #) IP datagrams with diff src IP addrs and/or src port #s are directed to the same socket src port addr is used as dst port addr in return seg

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Connection-Oriented Mux/Dumux (1)
TCP socket identified by 4-tuple (src IP addr, src port #, dst IP addr, dst port #) demultiplexing at receiving host 4-tuple used to direct seg to appropriate socket TCP segs with diff src IP addrs or src IP port #s are directed to two diff sockets (except TCP seg carrying connestablishment request) server host may support many simultaneous TCP sockets each socket identified by its own 4-tuple

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Connection-Oriented Mux/Dumux (2)

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Connection-Oriented Mux/Demux : Threaded Server
Today’s high-performing Web server uses only one process, but creating a new thread with a new conn for each new client conn connection sockets may be attached to the same process

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Chap.3 Transport Layer
Introduction and Transport-Layer Services Multiplexing and Demultiplexing Connectionless Transport: UDP UDP Segment Structure UDP Checksum Principle of Reliable Data Transfer Connection-Oriented Transport: TCP Principles of Congestion Control TCP Congestion Control

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User Datagram Protocol (UDP) [RFC 768]
no-frills, bare bones transport protocol : adds nothing to IP but, multiplexing/demultiplexing : src and dst port #s (light) error checking features of UDP unreliable best-effort service : no guarantee on correct delivery UDP segments may be lost and delivered out of order to app connectionless : no handshaking bw UDP sender and receiver Q: Isn’t TCP always preferable to UDP? A: No simple, but suitable to certain apps such as real-time apps stringent to delay, but tolerable to some data loss no conn establishment ⇒ no additional notable delay simple ⇒ no conn state, including send/receive buffers, congestion-control parameters, seq and ack # parameters small pkt header overhead : 8 bytes compared to 20 bytes in TCP
3-14

Popular Internet Apps and Their Protocols

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Controversy on UDP
UDP is lack of congestion control and reliable data transfer when many users starts streaming high-bit rate video, packet overflow at routers, resulting in high loss rates for UDP packets decrease TCP sending rate ⇒ adaptive congestion control, forcing all sources including UDP sources, required in particular streaming multimedia apps build reliability directly into app (e.g., adds ack/rexmission) many of today’s proprietary streaming apps run over UDP, but builds ack and rexmission into app in order to reduce pkt loss nontrivial, but can avoid xmission-rate constraint imposed by TCP’s congestion control mechanism

3-16

UDP Segment Structure
Source port #, dst port # : used for multiplexing/demultiplexing Length : length of UDP seg including header, in bytes Checksum : to detect errors (i.e., bits altered) on an end-end basis error source : noise in the links or while store in a router some link-layer protocol may not provide error checking

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UDP Checksum Calculation (1) : Sender
sum all of 16-bit words in segment in a row, with two words for each calculation with overflow wrapped around take 1’s complement of the sum; the result is the checksum value (ex) three 16-bit words
0110011001100000 0101010101010101 1000111100001100 0110011001100000 0101010101010101 1011101110110101

sum of first two words

adding third word

1011101110110101 1000111100001100 10100101011000001 1 wrapped around 0100101011000010 checksum value : 1011010100111101 1’s complement
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UDP Checksum Calculation (2) : Receiver
add all 16-bit words including checksum, and decide no error detected, if the result is 1111111111111111 error detected, otherwise nonetheless the decision is not perfect : error may actually have taken place even when no error detection is decided UDP is not responsible for recovering from error reaction to detecting errors depends on implementations simply discard damaged seg, or pass damaged seg to app with warning

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Chap.3 Transport Layer
Introduction and Transport-Layer Services Multiplexing and Demultiplexing Connectionless Transport: UDP Principle of Reliable Data Transfer Building a Reliable Data Transfer Protocol Pipelined Reliable Data Transfer Protocol Go-Back-N (GBN) Selective Repeat (SR) Connection-Oriented Transport: TCP Principles of Congestion Control TCP Congestion Control
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Reliable Data Transfer : Service Model and Implementation
reliable data transfer : no corruption, no loss, and in-order delivery of central importance to networking : not only at transport layer, but also at link layer and app layer
rdt_send() : deliver_data() : called by rdt

called from app

to deliver data to app

udt_send() : called by

rdt to sen pkt over unreliable channel

rdt_rcv() : called from

channel upon pkt arrival
3-21

Reliable Data Transfer: Implementation Consideration
characteristics of unreliable channel determines the complexity of reliable data transfer protocol We will incrementally develop sender and receiver sides of rdt protocol, considering increasingly complex model of underlying channel consider only unidirectional data transfer for simplicity purpose but, control packet is sent back and forth use finite state machines (FSM) to specify sender, receiver
dashed arrow : initial state state: next state uniquely state determined by event event causing state transition actions taken on state transition Λ : no event or no action

1

state 2
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rdt1.0 : Perfectly Reliable Channel
Assumptions of underlying channel perfectly reliable : no bit errors, no loss of packets separate FSMs for sender and receiver sender sends data into underlying channel receiver read data from underlying channel

