Chapter 3 Transport Layer
Transport Layer 3-1
Chapter 3: Transport Layer
our goals:
understand principles behind transport layer services:
multiplexing, demultiplexing
reliable data transfer
flow control
congestion control
learn about Internet transport layer protocols:
UDP: connectionless transport
TCP: connection-oriented reliable transport
TCP congestion control
Transport Layer 3-2
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
3.6 principles of congestion control
3.7 TCP congestion control
Transport Layer 3-3
Transport services and protocols
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
application
transport
network
data link
physical
application
transport
network
data link physical
Transport Layer 3-4
Transport vs. network layer
network layer: logical communication between hosts
household analogy:
12 kids in Ann’s house sending letters to 12 kids in Bill’s house:
hosts = houses
processes = kids
app messages = letters in envelopes
transport protocol = Ann and Bill who demux to in- house siblings
network-layer protocol = postal service
transport layer: logical
communication between processes
relies on, enhances, network layer services
Transport Layer 3-5
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
application
transport
network data link physical
application
transport
network
data link
physical
network
network
data link
data link physical
physical
network
data link
physical
network
data link
physical
network
data link physical
network
data link
physical
network data link physical
Transport Layer 3-6
IP Service
The IP service model is a best-effort delivery service. This means that IP makes its “best effort” to deliver segments between communicating hosts, but it makes no guarantees.
It does not guarantee segment delivery, it does not guarantee orderly delivery of segments, and it does not guarantee the integrity of the data in the segments. For these reasons, IP is said to be an unreliable service.
Transport Layer 3-7
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
3.6 principles of congestion control
3.7 TCP congestion control
Transport Layer 3-8
What do UDP and TCP do?
The most fundamental responsibility of UDP and TCP is to extend IP’s delivery service between two end systems to a delivery service between two processes running on the end systems.
Extending host-to-host delivery to process-to- process delivery is called transport-layer multiplexing and demultiplexing.
Transport Layer 3-9
Multiplexing/demultiplexing
multiplexing at sender:
handle data from multiple sockets, add transport header (later used for demultiplexing)
demultiplexing at receiver:
use header info to deliver received segments to correct socket
socket
process
application
P1 P2
transport
network
link
physical
application
P4
transport
network
link
physical
application
P3
transport
network
link
physical
Transport Layer 3-10
How demultiplexing works
host receives IP datagrams
each datagram has source IP address, destination IP address
each datagram carries one transport-layer segment
each segment has source, destination port number
host uses IP addresses & port numbers to direct segment to appropriate socket
32 bits
source port #
dest port #
other header fields
application data
(payload)
TCP/UDP segment format
Transport Layer 3-11
Connectionless demultiplexing
recall: created socket has recall: when creating
host-local port #:
DatagramSocket mySocket1
= new DatagramSocket(12534);
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
when host receives UDP segment:
checks destination port # in segment
directs UDP segment to socket with that port #
Transport Layer 3-12
Connectionless demux: example
DatagramSocket
mySocket2 = new
DatagramSocket
(9157);
DatagramSocket
serverSocket = new
DatagramSocket
(6428);
DatagramSocket
mySocket1 = new
DatagramSocket
(5775);
application
P1
application
P4
transport
network
link
transport
network
link
physical
application
P3
transport
network
link
physical
physical
source port: 6428 dest port: 9157
source port: ? dest port: ?
source port: ? dest port: ?
source port: 9157 dest port: 6428
Transport Layer 3-13
Connection-oriented demux
TCP socket identified by 4-tuple:
source IP address
source port number dest IP address
dest port number
demux: receiver uses all four values to direct segment to appropriate socket
server host may support many simultaneous TCP sockets:
each socket identified by its own 4-tuple
web servers have different sockets for each connecting client
non-persistent HTTP will have different socket for each request
Transport Layer 3-14
Connection-oriented demux: example
application
P4 P5 P6
transport
network
link
physical
application
P2 P3
transport
network
link
physical
application
P3
transport
network
link
physical
host: IP address A
source IP,port: B,80 dest IP,port: A,9157
source IP,port: A,9157 dest IP, port: B,80
server: IP address B
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
Transport Layer 3-15
Connection-oriented demux: example
threaded server
application
P4
transport
network
link
physical
application
P2 P3
transport
network
link
physical
application
P3
transport
network
link
physical
host: IP address A
source IP,port: B,80 dest IP,port: A,9157
source IP,port: A,9157 dest IP, port: B,80
server: IP address B
source IP,port: C,5775 dest IP,port: B,80
source IP,port: C,9157 dest IP,port: B,80
host: IP address C
Transport Layer 3-16
TCP vs UDP Sockets
Important
In contrast with UDP, two arriving TCP segments with different source IP addresses or source port numbers will (with the exception of a TCP segment carrying the original connection- establishment request) be directed to two different sockets.
Transport Layer 3-17
Why Use UDP? TCP Seems So much Easier?
Finer application-level control over
what data is sent, and when
TCP – Congestion Control, Resending Segments UDP – ‘Closer’ to the Network Layer
No connection establishment TCP – 3 Way Handshake, UDP – None
No connection state
Less overhead for server, server can handle more UDP connections compared to TCP
Small packet header overhead TCP – 20 Bytes, UDP 8 Bytes
Transport Layer 3-18
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
3.6 principles of congestion control
3.7 TCP congestion control
Transport Layer 3-19
UDP: User Datagram Protocol [RFC 768]
“no frills,” “bare bones” Internet transport protocol
UDP use:
streaming multimedia apps (loss tolerant, rate
“best effort” service, sensitive)
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
DNS SNMP
reliable transfer over UDP:
add reliability at application layer
application-specific error recovery!
