1
© From Computer Networking, by Kurose&Ross Transport Layer 3-1
Introduction to Computer Networking
Guy Leduc
Chapter 3 Transport Layer Computer Networking: A
Top Down Approach, 7th edition.
Jim Kurose, Keith RossAddison-Wesley, April
2016
© From Computer Networking, by Kurose&Ross Transport Layer 3-2
Chapter 3: Transport LayerOur 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
2
© From Computer Networking, by Kurose&Ross Transport Layer 3-3
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
© From Computer Networking, by Kurose&Ross Transport Layer 3-4
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: mainly TCP and
UDP
applicationtransportnetworkdata linkphysical
applicationtransportnetworkdata linkphysical
3
© From Computer Networking, by Kurose&Ross Transport Layer 3-5
Transport vs. network layer
❒ network layer: logical communication between hosts
❒ transport layer: logical communication between processes ❍ relies on, enhances,
network layer services
Analogy:❒ host = institute❒ process = office❒ app messages = letters
in envelopes❒ transport protocol =
janitor who demux incoming letters to offices
❒ network-layer protocol = postal service
© From Computer Networking, by Kurose&Ross Transport Layer 3-6
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
applicationtransportnetworkdata linkphysical
networkdata linkphysical
networkdata linkphysical
networkdata linkphysical
networkdata linkphysical
networkdata linkphysical
networkdata linkphysical
networkdata linkphysical
4
© From Computer Networking, by Kurose&Ross 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
© From Computer Networking, by Kurose&Ross Transport Layer 3-8
Multiplexing/demultiplexing
process
socket
use header info to deliverreceived segments to correct socket
demultiplexing at receiver:handle data from multiplesockets, add transport header (later used for demultiplexing)
multiplexing at sender:
transport
application
physicallink
network
P2P1
transport
application
physicallink
network
P4
transport
application
physicallink
network
P3
5
© From Computer Networking, by Kurose&Ross Transport Layer 3-9
How demultiplexing works❒ host receives IP datagrams
❍ each datagram has source and destination IP addresses in its header
❍ each datagram carries one transport-layer segment
❍ each segment has source and destination port numbers in its header
❒ host uses IP addresses (in network header) & port numbers (in transport header) to direct segment to appropriate socket
source port # dest port #
32 bits
applicationdata
(message)
other header fields
Transport (TCP/UDP) segment format(which is the payload of an IP datagram)
© From Computer Networking, by Kurose&Ross Transport Layer 3-10
Connectionless (UDP) demultiplexingrecall: socket created by app
has host-local port #: DatagramSocket mySocket1
= new DatagramSocket(12534);
❒ when UDP receives segment from below:❍ checks destination port # in
segment❍ directs UDP segment to
socket with that port #
recall: when creating datagram to send into UDP socket, app must specify “remote process id”, namely
! 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 destination
Socket interface
6
© From Computer Networking, by Kurose&Ross Transport Layer 3-11
Connectionless demux: exampleDatagramSocket serverSocket = new DatagramSocket (6428);
transport
application
physicallink
network
P3transport
application
physicallink
network
P1
transport
application
physicallink
network
P4
DatagramSocket mySocket1 = new DatagramSocket (5775);
DatagramSocket mySocket2 = new DatagramSocket (9157);
source port: 9157 (set by transport)dest port: 6428 (set by P3)
source port: 6428dest port: 9157
source port: 6428dest port: 5775
source port: 5775dest port: 6428
© From Computer Networking, by Kurose&Ross Transport Layer 3-12
Connection-oriented (TCP) demux❒ Recall: Web servers have
different TCP sockets for each connecting client❍ non-persistent HTTP will
even have different sockets for each request
❒ These many simultaneous TCP sockets are all associated with the same server port #:❍ destination port # is not
enough to direct segment to appropriate socket!
❒ TCP socket must be associated with a TCP connection
❒ TCP socket thus identified by a 4-tuple: ❍ source IP address❍ source port number❍ dest IP address❍ dest port number
❒ Demux: receiver TCP uses all four values to direct segment to appropriate socket
7
© From Computer Networking, by Kurose&Ross Transport Layer 3-13
Connection-oriented demux: example
transport
application
physicallink
network
P1transport
application
physicallink
P4
transport
application
physicallink
network
P2
source IP,port: A,9157dest IP,port: B,80
network
P6P5P3
source IP,port: C,5775dest IP,port: B,80
source IP,port: C,9157dest IP,port: B,80
Three segments destined for IP address B and dest port 80 are demultiplexed
to different sockets thanks to source IP and port #
client: IP address
A
client: IP address C
server: IP address B
© From Computer Networking, by Kurose&Ross Transport Layer 3-14
Connection-oriented demux: example
transport
application
physicallink
network
P1transport
application
physicallink
transport
application
physicallink
network
P2
network
P3
source IP,port: C,5775dest IP,port: B,80
source IP,port: C,9157dest IP,port: B,80
P4
threaded serverclient: IP address A
client: IP address C
server: IP address B
source IP,port: A,9157dest IP,port: B,80
8
© From Computer Networking, by Kurose&Ross Transport Layer 3-15
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
© From Computer Networking, by Kurose&Ross Transport Layer 3-16
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
" UDP use:o multimedia apps (loss
tolerant, rate sensitive)o DNSo SNMP
" reliable transfer over UDP: o add reliability at
application layero application-specific error
recovery!
9
© From Computer Networking, by Kurose&Ross Transport Layer 3-17
UDP: segment header
source port # dest port #
32 bits
applicationdata
(payload)
UDP segment format
length checksum
length, in bytes, of UDP segment, including header
❒ 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
why is there a UDP?
© From Computer Networking, by Kurose&Ross Transport Layer 3-18
UDP checksum
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 ….
Goal: detect “errors” (e.g., flipped bits) in transmitted segment
10
© From Computer Networking, by Kurose&Ross Transport Layer 3-19
Internet checksum: example
example: add two 16-bit integers
1 1 1 1 0 0 1 1 0 0 1 1 0 0 1 1 0 1 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 0 1 0 1 0 0 0 1 0 0 0 1 0 0 0 0 1 1
wraparound
sumchecksum
Note: when adding numbers, a carryout from the most significant bit needs to be added to the result
© From Computer Networking, by Kurose&Ross Transport Layer 3-20
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❍ Alternating bit protocol❍ Pipelined protocols
❒ 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
11
© From Computer Networking, by Kurose&Ross Transport Layer 3-21
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)
© From Computer Networking, by Kurose&Ross Transport Layer 3-22
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)
12
© From Computer Networking, by Kurose&Ross Transport Layer 3-23
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)
© From Computer Networking, by Kurose&Ross Transport Layer 3-24
Reliable data transfer: getting started
sendside
receiveside
rdt_send(): called from above, (e.g., by app.). Passed data to
deliver to receiver upper layer
udt_send(): called by rdt,to transfer packet over
unreliable channel to receiver
rdt_rcv(): called when packet arrives on rcv-side of channel
deliver_data(): called by rdt to deliver data to upper
13
© From Computer Networking, by Kurose&Ross Transport Layer 3-25
Reliable data transfer: getting startedWe’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
state1
state2
event causing state transitionactions taken on state transition
state: when in this “state” next state uniquely determined by next
eventeventactions
© From Computer Networking, by Kurose&Ross Transport Layer 3-26
Rdt1.0: reliable transfer over a reliable channel
❒ assume: 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 packet = make_pkt(data)udt_send(packet)
rdt_send(data)data = extract (packet)deliver_data(data)
Wait for call from
below
rdt_rcv(packet)
sender receiver
14
© From Computer Networking, by Kurose&Ross Transport Layer 3-27
❒ assume: underlying channel may flip bits in packet❍ checksum to detect bit errors
❒ the question: how to recover from errors?
rdt2.0: channel with bit errors
How do humans recover from “errors”during conversation?