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rdt2.0 : Channel with Errors
New assumptions of underlying channel may be corrupted when transmitted, propagated, or buffered no loss and in-order delivery Automatic Repeat reQuest (ARQ) protocols error detection : extra bits placed in checksum field receiver feedback : ACK/NAK pkt explicitly sent back to sender ACK (positive acknowledgement) : when pkt received OK NAK (negative acknowledgement) : when pkt received in error rexmission : sender rexmits pkt on receipt of NAK

3-24

rdt2.0 : not Corrupted

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rdt2.0 : Corrupted

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rdt2.0 : Fatal Flaw
Q: How to recover from errors in ACK or NAK pkts? minimally, need to add checksum bits to ACK/NAK pkts possible solutions repeated requests from sender/receiver for a garbled ACK and NAK : hard to find a clue to way out add enough checksum bits for correction : not applicable for lost pkt simply resend the pkt when receiving a garbled ACK or NAK ⇒ incurs possible duplicate at receiver receiver doesn’t know whether it is a new pkt or a rexmission (i.e., a duplicate pkt) handling duplicates : add a new field (seq # field) to the packet sender puts a seq # into this field, and receiver discards duplicate pkt 1-bit seq # suffice for stop-and-stop protocol rdt2.0 is stop-and-wait protocol : sender sends one pkt, then waits for receiver response
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Description of sol 1 of Fatal Flaw of rdt2.0

A dict ates s ometh in g
“p

to B

A didn’t understand

B re What did yo u

r ok o l ie s p

” peat e re leas

say? b ut

o did y hat W

corrup ted B has no idea whether it is part of dictation ? or request for repetition of last reply u say

3-28

rdt2.1 : Employing Seq # - Sender

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rdt2.1 : Employing Seq # - Receiver

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rdt2.1 : Discussion
sender seq # added to pkt two seq #’s (0,1) will suffice must check if received ACK/NAK corrupted twice as many states state must remember whether current pkt has seq # of 0 or 1 receiver must check if received pkt is duplicate state indicates whether 0 or 1 is expected pkt seq # receiver cannot know if its last ACK/NAK received OK at sender

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rdt2.2 : NAK-free
accomplish the same effect as a NAK, by sending an ACK for the last correctly received pkt receiver must explicitly include seq # of pkt being ACKed sender that receives two ACKs (i.e., duplicate ACKs) knows that receiver didn’t correctly receive the pkt following the pkt being acked twice, thus rexmits the latter

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rdt2.2 : NAK-free (Sender)

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rdt2.2 : NAK-free (Receiver)

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rdt3.0 : Channel with Errors and Loss
new assumptions of underlying channels : can lose pkts (data or ACKs) Q : how to detect pkt loss and what to do when pkt loss occurs checksum, seq #, ACKs, rexmissions are of help, but not enough approaches sender waits proper amount of time (at least round-trip delay + processing time at receiver) to convince itself of pkt loss rexmits the pkt if ACK not received within this time if a pkt (or its ACK) just overly delayed, sender may rexmit the pkt even though it has not been lost but, seq # handles the possibility of duplicate pkts implementation countdown timer set appropriately starts each time pkt is sent rexmit pkt when the timer is expired
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rdt3.0 : Channel with Errors & Loss (Sender)

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rdt3.0 : Channel with Errors & Loss – Operation (1)

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rdt3.0 : Channel with Errors & Loss – Operation (2)

3-38

Performance of rdt3.0 (Stop-and-Wait Protocol)

assumption : ignore xmission time of ACK pkt (which extremely small) and processing time of pkt at the sender and receiver sender utilization Usender : frac. of time sender is busy sending into ch ex) 1 Gbps link, 30 ms RTT, 1 KB packet very poor! net protocol limits the capabilities provided by underlying net HW
3-39

U sender =

ttrans 0.008 8, 000 bits/packet = ≈ 0.00027 ; ttrans = L R = = 0.008 ms 9 RTT + ttrans 30 + 0.008 10 bits/sec

Pipelining
sends multiple pkts without waiting for acks range of seq #s is increased buffering at sender and/or receiver required sender : pkts that have been xmitted by not yet acked receiver : pkts correctly receiver

sender is assumed to send 3 pkts before being acked U sender =

3ttrans 0.024 = ≈ 0.0008 : essentially tripled RTT + ttrans 30.008
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two generic forms of pipelined protocols: go-Back-N, selective repeat

Go-Back-N (GBN) Protocol
sender’s view of seq #s in GBN

window size N : # of pkts allowed to send without waiting for ACK GBN often referred to as sliding window protocol pkt’s seq # : carried in a k-bit field in pkt header range of seq # : [0, 2k-1] with modulo 2k arithmetic events at GBN sender invocation from above : before sending, check if window isn’t full receipt of an ACK : cumulative ack - ack with seq # n indicates all pkts with a seq up to and including n have been correctly received timeout : resend all pkts previously xmitted but not yet acked drawback of GBN : when widow size and bw-delay product are large, a single pkt error cause a large # of unnecessarily rexmissions
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Go-Back-N (GBN) Protocol : Sender

a single timer : for the oldest xmitted but not yet acked pkt upon receipt of an ACK, if there are no outstanding unacked pkts, the timer is stopped still xmitted but not yet 3-42 acked pkts, the timer is restarted