Transport Layer 3-20
UDP: segment header
32 bits
length, in bytes of UDP segment, including header
why is there a UDP?
source port #
dest port #
length
application data
(payload)
checksum
no connection establishment (which can add delay) ie DNS
simple: no connection state at sender, receiver
small header size
no congestion control: UDP can blast away as fast as desired
UDP segment format
Transport Layer 3-21
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 (one’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. But maybe errors nonetheless? More later ….
Transport Layer 3-22
Internet checksum: example
example: add two 16-bit integers
11110011001100110 11101010101010101
wraparound
sum checksum
11011101110111011 11011101110111100
10100010001000011 Note: when adding numbers, a carryout from the most
significant bit needs to be added to the result
Transport Layer 3-23
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
3.6 principles of congestion control
3.7 TCP congestion control
Transport Layer 3-24
Principles of reliable data transfer
important in application, transport, link layers top-10 list of important networking topics!
characteristics of unreliable channel will determine complexity of reliable data transfer protocol (rdt)
Transport Layer 3-25
Principles of reliable data transfer
important in application, transport, link layers top-10 list of important networking topics!
characteristics of unreliable channel will determine complexity of reliable data transfer protocol (rdt)
Transport Layer 3-26
Principles of reliable data transfer
important in application, transport, link layers top-10 list of important networking topics!
characteristics of unreliable channel will determine complexity of reliable data transfer protocol (rdt)
Transport Layer 3-27
Reliable data transfer: getting started
rdt_send(): called from above, (e.g., by app.). Passed data to deliver to receiver upper layer
deliver_data(): called by rdt to deliver data to upper
send receive side side
udt_send(): called by rdt, to transfer packet over unreliable channel to receiver
rdt_rcv(): called when packet arrives on rcv-side of channel
Transport Layer 3-28
Reliable data transfer: getting started
we’ll:
incrementally develop sender, receiver sides of
reliable data transfer protocol (rdt)
consider only unidirectional data transfer
but control info will flow on both directions!
use finite state machines (FSM) to specify sender,
receiver
state: when in this “state” next state uniquely determined by next event
event causing state transition actions taken on state transition
state 1
event actions
state 2
Transport Layer 3-29
rdt1.0: reliable transfer over a reliable channel
underlying channel perfectly reliable no bit errors
no loss of packets
separate FSMs for sender, receiver:
sender sends data into underlying channel
receiver reads data from underlying channel
Wait for call from above
rdt_send(data)
packet = make_pkt(data) udt_send(packet)
sender
Wait for call from below
rdt_rcv(packet)
extract (packet,data) deliver_data(data)
receiver
Transport Layer 3-30
rdt2.0: channel with bit errors
underlying channel may flip bits in packet checksum to detect bit errors
the question: how to recover from errors:
acknowledgements (ACKs): receiver explicitly tells sender that pkt received OK
negative acknowledgements (NAKs): receiver explicitly tells sender that pkt had errors
sender retransmits pkt on receipt of NAK
How do humans recover from “errors”
new mechanisms in rdt2.0 (beyond rdt1.0): during conversation?
error detection
receiver feedback: control msgs (ACK,NAK) rcvr- >sender
Transport Layer 3-31
rdt2.0: channel with bit errors
underlying channel may flip bits in packet checksum to detect bit errors
the question: how to recover from errors:
acknowledgements (ACKs): receiver explicitly tells sender
that pkt received OK
negative acknowledgements (NAKs): receiver explicitly tells sender that pkt had errors
sender retransmits pkt on receipt of NAK
new mechanisms in rdt2.0 (beyond rdt1.0):
error detection
feedback: control msgs (ACK,NAK) from receiver to sender
Transport Layer 3-32
rdt2.0: FSM specification
rdt_send(data)
sndpkt = make_pkt(data, checksum)
receiver
rdt_rcv(rcvpkt) && corrupt(rcvpkt)
udt_send(NAK)
Wait for call from below
rdt_rcv(rcvpkt) && notcorrupt(rcvpkt)
extract(rcvpkt,data) deliver_data(data) udt_send(ACK)
udt_send(sndpkt)
rdt_rcv(rcvpkt) && isNAK(rcvpkt)
udt_send(sndpkt) rdt_rcv(rcvpkt) && isACK(rcvpkt)
L
sender
Wait for call from above
Wait for ACK or NAK
Transport Layer 3-33
rdt2.0: operation with no errors
rdt_send(data)
snkpkt = make_pkt(data, checksum)
udt_send(sndpkt)
rdt_rcv(rcvpkt) && isNAK(rcvpkt)
udt_send(sndpkt)
Wait for call from above
Wait for ACK or NAK
rdt_rcv(rcvpkt) && corrupt(rcvpkt)
udt_send(NAK)
Wait for call from below
rdt_rcv(rcvpkt) && notcorrupt(rcvpkt)
extract(rcvpkt,data) deliver_data(data) udt_send(ACK)
rdt_rcv(rcvpkt) && isACK(rcvpkt)
L
Transport Layer 3-34
rdt2.0: error scenario
rdt_send(data)
snkpkt = make_pkt(data, checksum)
udt_send(sndpkt)
rdt_rcv(rcvpkt) && isNAK(rcvpkt)
udt_send(sndpkt)
Wait for call from above
Wait for ACK or NAK
rdt_rcv(rcvpkt) && corrupt(rcvpkt)
udt_send(NAK)
Wait for call from below
rdt_rcv(rcvpkt) && notcorrupt(rcvpkt)
extract(rcvpkt,data) deliver_data(data) udt_send(ACK)
rdt_rcv(rcvpkt) && isACK(rcvpkt)
L
Transport Layer 3-35
rdt2.0 has a fatal flaw!