© From Computer Networking, by Kurose&Ross Transport Layer 3-28
❍ 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
rdt2.0: channel with bit errors
pkt A
pkt B
pkt B
ACK
NAK
ACK OK
Error
OK
❒ assume: underlying channel may flip bits in packet❍ checksum to detect bit errors
❒ the question: how to recover from errors:
15
© From Computer Networking, by Kurose&Ross Transport Layer 3-29
rdt2.0: FSM specification
Wait for call from
above
sndpkt = make_pkt(data, checksum)udt_send(sndpkt)
data = extract(rcvpkt)deliver_data(data)udt_send(ACK)
rdt_rcv(rcvpkt) && not corrupt(rcvpkt)
rdt_rcv(rcvpkt) && isACK(rcvpkt)
udt_send(sndpkt)
rdt_rcv(rcvpkt) && isNAK(rcvpkt)
udt_send(NAK)
rdt_rcv(rcvpkt) && corrupt(rcvpkt)
Wait for ACK or
NAK
Wait for call from
belowsender
receiverrdt_send(data)
Λ
© From Computer Networking, by Kurose&Ross Transport Layer 3-30
rdt2.0: operation with no errors
Wait for call from
above
snkpkt = make_pkt(data, checksum)udt_send(sndpkt)
extract(rcvpkt,data)deliver_data(data)udt_send(ACK)
rdt_rcv(rcvpkt) && not corrupt(rcvpkt)
rdt_rcv(rcvpkt) && isACK(rcvpkt)
udt_send(sndpkt)
rdt_rcv(rcvpkt) && isNAK(rcvpkt)
udt_send(NAK)
rdt_rcv(rcvpkt) && corrupt(rcvpkt)
Wait for ACK or
NAK
Wait for call from
below
rdt_send(data)
Λ
16
© From Computer Networking, by Kurose&Ross Transport Layer 3-31
rdt2.0: error scenario
Wait for call from
above
snkpkt = make_pkt(data, checksum)udt_send(sndpkt)
extract(rcvpkt,data)deliver_data(data)udt_send(ACK)
rdt_rcv(rcvpkt) && notcorrupt(rcvpkt)
rdt_rcv(rcvpkt) && isACK(rcvpkt)
udt_send(sndpkt)
rdt_rcv(rcvpkt) && isNAK(rcvpkt)
udt_send(NAK)
rdt_rcv(rcvpkt) && corrupt(rcvpkt)
Wait for ACK or
NAK
Wait for call from
below
rdt_send(data)
Λ
© From Computer Networking, by Kurose&Ross Transport Layer 3-32
rdt2.0 has a fatal flaw!What happens if ACK/NAK
corrupted?❒ garbled ACK/NAK detected by
checksum too❒ garbled ACK/NAK discarded❒ but sender doesn’t know what
happened at receiver!❒ can’t just retransmit: possible
duplicate
Handling duplicates: ❒ sender retransmits current
pkt if ACK/NAK garbled❒ sender adds sequence
number to each pkt❒ receiver discards (doesn’t
deliver up) duplicate pkt
Sender sends one packet, then waits for receiver response
stop and waitpkt
same pkt
garbled ACK
Duplicate!
OK
17
© From Computer Networking, by Kurose&Ross Transport Layer 3-33
rdt2.1: sender, handles garbled ACK/NAKs
Wait for call 0 from
above
sndpkt = make_pkt(0, data, checksum)udt_send(sndpkt)
rdt_send(data)
Wait for ACK or NAK 0 udt_send(sndpkt)
rdt_rcv(rcvpkt) && ( corrupt(rcvpkt) || isNAK(rcvpkt) )
sndpkt = make_pkt(1, data, checksum)udt_send(sndpkt)
rdt_send(data)
rdt_rcv(rcvpkt) && not corrupt(rcvpkt) && isACK(rcvpkt)
udt_send(sndpkt)
rdt_rcv(rcvpkt) && ( corrupt(rcvpkt) ||isNAK(rcvpkt) )
rdt_rcv(rcvpkt) && not corrupt(rcvpkt) && isACK(rcvpkt)
Wait for call 1 from
above
Wait for ACK or NAK 1
ΛΛ
© From Computer Networking, by Kurose&Ross Transport Layer 3-34
rdt2.1: receiver, handles garbled ACK/NAKs
Wait for 0 from below
sndpkt = make_pkt(NAK, chksum)udt_send(sndpkt)
rdt_rcv(rcvpkt) && not corrupt(rcvpkt) && has_seq0(rcvpkt)
rdt_rcv(rcvpkt) && not corrupt(rcvpkt) && has_seq1(rcvpkt)
data = extract(rcvpkt)deliver_data(data)sndpkt = make_pkt(ACK, chksum)udt_send(sndpkt)
Wait for 1 from below
rdt_rcv(rcvpkt) && not corrupt(rcvpkt) && has_seq0(rcvpkt) data = extract(rcvpkt)deliver_data(data)sndpkt = make_pkt(ACK, chksum)udt_send(sndpkt)
rdt_rcv(rcvpkt) && corrupt(rcvpkt)
sndpkt = make_pkt(ACK, chksum)udt_send(sndpkt)
rdt_rcv(rcvpkt) && not corrupt(rcvpkt) && has_seq1(rcvpkt)
rdt_rcv(rcvpkt) && corrupt(rcvpkt)
sndpkt = make_pkt(ACK, chksum)udt_send(sndpkt)
sndpkt = make_pkt(NAK, chksum)udt_send(sndpkt)
Note: receiver sends ACK in this case.
Why?
18
© From Computer Networking, by Kurose&Ross Transport Layer 3-35
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 “current” pkt has seq. # of 0 or 1
Receiver:❒ must check if received
packet is duplicate❍ state indicates whether 0
or 1 is expected pkt seq #
© From Computer Networking, by Kurose&Ross Transport Layer 3-36
rdt3.0: channels with errors and loss
New assumption: underlying channel can also lose packets (data or 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
❒ requires countdown timer
pkt(0)
pkt(0)ACK
Timeout
pkt(0)
pkt(0)ACK
ACKTimeout Discard but ACK!
No duplicate
X X
19
Transport Layer 3-37
rdt3.1: actually no need for NAKs!❒ Up to now:
❍ Timer for packet loss❍ NAK for packet errors
❒ Simpler:❍ Timer for both packet loss
and errors!❒ NAK would improve recovery
time, but it’s not our concern here!
pkt(0)
pkt(1)
pkt(1)
ACK
NAK
ACK
rdt3.0: With ACKs & NAKs
pkt(0)
pkt(1)
pkt(1)
ACK
ACK
rdt3.1: With ACKs only
Timeout
OK
OK
Error
OK
OK
Error
Transport Layer 3-38
Flaw in rdt3.1: Delayed packets or ACKs
❒ If pkt (or ACK) just delayed (not lost):❍ retransmission will be duplicate, but use
of seq. #’s already handles this
pkt(0)
pkt(0)
ACK
pkt(0)
pkt(1)ACK
ACKDiscard (no dupl)Resend ACK
❒ However, race conditions are possible between the received ACK and the retransmitted packet!
Discard again,Still expects #1!
B ACKedbut lost!
C ACKedbut lost!
