instructor: christopher cole some slides taken from kurose & ross book it 347: chapter 3...

Instructor: Christopher ColeSome slides taken from Kurose &

Ross book

IT 347: Chapter 3Transport Layer

• Network layer: logical communication between hosts• Transport layer: logical communication between processes

– end-to-end only– Routers, etc. don’t read segments

• Weird analogy of 2 families with Bill and Ann• IP provides a best-effort delivery service

– No guarantees!– unreliable

• Most fundamentally: extend host-to-host delivery to process-to-process delivery

Transport Layer 3-3

Internet transport-layer protocols

• 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

applicationtransportnetworkdata linkphysical

networkdata linkphysical






applicationtransportnetworkdata linkphysical

logical end-end transport

Multiplexing

• Everybody is on sockets– Multiplexing: passing segments to network layer– Demultiplexing: taking network layer and

distributing to sockets– How is it done?

• What are the 2 things that applications need to talk to each other? Source port/IP and destination port/IP

– Server side usually specifically names a port– Client side lets the transport layer automatically

assign a port

More multiplexing

• UDP: two-tuple– Identified by destination IP address and port

number– If two UDP segments have difference source IP

addresses and/or port numbers, but the same destination IP and port number, they will be directed to the same destination process via the same socket

• TCP: four-tuple– Identified by source IP/port and destination IP/port– Two segments with difference source info, but the same

destination info, will be directed to different sockets– TCP keeps the port open as a “welcoming socket”

• Server process creates new socket when connection is created• One server can have many sockets open at a time

• Web server– Spawns a new thread for each connection (How does it know

which collection belongs to who? Source port & IP)– Threads are like lightweight subprocesses– Many threads for one process

UDP

• A transport layer protocol must: provide multiplexing/demultiplexing– That’s all UDP does besides some light error checking– You’re practically talking directly to IP

• UDP process– Take the message from the application– Add source and destination port number fields– Add length & checksum fields– Send the segment to layer 3

• Connectionless (no handshaking)– Why is DNS using UDP?

Advantages of UDP

• Finer application control– Just spit the bits onto the wire. No congestion control,

flow control, etc.• Connectionless

– No extra RTT delay– Doesn’t create buffers, so a UDP server can take more

clients than a TCP server• Small packet overhead

– TCP overhead = 20 bytes– UDP overhead = 8 bytes

The UDP Controversy

• UDP doesn’t play nice– No congestion control

• Say UDP packets flood the lines…– Causes routers to get more congested– TCP sees this, and slows down packet sending– UDP doesn’t

• Only UDP packets end up getting sent

Transport Layer 3-10

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

Why is there a UDP?• no connection establishment

(which can add delay)• simple: no connection state at

sender, receiver• small segment header• no congestion control: UDP can

blast away as fast as desired


UDP: more

• often used for streaming multimedia apps– loss tolerant– rate sensitive

• other UDP uses– DNS– SNMP

• reliable transfer over UDP: add reliability at application layer– application-specific error

recovery!

source port # dest port #

32 bits

Applicationdata (message)

UDP segment format

length checksumLength, in

bytes of UDPsegment,including

header


UDP checksum

Sender:• treat segment contents as

sequence of 16-bit integers• checksum: addition (1’s

complement sum) of segment contents

• sender puts checksum value into UDP checksum field

Receiver:• compute checksum of received

segment• check if computed checksum

equals checksum field value:– NO - error detected– YES - no error detected.

• Throw the bit away OR• Pass it on with a warning

Goal: detect “errors” (e.g., flipped bits) in transmitted segment


Internet Checksum Example• Note

– When adding numbers, a carryout from the most significant bit needs to be added to the result

• Example: add two 16-bit integers

1 1 1 1 0 0 1 1 0 0 1 1 0 0 1 1 01 1 1 0 1 0 1 0 1 0 1 0 1 0 1 0 1

1 1 0 1 1 1 0 1 1 1 0 1 1 1 0 1 1

1 1 0 1 1 1 0 1 1 1 0 1 1 1 1 0 01 0 1 0 0 0 1 0 0 0 1 0 0 0 0 1 1

wraparound

sumchecksum

Reliable Data Transfer

• See book for state machines and full explanations

• Case 1: underlying channel completely reliable– Just have a sender and receiver.

• Case 2: bit errors (but no packet loss)– How do you know?