Go-Back-N (GBN) Protocol : Receiver
when pkt with seq # n is received correctly and in-order, receiver sends an ACK for pkt n and delivers data portion to upper layer receiver discards out-of-order pkts and resends an ACK for the most recently received in-order pkt simple receiver buffering : needn’t buffer any out-of-order pkts only info needed : seq # of next in-order pkt, expectedseqnum

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Go-Back-N (GBN) Protocol : Operation
window size = 4

3-44

Selective Repeat (SR) Protocol
sender rexmits only pkts for which ACK not received ⇒ avoid unnecessary rexmission receiver individually acks correctly received pkts regardless of their order out-of-order pkts are buffered until missing pkts are received

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SR Protocol : Sender/Receiver Events and Actions
sender data from above : if next available seq # is in window, send pkt timeout(n) : resend pkt n, restart timer each pkt has its own (logical) timer ACK(n) in [sendbase,sendbase+N] mark pkt n as received if n is equal to send_base, window base is moved forward to next unacked pkt, and xmit unxmitted pkts in advanced window receiver pkt n in [rcvbase, rcvbase+N-1] correctly received : send ACK(n) if not previously received, it is buffered if n is equal to rcv_base, this pkt and previously buffered in-order pkts are delivered to upper layer, and receive window moved forward by the # of pkts delivered to upper layer pkt n in [rcvbase-N,rcvbase-1] correctly received an ACK generated even though previously acked if not acks, sender’s window may never move forward; for example, ack for send_base pkt in Figure 3.23 otherwise : ignore 3-46

SR Operation

3-47

Max. Window Size
stop-and-wait protocol window size N ≤ 2k-1 (k: # of seq field), not 2k, why? ex) k=2 ⇒ seq #s : 0, 1, 2, 3; max N = 3 SR protocol scenarios (a) : all acks are lost incorrectly sends duplicate as new (b) : all acks received correctly, but pkt 3 is lost receiver can’t distinguish xmission of pkt 0 in (b) from rexmission of pkt 0 in (a) further consideration on scenario (a) A rexmits pkt 0; B receives and buffer it B sends piggybacked ack for pkt 2 that is already acked but lost A advanced window 3 0 1, and sends pkt 3 B receives pkt 3, and delivers pkt 0 (no good!) in buffer and pkt 3 to upper layer wayout : avoid overlapping of SR windows N ≤ 2k-1, k: # of bits in seq field
A B

3-48

rdt : Comment on Packet Reordering
since seq #s are reused, old copies of a pkt with a seq/ack # of x can appear, even though neither sender’s nor receiver’s window contains x use of max pkt lifetime : constrain pkt to live in the net ~ 3 minutes in TCP for high-speed net

3-49

Summary of rdt Mechanisms

3-50

Chap.3 Transport Layer
Introduction and Transport-Layer Services Multiplexing and Demultiplexing Connectionless Transport: UDP Principle of Reliable Data Transfer Connection-Oriented Transport: TCP TCP Connection TCP Segment Structure Round-Trip Time Estimation and Timeout Reliable Data Transfer Flow Control TCP Connection Management Principles of Congestion Control TCP Congestion Control
3-51

TCP Connection
two processes established connection via 3-way handshake before sending data, and initialize TCP variables full duplex : bi-directional flow bw processes in the same conn point-to-point : bw one sender and one receiver multicasting is not possible with TCP a stream of data passes through a socket into send buffer TCP grab chunks of data from send buffer max seg size (MSS) : max amount of app-layer data in seg set based on Path MTU of link-layer typically, 1,460 bytes, 536 bytes, or 512 bytes each side of conn has send buffer and receive buffer

3-52

TCP Segment Structure

for reliable data xfer count in bytes, not pkts 4-bit # counting in 32-bit words for error detection for flow control, # of bytes receiver willing to receive typically, empty - time-stamping - mss, window scaling factor negotiation, etc.