what happens if ACK/NAK corrupted?
sender doesn’t know what happened at receiver!
can’t just retransmit: possible duplicate
handling duplicates:
sender retransmits current pkt if ACK/NAK corrupted
sender adds sequence number to each pkt
receiver discards (doesn’t deliver up) duplicate pkt
stop and wait
sender sends one packet, then waits for receiver response
Transport Layer 3-36
rdt2.1: sender, handles garbled ACK/NAKs
rdt_send(data)
sndpkt = make_pkt(0, data, checksum)
udt_send(sndpkt)
rdt_rcv(rcvpkt) && ( corrupt(rcvpkt) || isNAK(rcvpkt) )
udt_send(sndpkt)
Wait for call 0 from above
Wait for ACK or NAK 0
rdt_rcv(rcvpkt)
&& notcorrupt(rcvpkt) && isACK(rcvpkt)
L
rdt_rcv(rcvpkt) && ( corrupt(rcvpkt) || isNAK(rcvpkt) )
udt_send(sndpkt)
rdt_rcv(rcvpkt)
&& notcorrupt(rcvpkt) && isACK(rcvpkt)
L
Wait for ACK or NAK 1
rdt_send(data)
sndpkt = make_pkt(1, data, checksum) udt_send(sndpkt)
Wait for call 1 from above
Transport Layer 3-37
rdt2.1: receiver, handles garbled ACK/NAKs
rdt_rcv(rcvpkt) && (corrupt(rcvpkt)
sndpkt = make_pkt(NAK, chksum) udt_send(sndpkt)
rdt_rcv(rcvpkt) &&
not corrupt(rcvpkt) && has_seq1(rcvpkt)
sndpkt = make_pkt(ACK, chksum) udt_send(sndpkt)
rdt_rcv(rcvpkt) && notcorrupt(rcvpkt) && has_seq0(rcvpkt)
extract(rcvpkt,data) deliver_data(data)
sndpkt = make_pkt(ACK, chksum) udt_send(sndpkt)
Wait for 0 from below
Wait for 1 from below
rdt_rcv(rcvpkt) && (corrupt(rcvpkt)
sndpkt = make_pkt(NAK, chksum) udt_send(sndpkt)
rdt_rcv(rcvpkt) &&
not corrupt(rcvpkt) && has_seq0(rcvpkt)
sndpkt = make_pkt(ACK, chksum) udt_send(sndpkt)
rdt_rcv(rcvpkt) && notcorrupt(rcvpkt) && has_seq1(rcvpkt)
extract(rcvpkt,data) deliver_data(data)
sndpkt = make_pkt(ACK, chksum) udt_send(sndpkt)
Transport Layer 3-38
rdt2.1: discussion
sender:
seq # added to pkt
two seq. #’s (0,1) will
suffice. Why?
must check if received ACK/NAK corrupted
twice as many states
state must “remember” whether “expected” pkt should have seq # of 0 or 1
receiver:
must check if received packet is duplicate
state indicates whether 0 or 1 is expected pkt seq #
note: receiver can not know if its last ACK/NAK received OK at sender
Transport Layer 3-39
rdt2.2: a NAK-free protocol
same functionality as rdt2.1, using ACKs only
instead of NAK, receiver sends ACK for last pkt
received OK
receiver must explicitly include seq # of pkt being ACKed
duplicate ACK at sender results in same action as NAK: retransmit current pkt
Transport Layer 3-40
rdt2.2: sender, receiver fragments
rdt_send(data)
sndpkt = make_pkt(0, data, checksum)
udt_send(sndpkt)
Wait for call 0 from above
Wait for ACK 0
rdt_rcv(rcvpkt) && ( corrupt(rcvpkt) ||
isACK(rcvpkt,1) ) udt_send(sndpkt)
rdt_rcv(rcvpkt)
&& notcorrupt(rcvpkt) && isACK(rcvpkt,0)
L
rdt_rcv(rcvpkt) && (corrupt(rcvpkt) ||
has_seq1(rcvpkt)) udt_send(sndpkt)
Wait for 0 from below
sender FSM fragment
receiver FSM fragment
rdt_rcv(rcvpkt) && notcorrupt(rcvpkt) && has_seq1(rcvpkt)
extract(rcvpkt,data) deliver_data(data)
sndpkt = make_pkt(ACK1, chksum) udt_send(sndpkt)
Transport Layer 3-41
rdt3.0: channels with errors and loss
new assumption:
underlying channel can also lose packets (data, ACKs)
checksum, seq. #, ACKs, retransmissions will be of help … but not enough
approach: sender waits “reasonable” amount of time for ACK
retransmits if no ACK received in this time
if pkt (or ACK) just delayed (not lost):
retransmission will be duplicate, but seq. #’s already handles this
receiver must specify seq # of pkt being ACKed
requires countdown timer Transport Layer 3-42
rdt3.0 sender
rdt_send(data)
sndpkt = make_pkt(0, data, checksum) udt_send(sndpkt)
start_timer L
rdt_rcv(rcvpkt) &&
( corrupt(rcvpkt) ||
isACK(rcvpkt,1) )
rdt_rcv(rcvpkt)
L
Wait for call 0from above
Wait for ACK0
timeout udt_send(sndpkt) start_timer
rdt_rcv(rcvpkt)
&& notcorrupt(rcvpkt) && isACK(rcvpkt,0)
stop_timer
rdt_rcv(rcvpkt)
&& notcorrupt(rcvpkt) && isACK(rcvpkt,1)
stop_timer
timeout
udt_send(sndpkt) start_timer
rdt_rcv(rcvpkt) && ( corrupt(rcvpkt) || isACK(rcvpkt,0) )
L
Wait for ACK1
Wait for call 1 from above
rdt_rcv(rcvpkt)
L
rdt_send(data)
sndpkt = make_pkt(1, data, checksum) udt_send(sndpkt)
start_timer
Transport Layer 3-43
rdt3.0 in action
sender
send pkt0
rcv ack0 send pkt1
rcv ack1 send pkt0
receiver sender
receiver
rcv pkt0 send ack0
pkt0
rcv pkt0 ack0 send ack0
pkt1
rcv pkt1 ack1 send ack1
pkt0 rcv pkt0 ack0 send ack0