• Receiver has to acknowledge (ACK) or negative (NAK)• A NAK makes the sender resend the packet• NAK based on UDP checksum (talk about timers later)

– Reliable transfer based on retransmission = ARQ (Automatic Repeat reQuest) protocol

– 3 capabilities: Error detection, receiver feedback, retransmission

– Stop and wait protocol

• What if the ACK or NAK packet is corrupted?– Just resend the old packet.

• Duplicate packets: But how does the receiver know it is the same packet and not the next one?

• Add a sequence number!

• Do we really need a NAK?– Just ACK the last packet that was received.

• Case 3: bit errors and packet loss– What if a packet gets lost?

• Set a timer on each packet sent. When the timer runs out, send the packet again.

• Can we handle duplicate packets?

– How long?• At least as long as a RTT


Performance of rdt3.0

• rdt3.0 works, but performance stinks• ex: 1 Gbps link, 15 ms prop. delay, 8000 bit packet:

U sender: utilization – fraction of time sender busy sending

U sender

= .008

30.008 = 0.00027

microseconds

L / R

RTT + L / R =

1KB pkt every 30 msec -> 33kB/sec (264 kbps) thruput over 1 Gbps link network protocol limits use of physical resources!

dsmicrosecon8bps10

bits80009

R

Ldtrans

• Pipelining will fix it!– Make sure your sequence numbers are large

enough– Buffering packets

• Sender buffers packets that have been transmitted but not yet acknowledged

• Receiver buffers correctly received packets• Two basic approaches: Go-Back-N and selective repeat


Pipelined protocolsPipelining: 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

Go-Back-N (sliding window)

Sender:• k-bit seq # in pkt header• “window” of up to N, consecutive unACKed pkts allowed

3-21

ACK(n): ACKs all pkts up to, including seq # n - “cumulative ACK” may receive duplicate ACKs (see receiver)

timer for each in-flight pkt timeout(n): retransmit pkt n and all higher seq # pkts in window

ACK-only: always send ACK for correctly-received pkt with highest in-order seq #– may generate duplicate ACKs– need only remember expectedseqnum

• out-of-order pkt: – discard (don’t buffer) -> no receiver buffering!– Re-ACK pkt with highest in-order seq #

• Problems with GBN?– It can spit out a lot of needless packets onto the wire.

• A single error will really do some damage. A wire with lots of errors? Lots of needless duplicate packets


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– again limits seq #s of sent, unACKed pkts– The sender and receiver window will not always coincide!

• If your sequence numbers aren’t big enough, you won’t know which is which


Selective repeat: sender, receiver windows


Selective repeat

data from above :• if next available seq # in window, send pkt

timeout(n):• resend pkt n, restart timer

ACK(n) in [sendbase,sendbase+N]:

• mark pkt n as received• if n smallest unACKed pkt, advance window base to next unACKed seq #

senderpkt n in [rcvbase, rcvbase+N-1] send ACK(n) out-of-order: buffer in-order: deliver (also deliver

buffered, in-order pkts), advance window to next not-yet-received pkt

pkt n in [rcvbase-N,rcvbase-1] ACK(n)

otherwise: ignore

receiver


Pipelining Protocols - SummaryGo-back-N: overview• sender: up to N unACKed pkts in pipeline• receiver: only sends cumulative ACKs

– doesn’t ACK pkt if there’s a gap• sender: has timer for oldest unACKed pkt

– if timer expires: retransmit all unACKed packets

Selective Repeat: overview• sender: up to N unACKed

packets in pipeline• receiver: ACKs individual pkts• sender: maintains timer for

each unACKed pkt– if timer expires: retransmit only

unACKed packet

Reliable Data Transfer Mechanisms

• See table p. 242– Checksum– Timer– Sequence number– Acknowledgement– Negative acknowledgement– Window, pipelining

TCP

• Read 3.5 to the end• Point to point

– Single sender, single receiver– Multicasting (4.7) will not work with TCP

• 3 way handshake– SYN– SYN-ACK– ACK

• TCP sets aside a send buffer – where the application message data gets put

• TCP takes chunks of data from this buffer and sends segments

Vocabulary

• RTT = Round Trip Time• MSS = Maximum segment size

– Maximum amount of data that TCP can grab and place into a segment

– This is application layer data – does not include TCP headers, etc.