• ACK : indicates value in ack field is valid • SYN, RST, FIN : used for connection setup and teardown • PSH : receiver should pass data to upper layer immediately • URG : indicates there is an urgent data in the seg marked by sending-side upper layer - urgent data pointer indicates the last bytes of urgent data - generally, PSH and URG are not used 3-53

Seq Numbers and Ack Numbers
seq # : 1st byte in seg over xmitted bytes stream, not over series of xmitted segs TCP implicitly number each byte in data stream

initial seq # is chosen randomly rather than set 0, why? ack # : seq # of next byte expected from other side cumulative ACK Q : how to handle out-of-order segs at receiver? discard or buffer waiting for missing bytes to fill in the gaps TCP leaves the decision up to implementation, but the latter is chosen in practice
3-54

Telnet : Case Study of Seq and Ack Numbers
each ch typed by A is echoed back by B and displayed on A’s screen

ACK piggybacked on B-to-A data seg

explicit ACK with no data

3-55

Estimating Round-Trip Time (RTT)
clearly, TCP timeout value > RTT Q : How much larger? How to estimate RTT? Each seg exploited in estimating RTT? … estimating RTT SampleRTT : time measured from seg xmission until ACK receipt measured not for every seg xmitted, but for one of xmitted segs approximately once every RTT rexmitted segs are not considered in measurements fluctuates from seg to seg : atypical ⇒ needs some sort of avg Exponential Weighted Moving Average (EWMA) of RTT avg several recent measurements, not just current SampleRTT EstimatedRTT = (1 - α)⋅EstimatedRTT + α⋅SampleRTT recommended value of α : 0.125 more weight on recent samples than on old samples weight of a given sampleRTT decays exponentially fast as updates proceed
3-56

RTT Samples and RTT Estimates
variations in the Sample RTT are smoothed out in Estimated RTT

3-57

Retransmission Timeout Interval
DevRTT, variation of RTT : an estimate of how much SampleRTT deviates from EstimatedRTT DevRTT = (1-β)⋅DevRTT + β⋅|SampleRTT−EstimatedRTT| large (or small) when there is a lot of (or little) fluctuation recommended value of β : 0.25 TCP’s timeout interval should be larger, or unnecessarily rexmit! but, if too much larger, TCP wouldn’t quickly rexmit, leading to large data transfer delay thus, timeout interval should be EstimatedRTT plus some safety margin that varies as a function of fluctuation in SampleRTT TimeoutInterval = EstimatedRTT + 4⋅DevRTT

3-58

TCP Reliable Data Transfer
reliable data transfer service on top of IP’s unreliable service seq # : to identify lost and duplicate segs cumulative ack : positive ACK (i.e, NAK-free) timer a single rexmission timer is recommended [RFC 2988], even if there are multiple xmitted but not yet acked segs rexmissions triggered by when timed out 3 duplicate acks at sender : fast rexmit in certain versions We’ll discuss TCP rdt in two incremental steps highly simplified description : only timeouts considered more subtle description : duplicate acks as well as timeouts considered in both cases, error and flow control are not taken into account

3-59

Simplified TCP Sender
seq # is byte-stream # of the first data byte in seg

TimeoutInterval = EstimatedRTT + 4⋅DevRTT

some not-yet-acked segs are acked move window forward

3-60

TCP Retransmission Scenarios

SendBase=120 SendBase=100 SendBase=120 SendBase=100 SendBase=120

rexmission due to a lost ack

segment 100 not rexmitted

cumulative ack avoids rexmission of first seg
3-61

TCP Modifications : Doubling Timeout Interval
at each timeout, TCP rexmits and set next timeout interval to twice the previous value ⇒ timeout intervals grow exponentially after each rexmission but, for the other events (i.e., data received from app and ACK received) timeout interval is derived from most recent values of EstimatedRTT and DevRTT

3-62

TCP ACK Gen Recommendation [RFC 1122, 2581]

timeout period can be relatively long ⇒ may increase e-t-e delay when sending a large # of segs back to back (such as a large file), if one seg is lost, there will be likely many back-to-back ACKs for it

3-63

TCP Modifications : TCP Fast Retransmit
TCP Fast Retransmit : rexmits a (missing) seg before its timer expiration, if TCP sender receives 3 duplicate ACKs
if (y > SendBase) { // event: ACK received, with ACK field value of y SendBase = y if (there are currently not-yetacked segs) start timer } else { // a duplicate ACK for already ACKed segment increment count of dup ACKs received for y if (count of dup ACKs received for y = 3) // TCP fast retransmit resend seg with seq # y }

3-64

Is TCP Go-Back-N or Selective Repeat?
similarity of TCP with Go-Back-N TCP : cumulative ack for the last correctively received, in-order seg cumulative and correctly received but out-of-order segs are not individually acked ⇒ TCP sender need only maintain SendBase and NextSeqNum differences bw TCP and Go-Back-N : many TCP implementations buffer correctly received but out-of-order segs rather than discard also, suppose a seq of segs 1, 2, … N, are received correctively in-order, ACK(n), n < N, gets lost, and remaining N-1 acks arrive at sender before their respective timeouts TCP rexmits at most one seg, i.e., seg n, instead of pkts, n, n+1, …, N TCP wouldn’t even rexmit seg n if ACK(n+1) arrived before timeout for seg n a modification to TCP in [RFC 2018] : selective acknowledgement TCP receiver acks out-of-order segs selectively rather than cumulatively when combined with selective rexmission - skipping segs selectively acked by receiver – TCP looks a lot like generic SR protocol Thus, TCP can be categorized as a hybrid of GBN and SR protocols
3-65