(a) no loss
send pkt0 pkt0 rcv ack0 ack0
send pkt1 pkt1 X
timeout
loss
pkt1
resend pkt1
rcv pkt1 ack1 send ack1
pkt0
rcv pkt0 ack0 send ack0
rcv ack1 send pkt0
(b) packet loss
Transport Layer 3-44
rdt3.0 in action
sender
send pkt0
rcv ack0 send pkt1
receiver
rcv pkt0 send ack0
rcv pkt1 send ack1
sender
send pkt0
rcv ack0 send pkt1
receiver
pkt0 rcv pkt0 ack0 send ack0
pkt1 rcv pkt1 ack1 send ack1
pkt0
ack0
pkt1
ack1
pkt1 pkt0
ack1 ack0
pkt0
ack0
timeout
rcv ack1 send pkt0
rcv ack1 send pkt0
rcv pkt1 (detect duplicate)
resend pkt1
pkt1
X
timeout
loss
resend pkt1
rcv pkt1
ack1 (detect duplicate)
pkt0
rcv pkt0 ack0 send ack0
send ack1
rcv ack1 send pkt0
send ack1
rcv pkt0 send ack0
rcv pkt0 (detect duplicate)
(c) ACK loss
send ack0 (d) premature timeout/ delayed ACK
Transport Layer 3-45
Performance of rdt3.0
rdt3.0 is correct, but performance stinks
e.g.: 1 Gbps link, 15 ms prop. delay, 8000 bit packet:
D =L=8000bits = trans R 109 bits/sec
U sender: utilization – fraction of time sender busy sending Usender= L/R = .008 = 0.00027
8 microsecs
RTT + L / R 30.008
if RTT=30 msec, 1KB pkt every 30 msec: 33kB/sec thruput over 1 Gbps link
network protocol limits use of physical resources! Transport Layer 3-46
rdt3.0: stop-and-wait operation
sender first packet bit transmitted, t = 0
last packet bit transmitted, t = L / R RTT
ACK arrives, send next packet, t = RTT + L / R
Usender= L/R RTT + L / R
receiver
first packet bit arrives
last packet bit arrives, send ACK
= .008 30.008
= 0.00027
Transport Layer 3-47
Pipelined protocols
pipelining: sender allows multiple, “in-flight”, yet- to-be-acknowledged pkts
range of sequence numbers must be increased buffering at sender and/or receiver
two generic forms of pipelined protocols: go-Back-N, selective repeat
Transport Layer 3-48
Pipelining: increased utilization
sender
receiver
first packet bit transmitted, t = 0 last bit transmitted, t = L / R
RTT
ACK arrives, send next packet, t = RTT + L / R
first packet bit arrives
last packet bit arrives, send ACK
last bit of 2nd packet arrives, send ACK last bit of 3rd packet arrives, send ACK
3-packet pipelining increases utilization by a factor of 3!
.0024 = 0.00081 30.008
U sender = 3L / R RTT + L / R
=
Transport Layer 3-49
Pipelined protocols: overview
Go-back-N:
sender can have up to N unacked packets in pipeline
receiver only sends cumulative ack
doesn’t ack packet if there’s a gap
sender has timer for oldest unacked packet
when timer expires, retransmit all unacked packets
Selective Repeat:
sender can have up to N unack’ed packets in pipeline
rcvr sends individual ack for each packet
sender maintains timer for each unacked packet
when timer expires, retransmit only that unacked packet
Transport Layer 3-50
Go-Back-N: sender
k-bit seq # in pkt header
“window” of up to N, consecutive unack’ed pkts allowed
ACK(n): ACKs all pkts up to, including seq # n – “cumulative ACK”
may receive duplicate ACKs (see receiver) timer for oldest in-flight pkt
timeout(n): retransmit packet n and all higher seq # pkts in window
Transport Layer 3-51
GBN: sender extended FSM
rdt_send(data)
if (nextseqnum < base+N) {
sndpkt[nextseqnum] = make_pkt(nextseqnum,data,chksum) udt_send(sndpkt[nextseqnum])
if (base == nextseqnum)
L
base=1 nextseqnum=1
rdt_rcv(rcvpkt)
&& corrupt(rcvpkt)
start_timer nextseqnum++ }
else refuse_data(data)
Wait
timeout
start_timer udt_send(sndpkt[base]) udt_send(sndpkt[base+1])
... udt_send(sndpkt[nextseqnum-1])
rdt_rcv(rcvpkt) && notcorrupt(rcvpkt)
base = getacknum(rcvpkt)+1 If (base == nextseqnum)
stop_timer else
start_timer
Transport Layer 3-52
GBN: receiver extended FSM
L
expectedseqnum=1 sndpkt =
default udt_send(sndpkt)
Wait
rdt_rcv(rcvpkt)
&& notcurrupt(rcvpkt)
&& hasseqnum(rcvpkt,expectedseqnum)
extract(rcvpkt,data)
deliver_data(data)
sndpkt = make_pkt(expectedseqnum,ACK,chksum) udt_send(sndpkt)
expectedseqnum++
make_pkt(expectedseqnum,ACK,chksum)
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 #
Transport Layer 3-53
GBN in action
sender window (N=4)
012345678 012345678 45678 45678
012345678 012345678
sender
receiver
receive pkt0, send ack0 receive pkt1, send ack1
receive pkt3, discard, (re)send ack1
receive pkt4, discard,
(re)send ack1 receive pkt5, discard,
(re)send ack1
rcv pkt2, deliver, send ack2 rcv pkt3, deliver, send ack3 rcv pkt4, deliver, send ack4 rcv pkt5, deliver, send ack5