• MTU = Maximum transmission unit– The largest link-layer frame that can be sent by the local

sending host– This will have a lot of bearing on the MSS– Common values: 1,460, 536, and 512 bytes


TCP: Overview RFCs: 793, 1122, 1323, 2018, 2581

• full duplex data:– bi-directional data flow in

same connection– MSS: maximum segment

size

• connection-oriented: – handshaking (exchange of

control msgs) init’s 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

• send & receive buffers

socketdoor

T C Psend buffer

T C Preceive buffer

socketdoor

segm ent

applicationwrites data

applicationreads data


TCP segment structure

source port # dest port #

32 bits

applicationdata (variable length)

sequence number

acknowledgement numberReceive window

Urg data pointerchecksumFSRPAU

headlen

notused

Options (variable length)

URG: urgent data (generally not used)

ACK: ACK #valid

PSH: push data now(generally not used)

RST, SYN, FIN:connection estab(setup, teardown

commands)

# bytes rcvr willingto accept

countingby bytes of data(not segments!)

Internetchecksum

(as in UDP)


TCP seq. #’s and ACKsSeq. #’s:

– byte stream “number” of first byte in segment’s data

ACKs:– seq # of next byte

expected from other side

– cumulative ACKQ: how receiver handles out-

of-order segments– A: TCP spec doesn’t

say, - up to implementer

Host A Host B

Seq=42, ACK=79, data = ‘C’

Seq=79, ACK=43, data = ‘C’

Seq=43, ACK=80

Usertypes‘C’

host ACKsreceipt of echoed‘C’

host ACKsreceipt of‘C’, echoesback ‘C’

timesimple telnet scenario


TCP Round Trip Time and 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



EstimatedRTT = (1- )*EstimatedRTT + *SampleRTT

Exponential weighted moving average influence of past sample decreases exponentially fast typical value: = 0.125


Example RTT estimation:RTT: gaia.cs.umass.edu to fantasia.eurecom.fr

100

150

200

250

300

350

1 8 15 22 29 36 43 50 57 64 71 78 85 92 99 106

time (seconnds)

RTT

(mill

isec

onds

)

SampleRTT Estimated RTT



Setting the timeout• EstimtedRTT plus “safety margin”

– large variation in EstimatedRTT -> larger safety margin• first estimate of how much SampleRTT deviates from EstimatedRTT:

TimeoutInterval = EstimatedRTT + 4*DevRTT

DevRTT = (1-)*DevRTT + *|SampleRTT-EstimatedRTT|

(typically, = 0.25)

Then set timeout interval:


TCP reliable data transfer

• TCP creates rdt service on top of IP’s unreliable service• pipelined segments• cumulative ACKs• TCP uses single retransmission timer

• retransmissions are triggered by:– timeout events– duplicate ACKs

• initially consider simplified TCP sender:– ignore duplicate ACKs– ignore flow control, congestion control


TCP sender events: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

timeout:• retransmit segment that

caused timeout• restart timer ACK rcvd:• if acknowledges

previously unACKed segments– update what is known to be

ACKed– start timer if there are

outstanding segments


TCP sender(simplified)

NextSeqNum = InitialSeqNum SendBase = InitialSeqNum

loop (forever) { switch(event)

event: data received from application above create TCP segment with sequence number NextSeqNum if (timer currently not running) start timer pass segment to IP NextSeqNum = NextSeqNum + length(data)

event: timer timeout retransmit not-yet-acknowledged segment with smallest sequence number start timer

event: ACK received, with ACK field value of y if (y > SendBase) { SendBase = y if (there are currently not-yet-acknowledged segments) start timer }

} /* end of loop forever */

Comment:• SendBase-1: last cumulatively ACKed byteExample:• SendBase-1 = 71;y= 73, so the rcvrwants 73+ ;y > SendBase, sothat new data is ACKed


TCP ACK generation [RFC 1122, RFC 2581]

Event at Receiver

Arrival of in-order segment withexpected seq #. All data up toexpected seq # already ACKed

Arrival of in-order segment withexpected seq #. One other segment has ACK pending

Arrival of out-of-order segmenthigher-than-expect seq. # .Gap detected

Arrival of segment that partially or completely fills gap

TCP Receiver action

Delayed ACK. Wait up to 500msfor 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 thatsegment starts at lower end of gap


Fast Retransmit

• 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 for that

segment

• If sender receives 3 ACKs for same data, it assumes that segment after ACKed data was lost:– fast retransmit: resend

segment before timer expires


event: ACK received, with ACK field value of y if (y > SendBase) { SendBase = y if (there are currently not-yet-acknowledged segments) start timer } else { increment count of dup ACKs received for y if (count of dup ACKs received for y = 3) { resend segment with sequence number y }

Fast retransmit algorithm:

a duplicate ACK for already ACKed segment

fast retransmit

Go-Back-N or Selective Repeat?