Flow Control : Goal
receiving app may not read data in rcv buffer as quickly as supposed to be it may be busy with some other task may relatively slow at reading data, leading to overflowing receiver’s buffer by too much data too quickly sent by sender flow control : a speed-matching service, matching sending rate against reading rate of receiving app goal : eliminate possibility of sender overflowing receiver buffer (note) to make the discussion simple, TCP receiver is assumed to discard out-of-order segs

3-66

Flow Control : How It Works?
RevBuffer : size of buffer space allocated to a conn RcvWindow : amount of free buffer space at rcv’s buffer initial value of RcvWindow = RevBuffer

LastByteRcvd, LastByteRead : variables at receiver LastByteSent, LastByteAcked : variables at sender

at receiver not to overflow : LastByteRcvd – LastByteRead ≤ RcvBuffer LastByteRcvd – LastByteRead : # of bytes received not yet read RevWindow advertising : RcvWindow placed in receive window field in every seg sent to sender RcvWindow = RcvBuffer - [LastByteRcvd - LastByteRead] at sender : limits unacked # of bytes to RcvWindow LastByteSent – LastByteAcked ≤ RcvWindow LastByteSent – LastByteAcked : # of byte sent but not yet acked
3-67

Flow Control : Avoiding Sender Blocking
suppose A is sending to B, B’s rcv buffer becomes full so that RcvWindow = 0, and after advertising RcvWindow = 0 to A, B has nothing to send to A note that TCP at B sends a seg only if it has data or ack to send there is no way for B to inform A of some space having opened up in B’s rcv buffer ⇒ A is blocked, and can’t xmit any more! wayout : A continue to send segs with one data byte when RcvWindow = 0, which will be acked eventually, the buffer will begin to empty and ack will contain a nonzero RcvWindow value

3-68

TCP Connection Management : Establishment
3-way handshake 1. client sends SYN seg to server contains no app data randomly select client initial seq # 2. server replies with SYNACK seg server allocates buffers and variables to the connection contains no app data randomly select server initial seq # 3. client replies with ACK seg client allocates buffers and variables to the connection may contain data

SYN s egmen t
t men seg

CK YNA S

ACK se

gmen t

3-69

TCP Connection Management : Termination
Either of client or server can end the TCP connection duration of TIME_WAIT period : implementation dependent typically, 30 secs, 1 min, 2 mins RST seg : seg with RST flag set to 1 sent when receiving a TCP seg whose dst port # or src IP addr is not matched with on-going one

3-70

TCP State Transition : Client
Socket clientSocket = new Socket("hostname","port#");

3-71

TCP State Transition : Server
ServerSocket welcomeSocket = new ServerSocket(port#)

Socket connectionSocket = welcomeSocket.accept(); 3-72

Chap.3 Transport Layer
Introduction and Transport-Layer Services Multiplexing and Demultiplexing Connectionless Transport: UDP Principle of Reliable Data Transfer Connection-Oriented Transport: TCP Principles of Congestion Control The Causes and the Costs of Congestion Approaches to Congestion Control Network-Assisted Congestion-Control Example for ATM AVR TCP Congestion Control

3-73

Preliminary of Congestion Control
pkt loss (at least, perceived by sender) results from overflowing of router buffers as the net becomes congested rexmission treats a symptom, but not the cause, of net congestion cause of net congestion : too many sources attempting to send data at too high a rate basic idea of wayout : throttle senders in face of net congestion what’s different from flow control? ranked high in top-10 list of networking problem

3-74

Causes and Costs of Congestion : Scenario 1
assumptions no error control, flow control, and congestion control host A and B send data at an avg rate of λin bytes/sec, respectively share a router with outgoing link capacity of R and infinite buffer space ignore additional header info (transport-layer and lower-layer)

cost of congested net : avg delay grows unboundedly large as arrival rate nears link capacity
3-75

Causes and Costs of Congestion : Scenario 2 (1)
assumptions one finite buffer space each host with same λin, retransmit dropped packets

3-76

Causes and Costs of Congestion : Scenario 2 (2)

case a

case b

case c

case a (unrealistic) : host A can somehow determine if router buffer is free, and send a pkt when buffer is free no loss, thus no rexmission ⇒ λ’in= λin case b : a pkt is known for certain to be dropped R/3 : original data, R/6 : rexmitted data cost of congested net : sender must rexmit dropped pkt case c : premature timeout for each pkt ⇒ rexmit each pkt twice cost of congested net : unneeded rexmissions waste link bw

3-77

Causes and Costs of Congestion : Scenario 3
assumptions 4 routers, each with finite buffer space and link capacity of R each of 4 hosts has same λin, rexmits over 2-hop paths • consider A→C conn • a pkt dropped at R2 (due to high λin from B) wastes the work done by R1

cost of congested net : a pkt dropped at some point wastes the xmission capacity up to that point 3-78