X
Timeout
A
B
C
A received
B not recovered
C not recovered
Note: such race conditions are only possible over full-duplex channels
20
Transport Layer 3-39
rdt3.2: adding seq # in ACKs
❒ receiver must specify seq # of pkt being ACKed
pkt(0)
pkt(0)
ACK(1)
pkt(1)
pkt(1)ACK(0)
ACK(0)Discard (no dupl)Resend ACK
B not ACKed
X
Timeout
A
B
A received
B recoveredTimeout
© From Computer Networking, by Kurose&Ross Transport Layer 3-40
rdt3.2 sendersndpkt = make_pkt(0, data, chksum)udt_send(sndpkt)start_timer
rdt_send(data)
Wait for
ACK0
rdt_rcv(rcvpkt) && ( corrupt(rcvpkt) || isACK(rcvpkt,1) )
Wait for call 1 from
above
sndpkt = make_pkt(1, data, chksum)udt_send(sndpkt)start_timer
rdt_send(data)
rdt_rcv(rcvpkt) && not corrupt(rcvpkt) && isACK(rcvpkt,0)
rdt_rcv(rcvpkt) && ( corrupt(rcvpkt) || isACK(rcvpkt,0) )
rdt_rcv(rcvpkt) && not corrupt(rcvpkt) && isACK(rcvpkt,1)
stop_timerstop_timer
udt_send(sndpkt)start_timer
timeout
udt_send(sndpkt)start_timer
timeout
rdt_rcv(rcvpkt)
Wait for call 0from
above
Wait for
ACK1
Λrdt_rcv(rcvpkt)
ΛΛ
Λ
21
© From Computer Networking, by Kurose&Ross Transport Layer 3-41
sender receiver
rcv pkt(1)
rcv pkt(0)
send ack(0)
send ack(1)
send ack(0)
rcv ack(0)
send pkt(0)
send pkt(1)
rcv ack(1)
send pkt(0)rcv pkt(0)
pkt(0)
pkt(0)
pkt(1)
ack(1)
ack(0)
ack(0)
(a) no loss
sender receiver
rcv pkt(1)
rcv pkt(0)
send ack(0)
send ack(1)
send ack(0)
rcv ack(0)
send pkt(0)
send pkt(1)
rcv ack(1)
send pkt(0)rcv pkt(0)
pkt(0)
pkt(0)
ack(1)
ack(0)
ack(0)
(b) packet loss
pkt(1)X
loss
pkt(1)timeoutresend pkt(1)
rdt3.2 in action: Alternating-Bit Protocol (1969)
© From Computer Networking, by Kurose&Ross Transport Layer 3-42
rdt3.2 in action
rcv pkt(1)send ack(1)
(detect duplicate)
pkt(1)
sender receiver
rcv pkt(1)
rcv pkt(0)
send ack(0)
send ack(1)
send ack(0)
rcv ack(0)
send pkt(0)
send pkt(1)
rcv ack(1)
send pkt(0)rcv pkt(0)
pkt(0)
pkt(0)
ack(1)
ack(0)
ack(0)
(c) ACK loss
ack(1)X
loss
pkt(1)timeout
resend pkt(1)
rcv pkt(1)send ack(1)
(detect duplicate)
pkt(1)
sender receiver
rcv pkt(1)
send ack(0)rcv ack(0)
send pkt(1)
send pkt(0)rcv pkt(0)
pkt(0)
ack(0)
(d) premature timeout/ delayed ACK
pkt(1)timeout
resend pkt(1)
ack(1)
send ack(1)
send pkt(0)rcv ack(1)
ack(1)send pkt(0)rcv ack(1) pkt(0)
rcv pkt(0)send ack(0)ack(0)
22
Transport Layer 3-43
rdt3.2 still incorrect if pkt or ACK reordering is possible
pkt(0)
pkt(0)
ACK(0)
pkt(1)
ACK(0)
ACK(1)
Timeouton pkt(0)
Duplicate!But consideredas new pkt(0)!pkt(0)pkt(0)
ACKedbut lost!
pkt reordering
Solution: Choose timeout so large that when a pkt is retransmitted the sender is sure that the previous copy of this pkt and its ACK have disappeared from the network.Better solution: Use a much larger seq# space (see later).
X
Can be arbitrarily small
© From Computer Networking, by Kurose&Ross Transport Layer 3-44
Performance of rdt3.2: stop-and-wait operation
first packet bit transmitted, t = 0sender receiver
RTT
last packet bit transmitted, t = L / R
first packet bit arriveslast packet bit arrives, send ACK
ACK arrives, send next packet, t = RTT + L / R
€
Usender =LR
RTT + LR=
1
1+R ⋅ RTTL
R . RTT/2 = Bandwidth-delay product= “in-flight” bits
U sender : utilization = fraction of time sender busy sending
23
© From Computer Networking, by Kurose&Ross Transport Layer 3-45
Performance of rdt3.2❒ Was OK over local low-speed networks, but…❒ Example: 1 Gbps link, 15 ms end-to-end prop. delay, 1KB packet:
T transmit = 8 103 bits109 bps = 8 µsec
U sender = .008
30.008 = 0.00027
microseconds
L / R RTT + L / R
=
L (packet length in bits)R (transmission rate, bps) =
❍ 1 KByte packet every 30 msec -> 266.7 kbps throughput over 1 Gbps link
❍ network protocol limits use of physical resources!
© From Computer Networking, by Kurose&Ross Transport Layer 3-46
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❍ Alternating bit protocol❍ Pipelined protocols
❒ 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
24
© From Computer Networking, by Kurose&Ross Transport Layer 3-47
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
© From Computer Networking, by Kurose&Ross Transport Layer 3-48
Pipelining: increased utilization
first packet bit transmitted, t = 0sender receiver
RTT
last bit transmitted, t = L / R
first packet bit arriveslast packet bit arrives, send ACK
ACK arrives, send next packet, t = RTT + L / R
last bit of 2nd packet arrives, send ACKlast bit of 3rd packet arrives, send ACK
U sender = .024
30.008 = 0.0008
microseconds
3 * L / R RTT + L / R
=
3-packet pipelining increases utilization by a factor of 3!
