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Chapter 3 outline
3.1 transport-layer services 3.2 multiplexing and demultiplexing 3.3 connectionless transport: UDP 3.4 connection-oriented transport: TCP
segment structure reliable data transfer flow control connection management
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
household analogy:
12 kids in Anns house sending letters to 12 kids in Bills house: hosts = houses processes = kids app messages = letters in envelopes transport protocol = Ann and Bill who demux to inhouse siblings network-layer protocol = postal service
application transport network data link physical network data link physical
network data link physical network data link physical network data link physical network data link physical application transport network data link physical
Chapter 3 outline
3.1 transport-layer services 3.2 multiplexing and demultiplexing 3.3 connectionless transport: UDP 3.4 connection-oriented transport: TCP
segment structure reliable data transfer flow control connection management
Multiplexing/demultiplexing
multiplexing at sender: handle data from multiple sockets, add transport header (later used for demultiplexing)
application application
demultiplexing at receiver: use header info to deliver received segments to correct socket
P1
P2
application
P3
transport network link physical
transport network
P4
transport network link physical
socket process
link
physical
32 bits
source port # dest port #
host uses IP addresses & port numbers to direct segment to appropriate socket
Connectionless demultiplexing
recall: when creating datagram to send into UDP socket, must specify
destination IP address destination 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 3-10
P1
transport
P3
transport network link physical source port: 6428 dest port: 9157 source port: ? dest port: ? network link physical
P4
transport network link physical
Connection-oriented demux
demux: receiver uses all four values to direct segment to appropriate socket
P4
P5
transport
P6
application
P3
transport network link physical network
P2
P3
link
physical
server: IP address B
source IP,port: B,80 dest IP,port: A,9157 source IP,port: A,9157 dest IP, port: B,80
physical
host: IP address A
source IP,port: C,5775 dest IP,port: B,80 source IP,port: C,9157 dest IP,port: B,80
host: IP address C
three segments, all destined to IP address: B, dest port: 80 are demultiplexed to different sockets
P3
transport network link physical
P4
transport network
P2
P3
link
physical
server: IP address B
source IP,port: B,80 dest IP,port: A,9157 source IP,port: A,9157 dest IP, port: B,80
physical
host: IP address A
source IP,port: C,5775 dest IP,port: B,80 source IP,port: C,9157 dest IP,port: B,80
host: IP address C
Chapter 3 outline
3.1 transport-layer services 3.2 multiplexing and demultiplexing 3.3 connectionless transport: UDP 3.4 connection-oriented transport: TCP
segment structure reliable data transfer flow control connection management
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
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
Transport Layer 3-17
UDP checksum
Goal: detect errors (e.g., flipped bits) in transmitted segment
sender:
treat segment contents, including header fields, as sequence of 16-bit integers checksum: addition (ones 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. But maybe errors nonetheless? More later .
Transport Layer 3-18
Note: when adding numbers, a carryout from the most significant bit needs to be added to the result
Chapter 3 outline
3.1 transport-layer services 3.2 multiplexing and demultiplexing 3.3 connectionless transport: UDP 3.4 principles of reliable data transfer 3.5 connection-oriented transport: TCP
segment structure reliable data transfer flow control connection management
TCP: Overview
point-to-point:
one sender, one receiver
connection-oriented:
handshaking (exchange of control msgs) inits sender, receiver state before data exchange
pipelined:
TCP congestion and flow control set window size
flow controlled:
sender will not overwhelm receiver
Transport Layer 3-21
source port #
dest port #
sequence number
acknowledgement number
head not UAP R S F len used
checksum
window size
User types C
Seq=43, ACK=80
too short: premature timeout, unnecessary retransmissions too long: slow reaction to segment loss
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