send pkt0 send pkt1 send pkt2 send pkt3
0123
0123
012345678 012345678 01 678 01 678
rcv ack0, send pkt4 rcv ack1, send pkt5
ignore duplicate ACK
pkt 2 timeout
send pkt2 send pkt3 send pkt4 send pkt5
Xloss (wait)
2345
2345
Transport Layer 3-54
Selective repeat
receiver individually acknowledges all correctly received pkts
buffers pkts, as needed, for eventual in-order delivery to upper layer
sender only resends pkts for which ACK not received
sender timer for each unACKed pkt
sender window
N consecutive seq #’s
limits seq #s of sent, unACKed pkts
Transport Layer 3-55
Selective repeat: sender, receiver windows
Transport Layer 3-56
Selective repeat
sender
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 #
receiver
pkt 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
Transport Layer 3-57
Selective repeat in action
sender window (N=4)
012345678 012345678 45678 45678
012345678 012345678
sender
receiver
receive pkt0, send ack0 receive pkt1, send ack1
receive pkt3, buffer, send ack3
receive pkt4, buffer,
send ack4 receive pkt5, buffer,
send ack5
rcv pkt2; deliver pkt2, pkt3, pkt4, pkt5; send ack2
send pkt0 send pkt1 send pkt2 send pkt3
0123
0123
2345
2345
2345
2345
01 678 01 678 01 678 01 678
rcv ack0, send pkt4 rcv ack1, send pkt5
record ack3 arrived
pkt 2 timeout
send pkt2
record ack4 arrived record ack5 arrived
Q: what happens when ack2 arrives?
Xloss (wait)
Transport Layer 3-58
Selective repeat: dilemma
example:
seq #’s: 0, 1, 2, 3 window size=3
receiver sees no difference in two scenarios!
duplicate data accepted as new in (b)
Q: what relationship between seq # size and window size to avoid problem in (b)?
sender window (after receipt)
0123012 0123012 0123012
0123012 0 1 2 3 0 1 2
receiver window (after receipt)
pkt0
pkt1 0123012 pkt2 0123012
pkt3 0123012 X
pkt0 will accept packet
(a) no problem
with seq number 0
receiver can’t see sender side. receiver behavior identical in both cases! something’s (very) wrong!
0 1 2 3 0 1 2 pkt0
0123012 pkt1 0123012 0123012 pkt2 0123012
X 0123012 timeout X
retransmit pkt0 X
0 1 2 3 0 1 2 pkt0 will accept packet
(b) oops!
with seq number 0
Transport Layer 3-59
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
3.6 principles of congestion control
3.7 TCP congestion control
Transport Layer 3-60
TCP: Overview RFCs: 793,1122,1323, 2018, 2581
point-to-point:
one sender, one receiver
reliable, in-order byte steam:
no “message boundaries”
pipelined:
TCP congestion and flow control set window size
full duplex data:
bi-directional data flow
in same connection
MSS: maximum segment size
connection-oriented:
handshaking (exchange of control msgs) inits sender, receiver state before data exchange
flow controlled: sender will not
overwhelm receiver
Transport Layer 3-61
TCP segment structure
32 bits
URG: urgent data (generally not used)
ACK: ACK # valid
PSH: push data now (generally not used)
RST, SYN, FIN: connection estab (setup, teardown commands)
Internet checksum (as in UDP)
counting
by bytes
of data
(not segments!)
# bytes rcvr willing to accept
source port #
sequence number
acknowledgement number
dest port #
head len
not used
checksum
U
A
P
R
S
F
options (variable length)
application data
(variable length)
receive window
Urg data pointer
Transport Layer 3-62
TCP seq. numbers, ACKs
sequence numbers:
byte stream “number” of first byte in segment’s data
acknowledgements:
seq # of next byte expected from other side
cumulative ACK
Q: how receiver handles
out-of-order segments
A: TCP spec doesn’t say, - up to implementor
outgoing segment from sender
source port #
dest port #
sequence number
acknowledgement number
rwnd
checksum
urg pointer
window size N
sender sequence number space
sent ACKed
sent, not- yet ACKed (“in- flight”)
usable not but not usable yet sent
incoming segment to sender
source port #
dest port #
sequence number
acknowledgement number
A
rwnd
checksum
urg pointer
Transport Layer 3-63
TCP seq. numbers, ACKs
Host A Host B
User types ‘C’
host ACKs receipt
of echoed ‘C’
Seq=42, ACK=79, data = ‘C’
host ACKs
Seq=79, ACK=43, data = ‘C’
receipt of ‘C’, echoes back ‘C’
Seq=43, ACK=80
simple telnet scenario
Transport Layer 3-64
TCP round trip time, timeout
Q: how to set TCP timeout value?
longer than RTT but RTT varies
too short: premature timeout, unnecessary retransmissions
too long: slow reaction to segment loss
Q: how to estimate RTT?