Two Broad Approaches to Congestion Control
end-end congestion control no explicit support (by feedback) from net layer congestion inferred by end-system based on observed net behavior, e.g., pkt loss and delay approach taken by TCP congestion is inferred by TCP seg loss indicated by timeout or triple duplicate acks network-assisted congestion control routers provide explicit feedback to end systems regarding congestion state in the net single bit indication SNA, DECnet, TCP/IP ECN [RFC2481], ATM AVR congestion control explicit rate : the rate router can support on its outgoing link

3-79

Two Types of Feedback of Congestion Info
direct feedback : from a router to the sender by using choke pkt feedback via receiver router mark/update a field in a pkt flowing forward to indicate congestion upon receipt of the pkt, receiver notifies sender of congestion

3-80

ATM ABR Congestion Control
Asynchronous Transfer Mode (ATM) a virtual-circuit switching architecture info delivered in fixed size cell of 53 bytes each switch on src-to-dst path maintains per-VC state Available Bit Rate (ABR) : an elastic service if net underloaded, use as much as available bandwidth if net congested, sender rate is throttled to predetermined min guaranteed rate Resource Management (RM) cells interspersed with data cells, conveying congestion-related info rate of RM cell interspersion : tunable parameter default value : one every 32 data cells provides both feedback-via-receiver and direct feedback sent by src flowing thru switches to dst, and back to src switch possibly generate RM cell itself, and send directly to src
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Mechanisms of Congestion Indication in ATM AVR

Explicit Forward Congestion Indication (EFCI) bit EFCI bit in a data cell is set to 1 at congested switch if a data cell preceding RM cell has EFCI set, dst sets CI bit of RM cell, and sends it back to src CI (Congestion Indication) and NI (No Increase) bits set by congested switch, NI/CI bit for mild/severe congestion dst sends the RM cell back to src with CI and NI bits intact Explicit Rate (ER) : two-byte field in RM cell congested switch may lower ER value in a passing RM cell 3-82 when retuned back to src, it contains max supportable rate on the path

Chap.3 Transport Layer
Introduction and Transport-Layer Services Multiplexing and Demultiplexing Connectionless Transport : UDP Principle of Reliable Data Transfer Connection-Oriented Transport : TCP Principles of Congestion Control TCP Congestion Control Fairness TCP Delay Modeling

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Preliminary of TCP Congestion Control (1)
basic idea of TCP congestion control : limit sending rate based on the network congestion perceived by sender increase/reduce sending rate when sender perceives little/∗ congestion along the path bw itself and dst to keep the description concrete, sending a large file is assumed How does sender limit sending rate?
LastByteSent - LastByteAcked ≤ min{CongWin, RcvWindow} (1) CongWin : a variable limiting sending rate due to perceive

congestion henceforth, RcvWindow constraint ignored in order to focus on congestion control (1) limits the amount of unacked data, thus the sending rate consider conn for which loss and xmission delay are negligible CongWin then, sending rate ≈ RTT
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Preliminary of TCP Congestion Control (2)
How does sender perceive congestion on path bw itself and dst? a timeout or the receipt of three duplicate ACKs TCP is self-clocking : acks are used to trigger its increase on cong window size, thus the sending rate consider an optimistic case of cong-free, in which acks are taken as an indication that seg are successfully delivered to dst if acks arrive at a slow/high rate, cong window is increased more slowly/quickly How to regulate sending rate as a function of perceived congestion? TCP congestion control algorithms, consisting of 3 components additive-increase, multiplicative-decrease (AIMD) AIMD is a big-picture description; details are more complicated slow start reaction to timeout events
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Additive-Increase, Mulitplicative-Decrease
multiplicative decrease : cut CongWin in half down to 1 MSS when detecting a loss additive increase: increase CongWin by 1 MSS every RTT until a loss detected (i.e., when perceiving e-t-e path is congestion-free) commonly, accomplished by increasing CongWin by MSS⋅(MSS/CongWin) bytes for each receipt of new ack ex) MSS=1,460 bytes, ConWin=14,600 bytes ⇒ 10 segs sent within RTT congestion avoidance : linear increase phase of TCP cong control saw-toothed pattern of CongWin
an ACK for a seg increases CongWin by 1/10⋅MSS, thus after ack for all 10 segs (thus, for one RTT) CongWin is increased by MSS

3-86

TCP Slow Start
When a TCP conn begins, CongWin is typically initialized to 1 MSS ⇒ initial rate ≈ MSS/RTT ex) MSS = 500 bytes, RTT = 200 msec ⇒ initial sending rate : only about 20 kbps linear increase at init. phase results in a waste of bw, considering available bw may be >> MSS/RTT desirable to quickly ramp up to some respectable rate slow start (SS) : during initial phase, increase sending rate exponentially fast by doubling CongWin every RTT until a loss occurs achieved by increasing CongWin by 1 MSS for receipt of ack
3-87

