SUGAR-DFJ

transport service:
provide logical communication between app process running on different h
ost
run in end systems
send side: break app msg into segment, pass to network layer
receive side: reassemble segment into msg, pass to app layer
Internet: TCP and UDP
TCP: reliable, in-order delivery
congestion control
flow control
connection setup
UDP: unreliable, unordered delivery
no-frills extension of best-effort IP
no delay guarantee
no bandwidth guarantee
demultiplexig at rcv host:
delivering received segment to correct socket
multiplexing at send host
gathering data from multiple socket, envelope data with header
UDP: User Datagram Protocol
no fills, bare bones
best effort service
connectionless:
no handshaking
UDP segment handled independently
no connection establishment
simple
small segment header
no congestion control
streaming multimedia app
loss tolerant
rate sensitive
DNS, SNMP
reliable transfer over UDP: add at application layer
checksum: detect error in transmit segment
Reliable data transfer:
rdt1.0: reliable transfer over a reliable channel
no bit error
no loss of packet
rdt2.0: channel with bit error
ACK, NACK
error detection
receiver feedback -> sender
rdt2.1: handle garble ACK/NAK
stop and wait
retransmit current packet if ACK/NAK garble

sender add sequence number to each packet

receiver discard duplicate packet
rdt2.2: NAK free protocol
rdt3.0: channel with error and loss
sender wait reasonable amount of time for ACK
retransmit if no ACK receive in this time
require countdown timer
Pipelined protocol:
pipelining: sender allow multiple, "in-flight", yet-to-be-acknowledge pa
cket
range of sequence number must be increase
buffering at sender and receiver
go-back-N, selective repeat
Go-back-N:
sender: up to N unACK packet in pipeline
receiver: only send cumulative ACK
doesn't ACK packet if there is a gap
sender has timer for oldest unACK packet
if time expire, retransmit all unACK packet
Selective repeat:
sender: up to N unACK packet in pipeline
receiver: ACK individual packet
sender: maintain timer for each unACK packet
if time expire, retransmit only unACK packet
Go-back-N:
k-bit sequence number in packet header
window of up to N consecutive unACK packet allow
ACK all packet up to, including sequence number - cumulative ACK
may receive duplicate ACK
timer for each in-flight packet
time out: retransmit packet n and all higher sequence number packet in w
indow
ACK-only: always send ACK for correctly received packet with highest ino
rder sequence number
may generate duplicate ACK
need only remember expected sequence number
out of order packet
discard
re ACK packet with highest inorder sequence number
Selective repeat:
receiver ack all correctly receive packet
sender only resend packet for which ACK not receive
sender window:
N consecutive sequence number
limit sequence of sent, unACK packet
sender:
data from above:
if next available sequence number in window, send packet
timeout(n):
resend packet n, restart timer
ACK(n)

mark packet n as receive

if n smallest unACK packet, advance window base to next
unACK sequence number
receiver:
send ACK(n)
out of order: buffering
in order: deliver, advance window to next not-yet-receive packet
TCP:
point-to-point:
1 sender, 1 receiver
reliable, inorder
no message boundary
pipeline:
TCP congestion and flow control set window size
send and receive buffer
full duplex data:
bidirectional data flow in same connection
MSS: maximum segment size
connection-oriented:
handshaking
flow control:
sender will not overwhelm receiver
TCP segment:
source port number
dest port number
sequence number
ACK number
header length
URG: urgent data
Receive window
checksum
urg data pointer
options
application data
EstimatedRTT = (1 - alpha) * EstimatedRTT + alpha * SampleRTT
TCP reliable data transfer:
TCP create rdt service on top of IP unreliable service
pipelined segment
cumulative ACKs
TCP use single retransmission timer
retransmission are triggered by:
timeout event
duplicate ACK
initially consider simplified TCP sender:
ignore duplicate ACKs
ignore flow control, congestion control
sender:
create segment with seq #
seq # is byte-stream number of first data byte in segment
start timer if not already running

timeout:
retransmit segment that cause timeout
restart timer
ACK receive:
if acknowledge previously unACK segment:
update what is known to be ACK
start timer if there are outstanding segments
fast retransmit
timeout period often relatively long
long delay before resending lost packet
detect lost segment via duplicate ACK
sender often send many segmenet back-to-back
if segment is lost, there will be many duplicate ACK for
that segment
fast retransmit: resend segment before timer expire
flow control:
sender won't overflow receiver buffer by transmitting too much,
too fast
receiver advertise unused buffer space by including rwnd value i
n segment header
sender: limit # of unACK byte to rwnd
connection management:
establish connection before exchanging data segment
initialize TCP variables
seq #s, buffer, flow control info
3 way handshake:
client host send TCP-SYN segment to server
specify initial seq #, no data
server host receive SYN, reply with SYNACK segment:
server allocate buffer
specify server initial seq #
client receive SYNACK, reply with ACK segment, may conta
in data
closing connection:
client end system send TCP FIN control segment to server
server receive FIN, reply with an ACK, close connection,
send FIN
client receive FIN, reply with an ACK
enter time wait, will respond with ACK to receiv
e FIN
server receive ACK, close connection
Congestion control:
too many source sending too much data too fast for network to handle
different from flow control
manifestation:
lost packet (buffer overflow at router)
long delay (queueing in router buffer)
cost of congestion:
more work for given goodput
unneeded retransmission
when packet drop, any upstream transmission capacity use for tha
t packet was wasted
end-end congestion:

no explicit feedback from network

congestion infer from end-system observe loss, delay
approach taken by TCP
network-assisted congestion control:
router provide feedback to end system
single bit indicating congestion
explicit rate sender should send
ABR: avaliable bit rate:
elastic service
if sender's path underload:
sender should use available bandwidth
if sender's path congested:
sender throttled to minimum guarantee rate
RM (resource management):
sent by sender, intersperse with data cell
bit in RM cell set by switch
NI bit: no increase in rate
CI bit: congestion indication
RM cell return to sender by receiver, with bit intact
TCP congestion control:
should transmit as fast as possible, but without congesting network
decentralized: each TCP sender set its own rate, base on implicit feedba
ck:
ACK: segment receive, network not congest, increase sending rate
lost segment: assume loss due to congestion, decrease sending ra
te
probing for bandwidth:
increase transmission rate on receipt of ACK, until eventually l
oss occur, then decrease transmission rate
increase exponentially fast at connection start, or following timeout
congestion avoidance: increase linearly
TCP slow start:
when begin, cwnd = 1 MSS
increase rate exponentially until first loss event or when thres
hold reached
double cwnd every RTT
done by incrementing cwnd by 1 for every ACK received
AIMD: Additive Increase, Multiplicative Decrease
TCP fairness