SampleRTT: measured time from segment transmission until ACK receipt
ignore retransmissions
SampleRTT will vary, want
estimated RTT “smoother”
average several recent measurements, not just current SampleRTT
Transport Layer 3-65
TCP round trip time, timeout
EstimatedRTT = (1- )*EstimatedRTT + *SampleRTT
exponential weighted moving average
influence of past sample decreases exponentially fast
typical value: = 0.125
RTT: gaia.cs.umass.edu to fantasia.eurecom.fr
350
300
250
200
150
100
1 8 15 22 29 36 43 50 57 64 71 78 85 92 99 106
RTT: gaia.cs.umass.edu to fantasia.eurecom.fr
sampleRTT
EstimatedRTT
time (seconnds)
time (seconds)
Transport Layer 3-66
SampleRTT Estimated RTT
RTT (milliseconds)
RTT (milliseconds)
TCP round trip time, timeout
timeout interval: EstimatedRTT plus “safety margin”
large variation in EstimatedRTT -> larger safety margin estimate SampleRTT deviation from EstimatedRTT:
DevRTT = (1-)*DevRTT + *|SampleRTT-EstimatedRTT|
(typically, = 0.25) TimeoutInterval = EstimatedRTT + 4*DevRTT
estimated RTT
“safety margin” Transport Layer 3-67
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
3.6 principles of congestion control
3.7 TCP congestion control
Transport Layer 3-68
TCP reliable data transfer
TCP creates rdt service on top of IP’s unreliable service
pipelined segments
cumulative acks
single retransmission timer
retransmissions triggered by:
timeout events duplicate acks
let’s initially consider simplified TCP sender:
ignore duplicate acks ignore flow control,
congestion control
Transport Layer 3-69
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 ack acknowledges previously unacked segments
update what is known to be ACKed
start timer if there are still unacked segments
Transport Layer 3-70
TCP sender (simplified)
L
NextSeqNum = InitialSeqNum SendBase = InitialSeqNum
event
ACK received, with ACK field value y
data received from application above
create segment, seq. #: NextSeqNum
pass segment to IP (i.e., “send”) NextSeqNum = NextSeqNum + length(data) if (timer currently not running)
start timer
timeout
retransmit not-yet-acked segment with smallest seq. #
start timer
wait for
if (y > SendBase) {
SendBase = y
/* SendBase–1: last cumulatively ACKed byte */ if (there are currently not-yet-acked segments)
start timer else stop timer
}
Transport Layer 3-71
TCP: retransmission scenarios
Host A Host B
Seq=92, 8 bytes of data
Host A Host B SendBase=92
ACK=100
X
Seq=92, 8 bytes of data
ACK=100
lost ACK scenario
SendBase=100 SendBase=120
SendBase=120
Seq=92, 8 bytes of data Seq=100, 20 bytes of data
ACK=100 ACK=120
Seq=92, 8 bytes of data
ACK=120
premature timeout
Transport Layer 3-72
timeout
timeout
TCP: retransmission scenarios
Host A
Host B
Seq=92, 8 bytes of data
Seq=100, 20 bytes of data
X ACK=100 ACK=120
Seq=120, 15 bytes of data
cumulative ACK
Transport Layer 3-73
timeout
TCP ACK generation [RFC 1122, RFC 2581]
event at receiver
arrival of in-order segment with expected seq #. All data up to expected seq # already ACKed
arrival of in-order segment with expected seq #. One other segment has ACK pending
arrival of out-of-order segment higher-than-expect seq. # . Gap detected
arrival of segment that partially or completely fills gap
TCP receiver action
delayed ACK. Wait up to 500ms
for next segment. If no next segment, send ACK
immediately send single cumulative ACK, ACKing both in-order segments
immediately send duplicate ACK, indicating seq. # of next expected byte
immediate send ACK, provided that segment starts at lower end of gap
Transport Layer 3-74
TCP 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.
TCP fast retransmit
if sender receives 3 ACKs for same data
(“triple duplicate ACKs”), (“triple duplicate ACKs”),
resend unacked segment with smallest seq #
likely that unacked segment lost, so don’t wait for timeout
Transport Layer 3-75
TCP fast retransmit
Host A
Host B
Seq=92, 8 bytes of data
Seq=100, 20 bytes of data
X
ACK=100
ACK=100
ACK=100 ACK=100
Seq=100, 20 bytes of data
fast retransmit after sender receipt of triple duplicate ACK
Transport Layer 3-76
timeout
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
3.6 principles of congestion control
3.7 TCP congestion control
Transport Layer 3-77
TCP flow control
application process
TCP code
TCP socket receiver buffers
IP code
application may remove data from TCP socket buffers ….
… slower than TCP receiver is delivering (sender is sending)
application OS
flow control
receiver controls sender, so sender won’t overflow receiver’s buffer by transmitting too much, too fast
from sender
receiver protocol stack
Transport Layer 3-78
TCP flow control
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
sender limits amount of unacked (“in-flight”) data to receiver’srwnd value
guarantees receive buffer will not overflow
to application process
buffered data
free buffer space
RcvBuffer
rwnd
TCP segment payloads receiver-side buffering
Transport Layer 3-79
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
3.6 principles of congestion control
3.7 TCP congestion control
Transport Layer 3-80
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
network
application
connection state: ESTAB connection Variables:
seq # client-to-server server-to-client
rcvBuffer size at server,client
network
Socket clientSocket = newSocket(“hostname”,”port
Socket connectionSocket = welcomeSocket.accept();
number”);
Transport Layer 3-81
Agreeing to establish a connection
2-way handshake:
Q: will 2-way handshake always work in network?