Reaction to Congestion
Q: When does CongWin switch from exponential increase to linear increase? A: when CongWin is reached to Threshold Threshold : a variable set to a half of CongWin just before a loss initially set large, typically 65 Kbytes, so that it has no initial effect maintained until the next loss TCP Tahoe, early version of TCP CongWin is cut to 1 MSS both for a timeout and for 3 duplicate acks Jacobson’s algorithm [Jacobson 1988] TCP Reno [RFC2581, Stevens ’94] : reaction to loss depends on loss type for 3 duplicate acks receipt : CongWin is cut in half, then grows linearly for a timeout event : CongWin is set to 1 MSS (SS phase), then grows exponentially to a Threshold, then grows linearly (CA phase) idea : 3 dup acks anyhow indicates capability of delivering some pkts TCP Reno cancels SS phase after a triple duplicate ack : fast recovery many variations of TCP Reno [RFC 3782, RFC 2018] TCP Vegas [Brakmo 1995] idea : early warning - detect congestion in routers before pkt loss occurs when this imminent pkt loss, predicted by observing RTT, is detected, CongWin is lowered linearly; the longer the RTT, the greater the congestion 3-88

TCP Congestion Control Algorithms

• initial value of Threshold = 8 MSS • triple duplicate acks just after 8th round

3-89

TCP Reno Congestion Control Algorithm
[RFC 2581, Stevens 1994]

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Steady-State Behavior of a TCP Connection
Consider a highly simplified macroscopic model for steady-state behavior of TCP SS phases ignored since they are typically very short Letting W be the window size when a loss event occurs, RTT and W are assumed to be approximately constant during a conn Q : What’s avg throughput of a long-lived TCP conn as a function of window size and RTT? 0.75 ⋅ W A : avg throughput of a TCP connection = (2) RTT a pkt is dropped when the rate increases to W/RTT then the rate is cut in half and linearly increases by MSS/RTT every RTT until it again reaches W/RTT this process repeats over and over again

3-91

TCP Futures
TCP congestion control has evolved over the years and continue to evolve [RFC 2581] : a summary as of the late 1990s [Floyd 2001] : some recent developments traditional scheme is not necessarily good for today’s HTTP-dominated Internet or for a future Internet service high bandwidth delay product ex) Consider a high-speed TCP conn with 1500-byte segments, 100ms RTT, and want to achieve 10 Gbps throughput through this conn to meet this, from (2) required window size is
RTT 0.1 sec 1 107 10 ⋅ tput = ⋅ 10 bits/sec ⋅ = ≈ 111,111 segs W= 0.75 0.75 1,500 × 8 bits/seg 90

this is a lot of segs, so that there is high possibility of errors, leading us to derive a relationship bw throughput and error rate [prob. P39]

avg throughput of a TCP conn =

1.22 ⋅ MSS RTT L

⇒ L = 2⋅10-10, i.e., one loss for every 5 ⋅10-9segs : unattainably low! ⇒ new vers of TCP required for high-speed environments [RFC 3649, Jin 2004]
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TCP Fairness (1)
suppose K TCP conns pass though a bottleneck link bw of R, with each conn sending a large file ⇒ avg xmission rate of each conn is approximately R/K TCP congestion control is fair : each conn gets an equal share of bottleneck link’s bw among competing TCP conns consider a link of R shared by two TCP conn, with idealized assumptions same MSS and RTT, sending a large amount of data, operating in CA mode (AIMD) at all times, i.e., ignore SS phase

3-93

TCP Fairness (2)
bw realized by two conns fluctuates along equal bw share line, regardless of their initial rates in practice, RTT value differs from conn to conn conns with a smaller RTT grab the available bw more quickly (i.e., open their cong window faster), thus get higher throughput than those conns with larger RTTs

ideal operating point

D B

loss occurs

C A

CA phase

3-94

Some other Fairness Issues
Fairness and UDP multimedia apps, e.g., Internet phone and video conferencing do not want their rate throttled even if net is congested thus runs over UDP rather than TCP, pumping audio/video at const rate, and occasionally lose pkt rather than reducing rate when congested ⇒ UDP sources may crowd out TCP traffic research issue : TCP-friendly cong control goal : let UDP traffic behave fairly, thus prevent the Internet from flooding Fairness and parallel TCP connections a session can open multiple parallel TCP conn’s bw C/S, thus gets a large portion of bw in a congested link a Web browser to xfer multiple objects in a page ex) a link of rate R supporting 9 ongoing C/S apps a new app, asking for 1 TCP conn, gets an equal share of R/10 a new app, asking for 11 TCP conns, gets an unfair rate of R/2
3-95

TCP Delay Modeling
We’d compute the time for TCP to send an object for some simple models latency : defined as the time from when a client initiate a TCP conn until the time at which it receives the requested object assumptions : made in order not to obscure the central issues simple one-link net of rate R bps amount of data sender can xmit is limited solely by cong window pkts are neither lost or corrupted, thus no rexmission all protocol header overheads : ignored object consist of an integer # of MSS O: object size [bits], S : seg size [bits] (e.g., 536 bits) xmission time for segs including control info : ignored initial threshold of TCP cong control scheme is so large as not to be attained by cong window without cong window constraint : the latency is 2⋅RTT+O/R clearly, SS procedure, dynamic cong window increase this minimal latency