25
© From Computer Networking, by Kurose&Ross Transport Layer 3-49
Pipelined protocols: overview
Go-back-N:❒ Sender can have up to
N unacked packets in pipeline
❒ Receiver only sends cumulative ACKs❍ doesn’t ACK pkt if there’s
a gap❒ Sender has timer for
oldest unacked pkt❍ when timer expires,
retransmit all unacked packets
Selective Repeat:❒ Sender can have up to N
unacked packets in pipeline❒ Receiver sends individual
ACKs for each packet
❒ Sender maintains timer for each unacked packet❍ when timer expires, retransmit
only that unacked packet
© From Computer Networking, by Kurose&Ross Transport Layer 3-50
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❍ Alternating bit protocol❍ Pipelined protocols
• GBN• SR
❒ 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
26
© From Computer Networking, by Kurose&Ross Transport Layer 3-51
Go-Back-N (GBN): sender❒ k-bit seq # in pkt header❒ “window” of up to N, consecutive unacked packets allowed
❒ ACK(n): ACKs all packets up to, including seq # n - “cumulative ACK”❍ may receive duplicate ACKs (see receiver)
❒ single timer: conceptually for the oldest in-flight packet (i.e., sent but unacked packet)
❒ Timeout(n): retransmit packet n and all higher seq # packets in window
© From Computer Networking, by Kurose&Ross Transport Layer 3-52
GBN: sender extended FSM
Wait start_timerudt_send(sndpkt[base])udt_send(sndpkt[base+1])…udt_send(sndpkt[nextseqnum-1])
timeout
rdt_send(data)
if (nextseqnum < base+N) { sndpkt[nextseqnum] = make_pkt(nextseqnum,data,chksum) udt_send(sndpkt[nextseqnum]) if (base == nextseqnum) start_timer nextseqnum++ }else refuse_data(data)
base = getacknum(rcvpkt)+1If (base == nextseqnum) stop_timer else start_timer
rdt_rcv(rcvpkt) && not corrupt(rcvpkt)
base=1nextseqnum=1
rdt_rcv(rcvpkt) && corrupt(rcvpkt)
Λ
Λ
27
© From Computer Networking, by Kurose&Ross Transport Layer 3-53
GBN: receiver extended FSM
❒ ACK-only: always send ACK for correctly-received packet with highest in-order seq #❍ need only remember expectedseqnum
❒ out-of-order packet: ❍ discard (don’t buffer) -> no receiver buffering!❍ re-ACK packet with highest in-order seq #
• will generate duplicate ACKs
Wait
udt_send(sndpkt)default
rdt_rcv(rcvpkt) && not corrupt(rcvpkt) && hasseqnum(rcvpkt,expectedseqnum) data = extract(rcvpkt)deliver_data(data)sndpkt = make_pkt(expectedseqnum,ACK,chksum)udt_send(sndpkt)expectedseqnum++
expectedseqnum=1sndpkt = make_pkt(0,ACK,chksum)
Λ
© From Computer Networking, by Kurose&Ross Transport Layer 3-54
GBN in action
send pkt0send pkt1send pkt2send pkt3
(wait)
sender receiver
receive pkt0, send ack0receive pkt1, send ack1
receive pkt3, discard, (re)send ack1rcv ack0, send pkt4
rcv ack1, send pkt5
pkt 2 timeoutsend pkt2send pkt3send pkt4send pkt5
Xloss
receive pkt4, discard, (re)send ack1receive pkt5, discard, (re)send ack1
rcv pkt2, deliver, send ack2rcv pkt3, deliver, send ack3rcv pkt4, deliver, send ack4rcv pkt5, deliver, send ack5
ignore duplicate ACK
0 1 2 3 4 5 6 7 8
sender window (N=4)
0 1 2 3 4 5 6 7 8 0 1 2 3 4 5 6 7 8 0 1 2 3 4 5 6 7 8
0 1 2 3 4 5 6 7 8 0 1 2 3 4 5 6 7 8
0 1 2 3 4 5 6 7 8 0 1 2 3 4 5 6 7 8 0 1 2 3 4 5 6 7 8 0 1 2 3 4 5 6 7 8
28
Transport Layer 3-55
Maximum window size with GBN❒ If seq# size = K (“The modulo”)❒ Q: What is the maximum window size N to preserve correctness?
❍ Obviously N cannot exceed K, why?❍ Can N be equal to K?
pkt0pkt1pkt2pkt3
Suppose K = 4, N =4
W=[0..3]
XXXX
Duplicates!
All ACKs lost
W=[0]W=[1]W=[2]W=[3]
W=[0]
All packets received in order, and delivered again to app!
resend pkt0resend pkt1resend pkt2resend pkt3
Transport Layer 3-56
GBN: maximum window size
❒ Constraint: window size (N) ≤ seq# size (K) – 1❒ It is necessary and sufficient to achieve correctness
when there is no packet reordering in the network
pkt0pkt1pkt2
Now with K = 4, N =3
W=[0..2]
XXX
resend pkt0resend pkt1resend pkt2
All ACKs lost
W=[0]W=[1]W=[2]
W=[3]
All packets are now discarded!And ACKs sent again
29
© From Computer Networking, by Kurose&Ross Transport Layer 3-57
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❍ Alternating bit protocol❍ Pipelined protocols
• GBN• SR
❒ 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
© From Computer Networking, by Kurose&Ross Transport Layer 3-58
Selective Repeat (SR)
❒ receiver individually acknowledges all correctly received packets❍ buffers packets, as needed, for eventual in-order delivery
to upper layer❒ sender only resends packets for which ACK not
received❍ sender timer for each unacked packet
❒ sender window❍ N consecutive seq #’s❍ again limits seq #s of sent, unacked packets
30
© From Computer Networking, by Kurose&Ross Transport Layer 3-59
SR: sender, receiver windows
Relative positions of both windows
Transport Layer 3-60
Sender:
Receiver:
Initially:
Sender:
Receiver:
When the first packets have been received,but their ACKs are not received (yet):
Are the following relative positions possible? Why?
Sender:
Receiver:
Sender:
Receiver:
Sender:
Receiver:
31
© From Computer Networking, by Kurose&Ross Transport Layer 3-61
Selective repeat (SR)
data from above :❒ if next available seq # in
window, send packettimeout(n):❒ resend packet n, ❒ restart timer (n)ACK(n) in [sendbase,sendbase+N-1]:
❒ mark packet n as received❒ if n is smallest unacked
packet, 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
Sender can lag behind receiver, but at most N packets
© From Computer Networking, by Kurose&Ross Transport Layer 3-62
SR in action
send pkt0send pkt1send pkt2send pkt3
(wait)
sender receiver
receive pkt0, send ack0, deliver pkt0receive pkt1, send ack1 deliver pkt1receive pkt3, buffer, send ack3rcv ack0, send pkt4
rcv ack1, send pkt5
pkt 2 timeoutsend pkt2
Xloss
receive pkt4, buffer, send ack4receive pkt5, buffer, send ack5
rcv pkt2; deliver pkt2,pkt3, pkt4, pkt5; send ack2
record ack3 arrived
0 1 2 3 4 5 6 7 8
sender window (N=4)
0 1 2 3 4 5 6 7 8 0 1 2 3 4 5 6 7 8 0 1 2 3 4 5 6 7 8
0 1 2 3 4 5 6 7 8 0 1 2 3 4 5 6 7 8
0 1 2 3 4 5 6 7 8 0 1 2 3 4 5 6 7 8 0 1 2 3 4 5 6 7 8 0 1 2 3 4 5 6 7 8
record ack4 arrivedrecord ack5 arrived
Q: what happens when ack2 arrives?
32
© From Computer Networking, by Kurose&Ross Transport Layer 3-63
SR: dilemma
assume: ❒ seq #’s: 0, 1, 2, 3❒ window size=3
receiver window(after receipt)
sender window(after ack)
0 1 2 3 0 1 2
0 1 2 3 0 1 20 1 2 3 0 1 2
pkt0pkt1pkt2
0 1 2 3 0 1 2 pkt0
timeoutretransmit pkt0
0 1 2 3 0 1 2
0 1 2 3 0 1 20 1 2 3 0 1 2X
XX
will accept packetwith seq number 0(b) oops!
0 1 2 3 0 1 2
0 1 2 3 0 1 20 1 2 3 0 1 2
pkt0pkt1pkt2
0 1 2 3 0 1 2pkt0
0 1 2 3 0 1 2
0 1 2 3 0 1 20 1 2 3 0 1 2
Xwill accept packetwith seq number 0
0 1 2 3 0 1 2 pkt3
(a) no problem
receiver can’t see sender side.receiver behavior identical in both cases!
something’s (very) wrong!
" 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)?
Transport Layer 3-64
Maximum window size with SR❒ If seq# size = K (“The modulo”)❒ Q: What is the maximum window size N?❒ A: N ≤ K/2
❍ is proven correct when there is no packet reordering
(a) Initial situation with a window of size N=3. (b) After 3 frames have been sent and received but not ACK’ed.
The upper side of the rec window falls in the sending window(c) Initial situation with a window of size N=2. (d) After 2 frames sent and received but not ACK’ed.