• The book says it’s sorta both. To me it mostly looks like GBN– Out of order segments not individually ACKed

• However– Many TCP implementations will buffer out of

order segments– TCP will also usually only retransmit a single

segment rather than all of them

Flow Control (NOT congestion control)

• TCP creates a receive buffer– Data is put into the receive buffer once it has been received

correctly and in order– The application reads from the receive buffer

• Sometimes not right away.

• Flow control tries not to overflow this receive buffer• Each sender maintains a variable called the receive window

– What if the receive window goes to 0?– In this case, the sending host is required to send segments with 1

data byte• What happens in UDP when the UDP receive buffer

overflows?


TCP Flow Control

• receive side of TCP connection has a receive buffer:

• speed-matching service: matching send rate to receiving application’s drain rate

app process may be slow at reading from buffer

sender won’t overflowreceiver’s buffer bytransmitting too much, too fast

flow control

IPdatagrams

TCP data(in buffer)

(currently)unused bufferspace

applicationprocess


TCP Connection Management

Recall: TCP sender, receiver establish “connection” before exchanging data segments

• initialize TCP variables:– seq. #s– buffers, flow control info

(e.g. RcvWindow)• client: connection initiator Socket clientSocket = new

Socket("hostname","port

number"); • server: contacted by client Socket connectionSocket =

welcomeSocket.accept();

Three way handshake:Step 1: client host sends TCP SYN

segment to server– specifies initial seq #– no data

Step 2: server host receives SYN, replies with SYNACK segment

– server allocates buffers– specifies server initial seq. #

Step 3: client receives SYNACK, replies with ACK segment, which may contain data


TCP Connection Management (cont.)

Closing a connection:

client closes socket: clientSocket.close();

Step 1: client end system sends TCP FIN control segment to server

Step 2: server receives FIN, replies with ACK. Closes connection, sends FIN.

client

FIN

server

ACK

ACK

FIN

close

close

closed

timed

wai

t


TCP Connection Management (cont.)

Step 3: client receives FIN, replies with ACK.

– Enters “timed wait” - will respond with ACK to received FINs

Step 4: server, receives ACK. Connection closed.

Note: with small modification, can handle simultaneous FINs.

client

FIN

server

ACK

ACK

FIN

closing

closing

closed

timed

wai

tclosed


TCP Connection Management (cont)

TCP clientlifecycle

TCP serverlifecycle

SYN Flood Attack

• Bad guy sends a bunch of TCP SYN segments• Server opens up buffers to create this segment• Resources all become allocated to half open TCP

connections– This is called a SYN flood attack

• SYN Cookies– The server allocates on the resources upon receipt of a ACK (third

part of handshake) segment rather than a SYN segment– It knows because the sequence field the server sent out was a

special number (complex hash function of source and destination IP and port plus the server’s secret number)

– P. 269s

How does nmap work?

• To find out what is on a port, nmap sends a TCP SYN segment to that port– If the port responds with a SYNACK, it labels the

port open– If the response is a TCP RST segment, it means the

port is not blocked, but it is closed– If the response is nothing, the port is blocked by a

firewall

Congestion Control Principles

• Typical cause of congestion:– Overflowing of router buffers as the network

becomes congested.

Scenario 1

• Scenario 1: two senders, a router with infinite buffers– Always maximizing throughput looks good with throughput

alone. (left)– Maximizing throughput is bad when looking at delays

(right)

Scenario 2

• Scenario 2: Routers with finite buffers & retransmission– If you are constantly resending packets,

throughput is even lower since a % of the packets are retransmissions

– Creating these large delays, you may send needless retransmissions, wasting precious router resources

Scenario 3

• Scenario 3: multiple hops– When a packet is dropped along a path, the

transmission capacity at each of the upstream links ends up being wasted


Approaches towards congestion control

end-end congestion control:• no explicit feedback from

network• congestion inferred from end-

system observed loss, delay• approach taken by TCP

network-assisted congestion control:

• routers provide feedback to end systems– single bit indicating

congestion (SNA, DECbit, TCP/IP ECN, ATM)

– explicit rate sender should send at

two broad approaches towards congestion control:


TCP congestion control: goal: TCP sender should transmit as fast as possible, but

without congesting network Q: how to find rate just below congestion level

decentralized: each TCP sender sets its own rate, based on implicit feedback: ACK: segment received (a good thing!), network not

congested, so increase sending rate lost segment: assume loss due to congested network, so

decrease sending rate


TCP congestion control: bandwidth probing

“probing for bandwidth”: increase transmission rate on receipt of ACK, until eventually loss occurs, then decrease transmission rate continue to increase on ACK, decrease on loss (since available

bandwidth is changing, depending on other connections in network)

ACKs being received, so increase rate

X

X

XX

X loss, so decrease rate

send

ing

rate

time

Q: how fast to increase/decrease? details to follow

TCP’s“sawtooth”behavior


TCP Congestion Control: details

• sender limits rate by limiting number of unACKed bytes “in pipeline”:

– cwnd: differs from rwnd (how, why?)– sender limited by min(cwnd,rwnd)

• roughly,

• cwnd is dynamic, function of perceived network congestion

rate = cwnd

RTT bytes/sec

LastByteSent-LastByteAcked cwnd

cwndbytes

RTT

ACK(s)


TCP Congestion Control: more details

segment loss event: reducing cwnd

• timeout: no response from receiver– cut cwnd to 1

• 3 duplicate ACKs: at least some segments getting through (recall fast retransmit)– cut cwnd in half, less

aggressively than on timeout

ACK received: increase cwnd slowstart phase:

increase exponentially fast (despite name) at connection start, or following timeout

congestion avoidance: increase linearly


TCP Slow Start• when connection begins, cwnd = 1

MSS– example: MSS = 500 bytes & RTT

= 200 msec– initial rate = 20 kbps

• available bandwidth may be >> MSS/RTT– desirable to quickly ramp up to

respectable rate• increase rate exponentially until first

loss event or when threshold reached– double cwnd every RTT– done by incrementing cwnd by 1

for every ACK received

Host A

one segment

RTT

Host B

time

two segments

four segments


Transitioning into/out of slowstartssthresh: cwnd threshold maintained by TCP• on loss event: set ssthresh to cwnd/2

– remember (half of) TCP rate when congestion last occurred • when cwnd >= ssthresh: transition from slowstart to congestion avoidance

phase

slow start timeout

ssthresh = cwnd/2cwnd = 1 MSSdupACKcount = 0retransmit missing segment

timeoutssthresh = cwnd/2 cwnd = 1 MSSdupACKcount = 0retransmit missing segment

Lcwnd > ssthresh

cwnd = cwnd+MSSdupACKcount = 0transmit new segment(s),as allowed

new ACKdupACKcount++

duplicate ACK

Lcwnd = 1 MSSssthresh = 64 KBdupACKcount = 0 congestion

avoidance


TCP: congestion avoidance• when cwnd > ssthresh

grow cwnd linearly

– increase cwnd by 1 MSS per RTT

– approach possible congestion slower than in slowstart

– implementation: cwnd = cwnd + MSS/cwnd for each ACK received

ACKs: increase cwnd by 1 MSS per RTT: additive increase

loss: cut cwnd in half (non-timeout-detected loss ): multiplicative decrease

AIMD

AIMD: Additive IncreaseMultiplicative Decrease


Popular “flavors” of TCP

ssthresh

ssthresh

TCP Tahoe

TCP Reno

Transmission round

cwnd w

indow

siz

e (

in

segm

en

ts)


Summary: TCP Congestion Control

• when cwnd < ssthresh, sender in slow-start phase, window grows exponentially.

• when cwnd >= ssthresh, sender is in congestion-avoidance phase, window grows linearly.

• when triple duplicate ACK occurs, ssthresh set to cwnd/2, cwnd set to ~ ssthresh

• when timeout occurs, ssthresh set to cwnd/2, cwnd set to 1 MSS.


Fairness (more)Fairness and UDP• multimedia apps often do

not use TCP– do not want rate throttled

by congestion control• instead use UDP:

– pump audio/video at constant rate, tolerate packet loss

Fairness and parallel TCP connections

• nothing prevents app from opening parallel connections between 2 hosts.

• web browsers do this • example: link of rate R

supporting 9 connections; – new app asks for 1 TCP, gets rate

R/10– new app asks for 11 TCPs, gets

R/2 !

instructor: christopher cole some slides taken from kurose & ross book it 347: chapter 3...

Documents

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