exponential weighted moving average influence of past sample decreases exponentially fast typical value: = 0.125
RTT: gaia.cs.umass.edu to fantasia.eurecom.fr
350
RTT (milliseconds)
300
RTT (milliseconds)
250
200
sampleRTT
150
EstimatedRTT
1 8 15 22 29 36 43 50 57 64 71 78 85 92 99 106
100
time (seconds)
SampleRTT
time (seconnds)
Estimated RTT
safety margin
Chapter 3 outline
3.1 transport-layer services 3.2 multiplexing and demultiplexing 3.3 connectionless transport: UDP 3.4 connection-oriented transport: TCP
segment structure reliable data transfer flow control connection management
timeout: 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
Transport Layer 3-30
L
NextSeqNum = InitialSeqNum SendBase = InitialSeqNum
timeout
retransmit not-yet-acked segment with smallest seq. # start timer
SendBase=92 Seq=92, 8 bytes of data timeout ACK=100 timeout Seq=92, 8 bytes of data Seq=100, 20 bytes of data
ACK=100 ACK=120 Seq=92, 8 bytes of data SendBase=100 ACK=100 SendBase=120 ACK=120 SendBase=120 Seq=92, 8 bytes of data
premature timeout
Transport Layer 3-32
ACK=100
ACK=120
cumulative ACK
Transport Layer 3-33
immediate send ACK, provided that segment starts at lower end of gap
Transport Layer 3-34
Host B
X
ACK=100 timeout ACK=100 ACK=100 ACK=100 Seq=100, 20 bytes of data
Chapter 3 outline
3.1 transport-layer services 3.2 multiplexing and demultiplexing 3.3 connectionless transport: UDP 3.4 connection-oriented transport: TCP
segment structure reliable data transfer flow control connection management
TCP code
receiver controls sender, so sender wont overflow receivers buffer by transmitting too much, too fast
flow control
IP code
from sender
receiver advertises free buffer space by including rwnd value in TCP header of receiver-to-sender segments
RcvBuffer size set via socket options (typical default is 4096 bytes) many operating systems autoadjust RcvBuffer
to application process
RcvBuffer rwnd
sender limits amount of unacked (in-flight) data to receivers rwnd value guarantees receive buffer will not overflow
receiver-side buffering
Chapter 3 outline
3.1 transport-layer services 3.2 multiplexing and demultiplexing 3.3 connectionless transport: UDP 3.4 connection-oriented transport: TCP
segment structure reliable data transfer flow control connection management
Connection Management
before exchanging data, sender/receiver handshake:
agree to establish connection (each knowing the other willing to establish connection) agree on connection parameters
application
connection state: ESTAB connection variables: seq # client-to-server server-to-client rcvBuffer size at server,client
application
connection state: ESTAB connection Variables: seq # client-to-server server-to-client rcvBuffer size at server,client
network
network
choose x ESTAB
req_conn(x)
acc_conn(x) ESTAB
variable delays retransmitted messages (e.g. req_conn(x)) due to message loss message reordering cant see other side
req_conn(x)
ESTAB
acc_conn(x) retransmit req_conn(x) ESTAB req_conn(x)
connection x completes
retransmit data(x+1)
server forgets x ESTAB client terminates
connection x completes
accept data(x+1)
client terminates
req_conn(x) data(x+1)
server forgets x
ESTAB accept data(x+1)
server state
LISTEN
SYNSENT
SYNbit=1, Seq=x
choose init seq num, y send TCP SYNACK SYN RCVD msg, acking SYN
received SYNACK(x) indicates server is live; ESTAB send ACK for SYNACK; this segment may contain client-to-server data
ACKbit=1, ACKnum=y+1
received ACK(y) indicates client is live
ESTAB
L
SYN(x)
SYNACK(seq=y,ACKnum=x+1) create new socket for communication back to client
listen
SYN(seq=x)
SYN sent
SYNACK(seq=y,ACKnum=x+1)
ACK(ACKnum=y+1)
ACK(ACKnum=y+1)
L
Transport Layer 3-45
server state
ESTAB
can no longer send but can receive data wait for server close
FIN_WAIT_1
FIN_WAIT_2
LAST_ACK
can no longer send data
CLOSED
Transport Layer 3-47
Chapter 3 outline
3.1 transport-layer services 3.2 multiplexing and demultiplexing 3.3 connectionless transport: UDP 3.4 connection-oriented transport: TCP
segment structure reliable data transfer flow control connection management
informally: too many sources sending too much data too fast for network to handle different from flow control! manifestations: lost packets (buffer overflow at routers) long delays (queueing in router buffers) a top-10 problem!
two senders, two receivers one router, infinite buffers output link capacity: R no retransmission
throughput:
lout
Host B
R/2
delay
lout
lout
Host B
lout
lin
R/2
copy
lout
Host B
lout
no buffer space!