The upper side of the rec window does not fall in the sending window
Sender 0 1 2 3 0 1 2 3 0 1 2 3 0 1 2 3
Receiver 0 1 2 3 0 1 2 3 0 1 2 3 0 1 2 3
(a) (b) (c) (d)
Suppose K = 4, N =3 K = 4, but N =2
Retransmissions of received pkts fall
outside rec window
33
Transport Layer 3-65
Maximum window size: general principle❒ Notations:
❍ seq# space = [0..K-1], i.e. modulo K❍ Maximum sender window size = Ns❍ Maximum receiver window size = Nr
❒ Without packet reordering in network, protocol is correct if and only if:❍ Ns + Nr ≤ K
❒ Particular cases:❍ GBN: Nr = 1, Ns ≤ K - 1❍ SR: Ns = Nr ≤ K/2❍ Alternating-bit: K = 2, Ns = Nr = 1
• Special case of both GBN and Selective Repeat
Transport Layer 3-66
GBN, when pkt or ACK reordering possible
pkt0pkt1pkt2
resend pkt0
Duplicate!but consideredas new pkt(0)!
pkt reorderingAck0
pkt3
W=[0..2] W=[0]
W=[1]W=[2]W=[3]
W=[0]
W=[1..3]
With K = 4, N = 3
Solution: Choose timeout so large that when a pkt is retransmitted the sender is sure that the previous copy of this pkt and its ACK have disappeared from the network.Better solution: Use a huge seq# space (K >>), keep N much smaller than K-1, and rely on the underlying network to ensure packets and ACKs do not live too long
34
Transport Layer 3-67
SR, when pkt or ACK reordering possible
pkt0pkt1
resend pkt0
Duplicate!but consideredas new pkt(0)!
pkt reordering
Solution: Choose timeout so large that when a pkt is retransmitted the sender is sure that the previous copy of this pkt and its ACK have disappeared from the network.Better solution: Use a huge seq# space (K >>), keep N much smaller than K/2, and rely on the underlying network to ensure packets and ACKs do not live too long
Ack0
pkt2
W=[0..1] W=[0..1]
W=[1..2]W=[2..3]
W=[3..0]
W=[1..2]
K = 4, N = 2
© From Computer Networking, by Kurose&Ross Transport Layer 3-68
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
35
© From Computer Networking, by Kurose&Ross Transport Layer 3-69
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) inits sender, receiver state before data exchange
❒ flow controlled:❍ sender will not overwhelm
receiver
❒ point-to-point:❍ one sender, one
receiver ❒ reliable, in-order byte
stream:❍ no “message
boundaries”❒ pipelined:
❍ TCP congestion and flow control set window size
© From Computer Networking, by Kurose&Ross Transport Layer 3-70
TCP segment structure
source port # dest port #
32 bits
applicationdata
(variable length)
sequence numberacknowledgement number
Receive windowUrg data pnterchecksum
FSRPAUheadlen
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)
36
© From Computer Networking, by Kurose&Ross Transport Layer 3-71
TCP seq. numbers, ACKssequence numbers:
❍ byte stream “number” of first byte in segment’s data
acknowledgements:❍ seq # of next byte expected
from other side❍ cumulative ACK❍ also the SACK option for
selective ACKsQ: how receiver handles out-of-order segments❍ A: TCP spec doesn’t say,
- up to implementorsource port # dest port #
sequence numberacknowledgement number
checksumrwnd
urg pointer
incoming segment to sender
A
sent ACKed
sent, not-yet ACKed(“in-flight”)
usablebut not yet sent
not usable
window size N
sender sequence number space
source port # dest port #
sequence numberacknowledgement number
checksumrwnd
urg pointer
outgoing segment from sender
© From Computer Networking, by Kurose&Ross Transport Layer 3-72
TCP seq. numbers, ACKs
Usertypes‘C’
host ACKs receipt of echoed‘C’
host ACKs receipt of ‘C’, echoes back ‘C’
simple telnet app scenario
Host BHost A
Seq=42, ACK=79, data = ‘C’
Seq=79, ACK=43, data = ‘C’
Seq=43, ACK=80
37
© From Computer Networking, by Kurose&Ross Transport Layer 3-73
TCP Round Trip Time and TimeoutQ: 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
0 1 2 3 4 5RTT (msec)
T
0
0.1
0.2
0.3
Prob.density
0
0.1
0.2
0.3
Prob.density
0 10 20 30 40 50RTT (msec)
T1 T2
Over a link
Over a network
From Computer Networks, by Tanenbaum © Prentice Hall
© From Computer Networking, by Kurose&Ross Transport Layer 3-74
TCP Round Trip Time and Timeout
EstimatedRTT = (1-α)*EstimatedRTT + α*SampleRTT
❒ Exponential weighted moving averageo Influence of past samples decreases exponentially fasto Weights of samples (backward): α, α(1-α), α(1-α)2, α(1-α)3…
❒ Typical value: α = 0.125
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
38
© From Computer Networking, by Kurose&Ross Transport Layer 3-75
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
RTT
(milli
seco
nds)
RTT: gaia.cs.umass.edu to fantasia.eurecom.fr
sampleRTTEstimatedRTT
time (seconds)
Example RTT estimation:
© From Computer Networking, by Kurose&Ross Transport Layer 3-76
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-β)*DevRTT + β*|SampleRTT-EstimatedRTT|
(typically, β = 0.25)
TimeoutInterval = EstimatedRTT + 4*DevRTT
estimated average RTT
“safety margin”
39
© From Computer Networking, by Kurose&Ross Transport Layer 3-77
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
© From Computer Networking, by Kurose&Ross Transport Layer 3-78
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
40
© From Computer Networking, by Kurose&Ross Transport Layer 3-79
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 (one)
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
© From Computer Networking, by Kurose&Ross Transport Layer 3-80
TCP sender (simplified)
waitfor
event
NextSeqNum = InitialSeqNumSendBase = InitialSeqNum
Λ
create segment, seq. #: NextSeqNumpass segment to IP (i.e., “send”)NextSeqNum = NextSeqNum + length(data) if (timer currently not running) start timer
data received from application above
retransmit not-yet-acked segment with smallest seq. #
start timer
timeout
if (y > SendBase) { SendBase = y /* SendBase–1: last cumulatively ACKed byte */ if (there are currently not-yet-acked segments) start timer else stop timer }
ACK received, with ACK field value y
41
© From Computer Networking, by Kurose&Ross Transport Layer 3-81
TCP: retransmission scenarios
lost ACK scenario
Host BHost A
Seq=92, 8 bytes of data
ACK=100
Seq=92, 8 bytes of data
Xtimeo
ut
ACK=100
premature timeout
Host BHost A
Seq=92, 8 bytes of data
ACK=100
Seq=92, 8bytes of data
timeo
utACK=120
Seq=100, 20 bytes of data
ACK=120
SendBase=100
SendBase=120
SendBase=120
SendBase=92
© From Computer Networking, by Kurose&Ross Transport Layer 3-82
TCP: retransmission scenarios
X
cumulative ACK
Host BHost A
Seq=92, 8 bytes of data
ACK=100
Seq=120, 15 bytes of data
timeo
ut
Seq=100, 20 bytes of data
ACK=120
42
© From Computer Networking, by Kurose&Ross Transport Layer 3-83
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 200msfor 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(if SACK option, also send selective ACK)
Immediately send ACK, provided thatsegment starts at lower end of gap
© From Computer Networking, by Kurose&Ross Transport Layer 3-84
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
if sender receives 4 ACKs for same data(“triple duplicate ACKs”), resend unacked segment with smallest seq #! likely that unacked
segment lost, so don’t wait for timeout
TCP fast retransmit
43
© From Computer Networking, by Kurose&Ross Transport Layer 3-85
X
fast retransmit after sender receipt of triple duplicate ACK
Host BHost A
Seq=92, 8 bytes of data
ACK=100tim
eout
ACK=100
ACK=100ACK=100
TCP fast retransmit
Seq=100, 20 bytes of data
Seq=100, 20 bytes of data
Transport Layer 3-86
Error recovery: TCP vs GBN vs SR
❒ Like GBN❍ TCP has cumulative ACKs❍ TCP has a single retransmission timer
❒ Like SR❍ TCP has receiver buffer❍ TCP only retransmits oldest unacked packet on time-out❍ (TCP has a SACK option, similar to Selective Repeat)
❒ In addition❍ TCP has a Fast retransmit mechanism❍ TCP has delayed ACKs
44
© From Computer Networking, by Kurose&Ross 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
© From Computer Networking, by Kurose&Ross Transport Layer 3-88
TCP flow controlapplication
process
TCP socketreceiver buffers
TCPcode
IPcode
application
OS
receiver protocol stack
application may remove data from
TCP socket buffers ….