variable delays
retransmitted messages (e.g. req_conn(x)) due to message loss
message reordering
can’t “see” other side
Let’s talk OK
ESTAB
ESTAB
choose x
ESTAB
req_conn(x) acc_conn(x)
ESTAB
Transport Layer 3-82
Agreeing to establish a connection
2-way handshake failure scenarios:
choose x
retransmit req_conn(x)
ESTAB
client terminates
req_conn(x)
choose x
retransmit req_conn(x)
ESTAB
retransmit data(x+1)
client terminates
req_conn(x)
ESTAB
accept data(x+1)
server forgets x
ESTAB
accept data(x+1)
ESTAB
acc_conn(x)
acc_conn(x) data(x+1)
connection x completes
req_conn(x) data(x+1)
req_conn(x)
connection x completes
server forgets x
ESTAB
half open connection! (no client!)
Transport Layer 3-83
TCP 3-way handshake
client state
LISTEN
SYNSENT
choose init seq num, x send TCP SYN msg
server state
LISTEN
SYN RCVD
SYNbit=1, Seq=x
SYNbit=1, Seq=y ACKbit=1; ACKnum=x+1
ACKbit=1, ACKnum=y+1
choose init seq num, y send TCP SYNACK msg, acking SYN
ESTAB
received SYNACK(x) indicates server is live; send ACK for SYNACK;
this segment may contain client-to-server data
received ACK(y) indicates client is live
ESTAB
Transport Layer 3-84
TCP 3-way handshake: FSM
closed
Socket connectionSocket = welcomeSocket.accept();
SYN(x)
L
listen
Socket clientSocket = newSocket(“hostname”,”port
number”);
SYN(seq=x)
SYN sent
SYNACK(seq=y,ACKnum=x+1) ACK(ACKnum=y+1)
SYNACK(seq=y,ACKnum=x+1) create new socket for
communication back to client
SYN rcvd
ACK(ACKnum=y+1)
L
ESTAB
Transport Layer 3-85
TCP: closing a connection
client, server each close their side of connection send TCP segment with FIN bit = 1
respond to received FIN with ACK
on receiving FIN, ACK can be combined with own FIN
simultaneous FIN exchanges can be handled
Transport Layer 3-86
TCP: closing a connection
client state
ESTAB
server state
ESTAB CLOSE_WAIT
LAST_ACK
CLOSED
clientSocket.close()
FIN_WAIT_1 FIN_WAIT_2
TIMED_WAIT
can no longer send but can receive data
wait for server close
FINbit=1, seq=x ACKbit=1; ACKnum=x+1
FINbit=1, seq=y ACKbit=1; ACKnum=y+1
can still send data
CLOSED
timed wait for 2*max
segment lifetime
can no longer send data
Transport Layer 3-87
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
3.6 principles of congestion control
3.7 TCP congestion control
Transport Layer 3-88
Principles of congestion control
congestion:
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!
Transport Layer 3-89
Causes/costs of congestion: scenario 1
two senders, two receivers
one router, infinite buffers
output link capacity: R no retransmission
original data: lin Host A
throughput: lout unlimited shared
output link buffers
Host B
R/2
lin R/2
lin R/2
maximum per-connection large delays as arrival rate, lin, throughput: R/2 approaches capacity
Transport Layer 3-90
lout
delay
Causes/costs of congestion: scenario 2
one router, finite buffers
sender retransmission of timed-out packet
application-layer input = application-layer output: lin =
lout
transport-layer input includes retransmissions : l
l
‘
in
in
lin : original data
l’in: original data, plus
retransmitted data
lout
Host A
Host B
finite shared output link buffers
Transport Layer 3-91
Causes/costs of congestion: scenario 2
R/2
idealization: perfect knowledge
sender sends only when router buffers available
copy
A
lin R/2 lin : original data
lout
l’in: original data, plus retransmitted data
free buffer space!
Host B
finite shared output link buffers
Transport Layer 3-92
lout
Causes/costs of congestion: scenario 2
Idealization: known loss packets can be lost, dropped at router due to full buffers
sender only resends if packet known to be lost
copy
A
lin : original data
l’in: original data, plus
retransmitted data
no buffer space!
lout
Host B
Transport Layer 3-93
Causes/costs of congestion: scenario 2
Idealization: known loss packets can be lost, dropped at router due to full buffers
sender only resends if packet known to be lost
R/2
lin lin : original data
when sending at R/2, some packets are retransmissions but asymptotic goodput is still R/2 (why?)
R/2
lout
A
l’in: original data, plus retransmitted data
free buffer space!
Host B
Transport Layer 3-94
lout
Causes/costs of congestion: scenario 2
Realistic: duplicates
packets can be lost, dropped
at router due to full buffers
sender times out prematurely, sending two copies, both of which are delivered
lin l’in
R/2
timeout
lin
l
R/2
when sending at R/2, some packets are retransmissions including duplicated that are delivered!
copy
out
A
free buffer space!
Host B
Transport Layer 3-95
lout
Causes/costs of congestion: scenario 2
Realistic: duplicates
packets can be lost, dropped
at router due to full buffers
R/2
sender times out prematurely, sending two copies, both of which are delivered
lin
when sending at R/2, some packets are retransmissions including duplicated that are delivered!
R/2
“costs” of congestion:
more work (retrans) for given “goodput”
unneeded retransmissions: link carries multiple copies of pkt
decreasing goodput
Transport Layer 3-96
lout
Causes/costs of congestion: scenario 3
four senders
multihop paths
timeout/retransmit
Q: what happens as lin and lin’ increase ?