Host B
Transport Layer 3-53
lout
lin
R/2
lout
Host B
Transport Layer 3-54
packets can be lost, dropped at router due to full buffers sender times out prematurely, sending two copies, both of which are delivered
timeout copy
R/2 when sending at R/2, some packets are retransmissions including duplicated that are delivered!
lout
lin
R/2
lin l'in
A
lout
Host B
Transport Layer 3-55
packets can be lost, dropped at router due to full buffers sender times out prematurely, sending two copies, both of which are delivered
R/2 when sending at R/2, some packets are retransmissions including duplicated that are delivered!
lout
lin
R/2
costs of congestion:
more work (retrans) for given goodput unneeded retransmissions: link carries multiple copies of pkt decreasing goodput
increase ? A: as red lin increases, all arriving blue pkts at upper queue are dropped, blue throughput g 0
lout
Host B
Host D Host C
lout
lin
C/2
another cost of congestion: when packet dropped, any upstream transmission capacity used for that packet was wasted!
Transport Layer 3-58
no explicit feedback from network congestion inferred from end-system observed loss, delay approach taken by TCP
routers provide feedback to end systems single bit indicating congestion (SNA, DECbit, TCP/IP ECN, ATM) explicit rate for sender to send at
Transport Layer 3-59
Chapter 3 outline
3.1 transport-layer services 3.2 multiplexing and demultiplexing 3.3 connectionless transport: UDP 3.4 connection-oriented transport: TCP
segment structure reliable data transfer flow control connection management
approach: sender increases transmission rate (window size), probing for usable bandwidth, until loss occurs additive increase: increase cwnd by 1 MSS every RTT until loss detected multiplicative decrease: cut cwnd in half after loss
cwnd: TCP sender congestion window size
additively increase window size . until loss occurs (then cut window in half)
time
Transport Layer 3-61
TCP sending rate: roughly: send cwnd bytes, wait RTT for ACKS, then send more bytes
rate
~ ~
cwnd
RTT
bytes/sec
when connection begins, increase rate exponentially until first loss event:
initially cwnd = 1 MSS double cwnd every RTT done by incrementing cwnd for every ACK received
Host A
Host B
RTT
time
duplicate acks)
Implementation:
variable ssthresh on loss event, ssthresh is set to 1/2 of cwnd just before loss event
Transport Layer 3-65
New ACK!
new ACK cwnd = cwnd+MSS dupACKcount = 0 transmit new segment(s), as allowed cwnd > ssthresh L timeout ssthresh = cwnd/2 cwnd = 1 MSS dupACKcount = 0 retransmit missing segment
new ACK cwnd = cwnd + MSS (MSS/cwnd) dupACKcount = 0 transmit new segment(s), as allowed
New ACK!
slow start
congestion avoidance
duplicate ACK dupACKcount++
New ACK!
New ACK cwnd = ssthresh dupACKcount = 0 dupACKcount == 3 ssthresh= cwnd/2 cwnd = ssthresh + 3 retransmit missing segment
fast recovery
duplicate ACK
TCP throughput
3 W bytes/sec 4 RTT
W/2
example: 1500 byte segments, 100ms RTT, want 10 Gbps throughput requires W = 83,333 in-flight segments throughput in terms of segment loss probability, L
[Mathis 1997]:
. MSS 1.22 TCP throughput = RTT L to achieve 10 Gbps throughput, need a loss rate of L = 210-10 a very small loss rate!
TCP Fairness
fairness goal: if K TCP sessions share same bottleneck link of bandwidth R, each should have average rate of R/K
TCP connection 1
TCP connection 2
additive increase gives slope of 1, as throughout increases multiplicative decrease decreases throughput proportionally
R
equal bandwidth share
loss: decrease window by factor of 2 congestion avoidance: additive increase loss: decrease window by factor of 2 congestion avoidance: additive increase
Connection 1 throughput
R
Transport Layer 3-70
Fairness (more)
Fairness and UDP multimedia apps often do not use TCP Fairness, parallel TCP connections application can open do not want rate multiple parallel throttled by congestion connections between two control hosts instead use UDP: web browsers do this send audio/video at e.g., link of rate R with 9 constant rate, tolerate packet loss existing connections:
new app asks for 1 TCP, gets rate R/10 new app asks for 11 TCPs, gets R/2
Transport Layer 3-71
Chapter 3: summary
principles behind transport layer services: multiplexing, demultiplexing reliable data transfer flow control congestion control instantiation, implementation in the Internet
UDP TCP
next: leaving the network edge (application, transport layers) into the network core