… slower than TCP receiver is delivering
(sender is sending)
from sender
receiver controls sender, so sender won’t overflow receiver’s buffer by transmitting too much, too fast
flow control
45
© From Computer Networking, by Kurose&Ross Transport Layer 3-89
TCP flow control
buffered data
free buffer space rwnd
RcvBuffer
TCP segment payloads
to application process ❒ 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’s rwnd value
❒ guarantees receive buffer will not overflow
receiver-side buffering
Transport Layer 3-90
Flow control: improvementsWays to improve performance:
1. Use larger windows:❍ TCP throughput cannot exceed rcwd / RTT❍ 16 bits in TCP header to encode rcwd❍ Max rcwd = 216 bytes, thus max TCP throughput = 64 Kbytes per RTT❍ A TCP option (negotiated during TCP connection establishment) allows to
scale the window size by a factor 2k, with k ≤ 14, thus leading to a max rcwd = 230 bytes (while there are 232 byte numbers)
2. Nagle: Reducing overhead by grouping bytes when sending app writes one byte at a time in socket❍ See next slide
3. Silly window syndrome: reducing overhead when receiving app reads one byte at a time from socket❍ See next slides
46
Transport Layer 3-91
Nagle algorithm❒ When TCP receives data from socket
one byte at the time, or more generally in units much smaller than the TCP Maximum Segment Size (MSS)❍ send small packet immediately, and buffer
all the rest until the outstanding bytes are acknowledged
❍ send other segments only when all previous bytes are acked (or MSS bytes have been buffered)
❒ Useful for Telnet for exampleOtherwise: ❍ 41-byte segments containing 1 byte of
data❍ 40 bytes of TCP/IP header overhead
❒ Naggle can be disabled if needed❍ See Socket options: TCP_NoDelay
Host BHost A
small packet with few bytes received
packet with all buffered bytes
Naggle does not allowto send other bytes(unless MSS is reached)
a few bytes
Possible interference between Nagle and delayed ACKs
❒ Issue: additional delay at end of connection Transport Layer 3-92
Host BHost A
MSS packet, odd #
Assume app in B does not write on socket (which would trigger a pkt transmission with a piggybacked ACK)
Send ACK only after 200ms(delayed ACK)
Assume TCP buffer containsslightly more than 1 MSS
(e.g. at end of connection)
Small packet, even #
Naggle does not allow TCPto send the final bytes
(previous bytes not acked)
“Odd numbered” segment,so delayed ack active
47
Relaxed Naggle❒ To avoid interference with delayed ACK at receiver,
relaxed Naggle will allow to send a packet smaller than MSS if the previous packet was full-size (MSS)
Transport Layer 3-93
Host BHost A
MSS packet, odd #
Delayed ACK enabled,but no consequence asnext segment is receivedimmediately
Assume TCP buffer containsslightly more than 1 MSS
(e.g. at end of connection)
Small packet, even #
Relaxed Naggle allows TCPto send the final bytes
(previous segment was full-size)
© From Computer Networking, by Kurose&Ross Transport Layer 3-94
Silly window syndrome
Receiver’s buffer is full
Application reads 1 byte
Window update segment sent
New byte arrives
Receiver’s buffer is full
Header
Header
Room for one more byte
1 Byte
Still a problemwith applicationsreading one byteat a time
Clark's solution: receiver sends a windowupdate only if the buffer is half empty, or if a full segment can be received
From Computer Networks, by Tanenbaum © Prentice Hall
48
© From Computer Networking, by Kurose&Ross Transport Layer 3-95
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
© From Computer Networking, by Kurose&Ross Transport Layer 3-96
Connection Managementbefore exchanging data, sender/receiver “handshake”:❒ agree to establish connection (each knowing the other willing to
establish connection)❒ agree on connection parameters (MSS, options)
connection state: ESTAB connection variables:
seq # client-to-server server-to-client rcvBuffer size at server,client
application
network
connection state: ESTAB connection Variables:
seq # client-to-server server-to-client rcvBuffer size at server,client
application
network
Socket clientSocket = newSocket("hostname","port
number");
Socket connectionSocket = welcomeSocket.accept();
49
© From Computer Networking, by Kurose&Ross Transport Layer 3-97
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
2-way handshake:
Let’s talk
OKESTAB
ESTAB
choose x req_conn(x)ESTAB
ESTABacc_conn(x)
Agreeing to establish a connection
© From Computer Networking, by Kurose&Ross Transport Layer 3-98
Agreeing to establish a connection2-way handshake failure scenarios:
retransmitreq_conn(x)
ESTAB
req_conn(x)
half open connection!(no client!)