3-96

Static Congestion Window (1)
W : a positive integer, denoting a fixed-size static congestion window upon receipt of rqst, server immediately sends W segs back to back to client, then one seg for each ack from client 1st case : WS/R > RTT+S/R ack for 1st seg in 1st window received before sending 1st window’s worth of segs server xmit segs continuously until entire object is xmitted thus, the latency is 2⋅RTT+O/R

W=4

3-97

Static Congestion Window (2)
2nd case : WS/R < RTT+S/R ack for 1st seg in 1st window received after sending 1st window’s worth of segs latency = setup time + time for xmitting object + sum of times in idle state let K : # of windows covering object K = O/WS or ⎡K⎤ if K is not an integer # of times being in idle state = K-1 duration of server being in idle state S/R+RTT-WS/R thus, the latency is 2⋅RTT+O/R+(K-1)[S/R+RTT-WS/R]+ where [x]+ = max(x,0)
transmitting state idle state
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W=2

Dynamic Congestion Window (1)
cong window grows according to slow start, i.e., doubled every RTT O/S : # of segs in the object # of segs in kth window : 2k-1 K : # of windows covering object O⎫ ⎧ K = min ⎨k : 20 + 21 + + 2k −1 ≥ ⎬ S⎭ ⎩ O⎫ ⎧ = min ⎨k : 2k −1 − 1 ≥ ⎬ S⎭ ⎩ ⎧ ⎛O ⎞⎫ = min ⎨k : k ≥ log2 ⎜ + 1 ⎟ ⎬ ⎝S ⎠⎭ ⎩ ⎡ ⎛O ⎞⎤ = ⎢log2 ⎜ + 1 ⎟ ⎥ ⎝S ⎠⎥ ⎢ xmission time of kth window = (S/R)2k-1 duration in idle state of kth window =[S/R+RTT-2k-1(S/R)]+
O/S=15 K=4 Q=2 P=min{Q,K-1}=2

3-99

Dynamic Congestion Window (2)
latency = setup time + time for xmitting object + Σ times in idle state
O K −1 ⎡ S S⎤ latency = 2 ⋅ RTT + + ∑ ⎢ + RTT − 2k −1 ⎥ R k =1 ⎣ R R⎦
+

(3)

Q : # of times server being idle if object were of infinite size
Q = max k : S S RTT ⎫ ⎧ + RTT − 2k −1 ≥ 0 = max ⎨k : 2k −1 ≤ 1 + ⎬ R R S /R ⎭ ⎩ ⎧ RTT ⎞ ⎫ ⎢ RTT ⎞ ⎥ ⎛ ⎛ = max ⎨k : k ≤ log2 ⎜ 1 + + 1⎬ = ⎢log2 ⎜ 1 + ⎥ +1 S /R ⎟ ⎭ ⎣ S / R ⎟⎦ ⎝ ⎠ ⎝ ⎠ ⎩

{

}

actual # of times server is idle is P=min{Q, K-1}, then (3) becomes
O P ⎡S S⎤ latency = 2 ⋅ RTT + + ∑ ⎢ + RTT − 2k −1 ⎥ R k =1 ⎣ R R⎦ P O ⎡S ⎤ S = 2 ⋅ RTT + + P ⎢ + RTT ⎥ − ∑ 2k −1 R ⎣R ⎦ R k =1 O S ⎡S ⎤ = 2 ⋅ RTT + + P ⎢ + RTT ⎥ − (2P − 1 ) R R ⎣R ⎦

(4)
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Dynamic Congestion Window (3)
comparing TCP latency of (4) with minimal latency
P ⎡(S R ) RTT + 1⎤ − ⎡(2p − 1 ) (S R ) RTT ⎤ latency ⎣ ⎦ ⎣ ⎦ =1+ minimal latency 2 + (O R ) RTT

P + (S R ) RTT ⎡P + 1 − 2p ⎤ P ⎣ ⎦ ≤1+ =1+ 2 + (O R ) RTT 2 + (O R ) RTT
latency contributed by slow start

slow start significantly increase latency when object size is relatively small (implicitly, high xmission rate) and RTT is relatively large this is often the case with the Web See the examples in the text

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HTTP Modeling
Assume Web page consists of 1 base HTML page (of size O bits) M images (each of size O bits) non-persistent HTTP M+1 TCP conns in series response time = 2⋅(M+1)RTT + (M+1)O/R + sum of idle times persistent HTTP 2 RTT to request and receive base HTML file 1 RTT to request and receive M images response time = 3⋅RTT + (M+1)O/R + sum of idle times non-persistent HTTP with X parallel conns suppose M/X is integer 1 TCP conn for base file M/X sets of parallel conns for images response time = 2⋅(M/X + 1)RTT + (M+1)O/R + sum of idle times
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