A: as red lin’ increases, all arriving blue pkts at upper queue are dropped, blue throughput g 0
Host A
lin : original data
l’in: original data, plus
retransmitted data
finite shared output link buffers
lout Host B
Host D
Host C
Transport Layer 3-97
Causes/costs of congestion: scenario 3
C/2
lin’
another “cost” of congestion:
when packet dropped, any “upstream transmission capacity used for that packet was wasted!
C/2
Transport Layer 3-98
lout
Approaches towards congestion control
two broad approaches towards congestion control:
end-end congestion network-assisted
control:
no explicit feedback from network
congestion inferred from end-system observed loss, delay
approach taken by TCP
congestion control:
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 3-99
Case study: ATM ABR congestion control
ABR: available bit rate:
“elastic service” if sender’s path
“underloaded”:
sender should use
available bandwidth
if sender’s path congested:
sender throttled to minimum guaranteed rate
RM (resource management) cells:
sent by sender, interspersed with data cells
bits in RM cell set by switches (“network-assisted”)
NI bit: no increase in rate (mild congestion)
CI bit: congestion indication
RM cells returned to sender by receiver, with bits intact
Transport Layer 3-100
Case study: ATM ABR congestion control
RM cell data cell
two-byte ER (explicit rate) field in RM cell
congested switch may lower ER value in cell
senders’ send rate thus max supportable rate on path
EFCI bit in data cells: set to 1 in congested switch
if data cell preceding RM cell has EFCI set, receiver sets CI bit in returned RM cell
Transport Layer 3-101
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
3.6 principles of congestion control
3.7 TCP congestion control
Transport Layer 3-102
TCP congestion control: additive increase multiplicative decrease
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 additively increase window size …
…. until loss occurs (then cut window in half)
AIMD saw tooth behavior: probing for bandwidth
time
Transport Layer 3-103
cwnd: TCP sender congestion window size
TCP Congestion Control: details
sender sequence number space
TCP sending rate:
roughly: send cwnd bytes, wait RTT for ACKS, then send more bytes
cwnd
last byte ACKed
sent, not- yet ACKed (“in- flight”)
last byte sent
sender limits transmission:
cwnd is dynamic, function of perceived network congestion
rate
~ cwnd bytes/sec
~
RTT
LastByteSent- < LastByteAcked
cwnd
Transport Layer 3-104
TCP Slow Start
when connection begins, increase rate exponentially until first loss event:
initially cwnd = 1 MSS
doublecwndeveryRTT
done by incrementing cwnd for every ACK received
summary: initial rate is slow but ramps up exponentially fast
Host A
Host B
time
Transport Layer 3-105
RTT
TCP: detecting, reacting to loss
loss indicated by timeout:
cwnd set to 1 MSS;
window then grows exponentially (as in slow start) to threshold, then grows linearly
loss indicated by 3 duplicate ACKs: TCP RENO dup ACKs indicate network capable of delivering
some segments
cwnd is cut in half window then grows linearly
TCP Tahoe always sets cwnd to 1 (timeout or 3 duplicate acks)
Transport Layer 3-106
TCP: switching from slow start to CA
Q: when should the exponential increase switch to linear?
A: when cwnd gets to 1/2 of its value before timeout.
Implementation:
variable ssthresh
on loss event, ssthresh is set to 1/2 of cwnd just before loss event
Transport Layer 3-107
Summary: TCP Congestion Control
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!
transmit new segment(s), as allowed
duplicate ACK dupACKcount++
slow start
new ACK
.
cwnd = cwnd + MSS dupACKcount = 0
(MSS/cwnd)
L
cwnd = 1 MSS ssthresh = 64 KB dupACKcount = 0
timeout
ssthresh = cwnd/2 cwnd = 1 MSS dupACKcount = 0
retransmit missing segment
dupACKcount == 3
ssthresh= cwnd/2 cwnd = ssthresh + 3 retransmit missing segment
congestion avoidance
duplicate ACK dupACKcount++
New ACK!
New ACK
cwnd = ssthresh dupACKcount = 0
fast recovery
timeout
ssthresh = cwnd/2
cwnd = 1
dupACKcount = 0 retransmit missing segment
dupACKcount == 3
ssthresh= cwnd/2
cwnd = ssthresh + 3 retransmit missing segment
duplicate ACK
cwnd = cwnd + MSS
transmit new segment(s), as allowed
Transport Layer 3-108
TCP throughput
avg. TCP thruput as function of window size, RTT? ignore slow start, assume always data to send
W: window size (measured in bytes) where loss occurs avg. window size (# in-flight bytes) is 3⁄4 W
avg. thruput is 3/4W per RTT
avg TCP thruput = 3 W bytes/sec 4 RTT
W W/2
Transport Layer 3-109
TCP Futures: TCP over “long, fat pipes” 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]:
TCP throughput = 1.22 . MSS RTT L
➜ to achieve 10 Gbps throughput, need a loss rate of L = 2·10-10 – a very small loss rate!
new versions of TCP for high-speed
Transport Layer 3-110
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
bottleneck router capacity R
Transport Layer 3-111
Why is TCP fair?
two competing sessions:
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 3-112
Fairness (more)
Fairness and UDP
multimedia apps often do not use TCP
do not want rate throttled by congestion control
instead use UDP:
send audio/video at constant rate, tolerate packet loss
Fairness, parallel TCP connections
application can open multiple parallel connections between two hosts
web browsers do this
e.g., link of rate R with 9
existing connections:
new app asks for 1 TCP, gets rate R/10
new app asks for 11 TCPs, gets R/2 Transport Layer 3-113
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”
Transport Layer 3-114