client terminates
serverforgets x
connection x completes
retransmitreq_conn(x)
ESTAB
req_conn(x)
data(x+1)
retransmitdata(x+1)
acceptdata(x+1)
choose xreq_conn(x)
ESTAB
ESTAB
acc_conn(x)
client terminates
ESTAB
choose xreq_conn(x)
ESTABacc_conn(x)
data(x+1) acceptdata(x+1)
connection x completes server
forgets x
50
© From Computer Networking, by Kurose&Ross Transport Layer 3-99
TCP 3-way handshake
ESTAB
SYNbit=1, Seq=yACKbit=1; ACKnum=x+1
choose init seq num, ysend TCP SYNACKmsg, acking SYN
ACKbit=1, ACKnum=y+1
received SYNACK(x+1) indicates server is live;
send ACK for SYNACK;this segment may contain
client-to-server data with Seq=x+1 received ACK(y+1)
indicates client is live;allocate buffer
SYNSENT
ESTAB
SYN RCVD
client state
CLOSED
server state
LISTEN
If server port not open, TCP server sends back a RST segment
SYNbit=1, Seq=x
choose init seq num, xsend TCP SYN msg
without data;start timer for
SYN retransmission
SYN has a seq number: kind of fictitious first byte
Transport Layer 3-100
TCP 3-way handshake is robust
RetransmitSYN(x)
SYN(x)
retransmitdata(x+1)
Reject ACK: y ≠ y’
client terminates
ESTAB
choose x SYN(x)
ESTAB
SYNACK(y, x+1)
ACK(y+1) and data(x+1)
acceptdata(x+1)
connection completes server
forgets x
SYNACK(y’, x+1)
ACK(y+1) and data(x+1)
choose y
choose y’, must not have been used in recent past!(Here “recent” means a maximum packet life time)
Notations:
SYN(x) means SYNbit = 1 and Seq = x
SYNACK(y,x) means SYNbit = 1 and Seq = y and ACKbit = 1 and ACKnum = x
ACK(y) means ACKbit = 1 and ACKnum = y
51
© From Computer Networking, by Kurose&Ross Transport Layer 3-101
TCP: choosing initial 32-bit seq#Choosing x and y:❒ Constraint: x and y not used
recently in a former connection between this client (i.e. same IP and port#) and this server (same IP and port#)
❒ Recently = less than TCP maximum segment lifetime (MSL)
❒ Otherwise a still alive duplicate segment (of SYN, SYNACK, ACK) from a former connection could be confused with this connection
❒ Too difficult to keep track of seq# used in each recent connection
❒ In practice, x and y are picked at random by TCP
client
SYN (seq=x)
server
SYNACK (seq=y, ack=x+1)
ACK (seq=x+1, ack=y+1)
connect listen+ accept
connected
connected
© From Computer Networking, by Kurose&Ross Transport Layer 3-102
TCP 3-way handshake: FSM
closed
L
listen
SYNrcvd
SYNsent
ESTAB
Socket clientSocket = newSocket("hostname","port
number");
SYN(seq=x)
Socket connectionSocket = welcomeSocket.accept();
SYN(x)SYNACK(seq=y,ACKnum=x+1)
create new socket for communication back to client
SYNACK(seq=y,ACKnum=x+1)ACK(ACKnum=y+1)ACK(ACKnum=y+1)
Λ
client sideserver side
52
© From Computer Networking, by Kurose&Ross Transport Layer 3-103
TCP: closing a connection❒ client, server each close their “sending” side of
connection❍ send TCP segment with FIN bit = 1❍ symmetric/graceful release
❒ respond to received FIN with ACK❍ on receiving FIN, ACK can be combined with own FIN
❒ simultaneous FIN exchanges can be handled
© From Computer Networking, by Kurose&Ross Transport Layer 3-104
FIN_WAIT_2
CLOSE_WAIT
FINbit=1, seq=y
ACKbit=1; ACKnum=y+1
ACKbit=1; ACKnum=x+1 wait for server
close
can stillsend data
can no longersend data
LAST_ACK
CLOSED
TIMED_WAIT
timed wait for 2 * max segment lifetime (MSL);
will respond with ACKto received FINs
CLOSED
TCP: closing a connection
FIN_WAIT_1 FINbit=1, seq=xcan no longersend but can receive data
clientSocket.close()
client state server state
ESTABESTAB
FINs have seq# to recover loss of last data bytesACKs are sent when all data received. So there are also retransmissions when FINs are not acked
Last data from client (possibly empty)
53
Transport Layer 3-105
TCP: closing a connection (flaws)❒ Q: What if last FIN and its retransmissions are all lost?
❒ A: Need to disconnect after k FIN transmission attempts❒ Q: But what about the other side then?
❒ A: Need to add a heart beat mechanism to detect other side has closed “without” notification
❒ Q: But what if the heart beat messages are lost?❒ A: Argh, the other side will close, while it shouldn’t!
❒ Last word: no perfect solution (similar to the Two-army problem)
Transport Layer 3-106
Closing a TCP connection: FSM
ESTABFIN
TIMEWAIT
LASTACK
CLOSED
ACKconnectionSocket.close()
ACKclose socket
Wait 2 * MSL
passive closeactive close
CLOSEWAIT
FIN
FINconnectionSocket.close()
ACK
FIN+ACKFINWAIT
2CLOSING
ACKFIN
ACKFIN
Λ
ACK
Λ
ACK
close socket
simultaneousclose
FINWAIT
1
Side that actively closes socket waits a double MSL (safety margin). Therefore same TCP 4-tuple cannot be reused sooner (e.g. with same client port#). It ensures that no duplicate segment from an earlier connection with the same 4-tuple can jump in the current connection. But this may lead to some starvation of resources.
All pending data are still sent reliably
After a socket “close”, TCP continues to send
the previous bytes reliably
54
© From Computer Networking, by Kurose&Ross Transport Layer 3-107
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 3-108
Principles of Congestion Control
❒ 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 (queuing in router buffers)
❒ a top-10 problem!
Transmissionrate adjustment
Transmissionnetwork
Small-capacityreceiver
Large-capacityreceiver
InternalcongestionNeed
flowcontrol
Needcongestion
control
From Computer Networks, by Tanenbaum © Prentice Hall
55
© From Computer Networking, by Kurose&Ross Transport Layer 3-109
Causes/costs of congestion: scenario 1
❒ two senders, two receivers
❒ one router, infinite buffers
❒ output link capacity: R❒ no retransmission
maximum per-connection throughput: R/2
unlimited shared output link buffers
Host A
original data: λin
Host B
throughput: λout
R/2
R/2
λ out
λin R/2 de
lay
λin large delays as arrival rate, λin,
approaches capacity
R
© From Computer Networking, by Kurose&Ross Transport Layer 3-110
❒ one router, finite buffers ❒ sender retransmission of timed-out packet
❍ application-layer input = application-layer output: λin = λout • “goodput”
❍ transport-layer input includes retransmissions : λ’in ≥ λin
finite shared output link buffers
Host A
λin : original data
Host B
λout λ'in: original data, plus retransmitted data
Causes/costs of congestion: scenario 2
56
© From Computer Networking, by Kurose&Ross Transport Layer 3-111
Idealization 1: perfect knowledge= sender sends only when router buffers available
finite shared output link buffers
λin : original data λout λ'in: original data, plus
retransmitted data copy
free buffer space!
R/2
R/2
λ out
λin
Causes/costs of congestion: scenario 2
Host B
Host A R
© From Computer Networking, by Kurose&Ross Transport Layer 3-112
λin : original data λout λ'in: original data, plus
retransmitted data copy
no buffer space!
Idealization 2: known loss - packets can be lost, dropped at router due to full buffers- sender only resends if packet known to be lost
Causes/costs of congestion: scenario 2
Host A
Host B
57
© From Computer Networking, by Kurose&Ross Transport Layer 3-113
λin : original data λout λ'in: original data, plus
retransmitted data
free buffer space!
Causes/costs of congestion: scenario 2 Idealization 2:
known lossR/2
R/2 λin
λ out
when sending at R/2, some packets are retransmissions but asymptotic goodput is still R/2 (why?)
Host A
Host B
R
© From Computer Networking, by Kurose&Ross Transport Layer 3-114
Host A
λin λout λ'in copy
free buffer space!
timeout
R/2
R/2 λin
λ out
When sending at R/2, some packets are retransmissions including duplicates that are delivered! Asymptotic goodput now < R/2
Host B
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
Causes/costs of congestion: scenario 2
58
© From Computer Networking, by Kurose&Ross Transport Layer 3-115
R/2
λ out
when sending at R/2, some packets are retransmissions including duplicates that are delivered!
“costs” of congestion: # more work (retransmissions) for given “goodput”# unneeded retransmissions: link carries multiple copies of
packet! decreasing goodput
R/2 λin
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
© From Computer Networking, by Kurose&Ross Transport Layer 3-116
❒ four senders❒ multihop paths❒ timeout/retransmit
Q: what happens as λin and λ’in increase ?
finite shared output link buffers
Host A λout
Causes/costs of congestion: scenario 3
Host B
Host C Host D
λin : original data λ'in: original data, plus
retransmitted data
A: as red λ’in increases, all arriving blue packets at upper queue are dropped, blue throughput ! 0
59
© From Computer Networking, by Kurose&Ross Transport Layer 3-117
another “cost” of congestion: when packet dropped, any “upstream” transmission capacity used for that packet was wasted!
Causes/costs of congestion: scenario 3
R/2
R/2
λ out
λ’in
© From Computer Networking, by Kurose&Ross Transport Layer 3-118
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 for sender to send at (ATM ABR)
❒ not studied in this course
Two broad approaches towards congestion control:
60
© From Computer Networking, by Kurose&Ross Transport Layer 3-119
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
© From Computer Networking, by Kurose&Ross Transport Layer 3-120
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❍ note: loss may also be due to bit errors (e.g. on wireless
links)
61
© From Computer Networking, by Kurose&Ross Transport Layer 3-121
TCP congestion control: additive increase multiplicative decrease
# approach: sender increases transmission rate (congestion window size, cwnd), probing for usable bandwidth, until loss occurs! additive increase: increase cwnd by 1 MSS (Maximum Segment
Size, not counting header) every RTT until loss detected! multiplicative decrease: cut cwnd in half after loss
cwnd
: TC
P se
nder
co
nges
tion
win
dow
siz
e
AIMD saw toothbehavior: probing
for bandwidth
additively increase window size ……. until loss occurs (then cut window in half)
time
© From Computer Networking, by Kurose&Ross Transport Layer 3-122
TCP Congestion Control: details
❒ sender limits transmission:
❒ cwnd is dynamic, function of perceived network congestion
❒ sender limited to min (cwnd, rwnd)
TCP sending rate:❒ roughly: send cwnd
bytes, wait RTT for ACKs, then send more bytes
last byte ACKed sent, not-
yet ACKed (“in-flight”)
last byte sent
cwnd
LastByteSent - LastByteAcked
< cwnd
sender sequence number space
rate ~ ~ cwnd
RTT bytes/sec
cwndbytes
RTT
ACK(s)
62
© From Computer Networking, by Kurose&Ross Transport Layer 3-123
TCP Slow Start ❒ 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❒ summary: initial rate is slow
but ramps up exponentially fast
Host A
one segment
RTT
Host B
time
two segments
four segments
1
2
3 4
cwnd
© From Computer Networking, by Kurose&Ross Transport Layer 3-124
TCP: detecting, reacting to loss❒ TCP RENO: ❒ If loss indicated by 3 duplicate ACKs (mild congestion):
❍ dup ACKs indicate network capable of delivering some segments ❍ Note: can only work if cwnd ≥ 4 MSS❍ cwnd is cut in half, window then grows linearly
❒ Otherwise, loss indicated by timeout (severe congestion):❍ cwnd set to 1 MSS; ❍ window then grows exponentially (as in slow start) to threshold, then
grows linearly
❒ TCP Tahoe (earlier TCP version) sets cwnd to 1 in both cases (timeout or 3 duplicate acks)
63
© From Computer Networking, by Kurose&Ross Transport Layer 3-125
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❒ linear growth:
❍ cwnd = cwnd + (MSS/cwnd)MSS, for each ACK received
TCP: switching from slow start to CA
© From Computer Networking, by Kurose&Ross Transport Layer 3-126
Summary: TCP Congestion Control
timeout ssthresh = cwnd/2
cwnd = 1 MSS dupACKcount = 0
retransmit missing segment
Λcwnd > ssthresh
congestion avoidance
cwnd = cwnd + MSS (MSS/cwnd) dupACKcount = 0
transmit new segment(s), as allowed
new ACK .
dupACKcount++ duplicate ACK
fast recovery
cwnd = cwnd + MSS transmit new segment(s), as allowed
duplicate ACK
ssthresh= cwnd/2 cwnd = ssthresh + 3 MSS
retransmit missing segment
dupACKcount == 3
timeout ssthresh = cwnd/2 cwnd = 1 MSS dupACKcount = 0 retransmit missing segment
ssthresh= cwnd/2 cwnd = ssthresh + 3 MSS retransmit missing segment
dupACKcount == 3 cwnd = ssthresh dupACKcount = 0
New ACK
slow start
timeout ssthresh = cwnd/2
cwnd = 1 MSS dupACKcount = 0
retransmit missing segment
cwnd = cwnd+MSS dupACKcount = 0 transmit new segment(s), as allowed
new ACK dupACKcount++ duplicate ACK
Λcwnd = 1 MSS
ssthresh = 64 KB dupACKcount = 0
New ACK!
New ACK!
New ACK!
64
Transport Layer 3-127
TCP “goodput” in steady state❒ Goodput = not counting overhead and retransmitted bytes❒ What’s the average goodput of TCP as a function of window size and RTT?
❍ Ignoring slow start❒ Let W be the window size (measured in MSS) when losses occur❒ When window is W, goodput is W/RTT❒ Just after loss, window drops to W/2, goodput to 0.5 W/RTT ❒ Average goodput: 0.75 W/RTT
Win
dow
siz
e in
MSS W
W/2
W/2 . RTT
3W/4 . W/2MSS percycle
Transport Layer 3-128
TCP goodput in steady state (2)
❒ Average window size (in MSS) = 3W/4❒ Number of MSS per cycle = 3W/4 . W/2 = 3W2/8 = 1/p
❍ Where p is the packet loss ratio• One packet loss per cycle! (if p small enough)
❍ So
❒ Average goodput (in MSS/sec) = aver. window / RTT = 3W / 4RTT
❒ Average TCP goodput (in bps)
€
W = 83p
= 32
1RTT p
€
= 32
MSSRTT p
65
© From Computer Networking, by Kurose&Ross Transport Layer 3-129
TCP Futures: TCP over “long, fat pipes”
❒ Example: 1500 byte segments, 100ms RTT, want 10 Gbps throughput
❒ Requires average window size W = 83333 in-flight segments!
❒ Throughput in terms of loss rate:
❒ Needs packet loss rate p = 2 . 10-10 Very small !❒ New versions of TCP for high-speed needed!
€
1.22 ⋅ MSSRTT p
© From Computer Networking, by Kurose&Ross Transport Layer 3-130
Fairness goal: if K TCP sessions share same bottleneck link of bandwidth R, each should have average rate of R/K
TCP Fairness
TCP connection 1
bottleneckrouter
capacity RTCP connection 2
66
© From Computer Networking, by Kurose&Ross Transport Layer 3-131
Why is TCP fair?Two competing sessions having the same RTT:❒ Additive increase gives slope of 1, as throughout increases❒ Multiplicative decrease decreases throughput proportionally
R
R
equal bandwidth share
Connection 1 throughput
Con
nect
ion
2 th
roug
hput
1. congestion avoidance: additive increase2. loss: decrease window by factor of 2
3. congestion avoidance: additive increase4. loss: decrease window by factor of 2
Transport Layer 3-132
And when RTTs are different?❒ If RTT of connection 2 = 2 x RTT of connection 1❒ Connection 1 ramps up twice more quickly
R
R
bandwidth share inversely
proportional to RTT
Connection 1 throughput
Con
nect
ion
2 th
roug
hput
67
© From Computer Networking, by Kurose&Ross Transport Layer 3-133
Fairness (more)Fairness and UDP❒ Some multimedia apps
do not use TCP❍ do not want rate throttled
by congestion control❒ Instead use UDP:
❍ pump audio/video at constant rate, tolerate packet loss
❒ UDP is not “TCP friendly”
Fairness and parallel TCP connections
❒ Application can open multiple parallel connections between 2 hosts❍ e.g. Web browsers may do this
❒ Example: link of rate R supporting 9 connections; ❍ if new app opens 1 TCP,
it gets rate R/10❍ if new app opens 10 // TCPs,
it gets more than R/2 !
So ensuring fairness among TCP flows is not enough
Transport Layer 3-134
Chapter 3: Summary❒ principles behind transport layer
services:❍ multiplexing, demultiplexing❍ reliable data transfer❍ flow control❍ congestion control❍ connection management
❒ instantiation and implementation in the Internet❍ UDP❍ TCP
To probe further:❒ other congestion control
algorithms (e.g. CUBIC, BBR)
❒ MPTCP (Multipath TCP)❒ Google’s application
layer transport (QUIC) over UDP
Next:❒ leaving the network
“edge” (application, transport layers)
❒ into the network “core”