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Transport Layer 3-1 Chapter 3: Transport Layer. Transport Layer 3-2 Chapter 3: Transport Layer our goals:  understand principles behind transport layer.

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Presentation on theme: "Transport Layer 3-1 Chapter 3: Transport Layer. Transport Layer 3-2 Chapter 3: Transport Layer our goals:  understand principles behind transport layer."— Presentation transcript:

1 Transport Layer 3-1 Chapter 3: Transport Layer

2 Transport Layer 3-2 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

3 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

4 Transport Layer 3-4 Transport services and protocols  provide logical (instead of physical!) 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 logical end-end transport application transport network data link physical

5 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 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 collect from/distribute to in-house siblings each week => give to/get from postal service mail carrier  network-layer protocol = postal service household analogy:

6 Transport Layer 3-6 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

7 Transport Layer 3-7 Multiplexing/demultiplexing process socket use header info to deliver received segments to correct socket (Bill receives batch of mails from carrier and delivers to his brothers and sister) demultiplexing at receiver: handle data from multiple sockets, add transport header (later used for demultiplexing) multiplexing at sender: transport application physical link network P2P1 transport application physical link network P4 transport application physical link network P3

8 Transport Layer 3-8 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 source port #dest port # 32 bits application data (payload) other header fields TCP/UDP segment format Remark: we call packets of the network layer DATAGRAMS and packets of the transport layer are called SEGMENTS

9 Transport Layer 3-9 Connectionless demultiplexing  when host receives UDP segment:  checks destination port # in segment  directs UDP segment to socket with that port #  recall: when creating datagram to send into UDP socket, one must specify  destination IP address  destination port #  port number is typically assigned autom. in client app IP datagrams with same dest. port # and same destination IP address, but different source IP addresses and/or source port numbers will be directed to same socket at destination

10 Transport Layer 3-10 Connectionless demux: example P1: port number 6428 transport application physical link network P3 transport application physical link network P1 transport application physical link network P4 P4: port number 5775 P3: port number 9157 UDP segm source port: 9157 dest port: 6428 UDP segm source port: 6428 dest port: 9157 UDP segm source port: 6428 dest port: 5775 UDP segm source port: 5775 dest port: 6428

11 Transport Layer 3-11 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

12 Transport Layer 3-12 Connection-oriented demux: example transport application physical link network P1 transport application physical link P4 transport application physical link network P2 TCP segm source IP,port: A,9157 dest IP, port: B,80 TCP segm source IP,port: B,80 dest IP,port: A,9157 host: IP address A host: IP address C network P6 P5 P3 TCP segm source IP,port: C,5775 dest IP,port: B,80 TCP segm source IP,port: C,9157 dest IP,port: B,80 three segments, all destined to IP address: B, dest port: 80 are demultiplexed to different sockets (different source IP/source port number!); one process per connection socket server: IP address B

13 Transport Layer 3-13 Connection-oriented demux: example transport application physical link network P1 transport application physical link transport application physical link network P2 TCP segm source IP,port: A,9157 dest IP, port: B,80 TCP segm source IP,port: B,80 dest IP,port: A,9157 host: IP address A host: IP address C network P3 TCP segm source IP,port: C,5775 dest IP,port: B,80 TCP segm source IP,port: C,9157 dest IP,port: B,80 three segments, all destined to IP address: B, dest port: 80 are demultiplexed to different sockets (different source IP/source port number!); one THREAD per connection socket server: IP address B P4 threaded server

14 Transport Layer 3-14 Connection-oriented demux: example transport application physical link network P1 transport application physical link transport application physical link network P2 TCP segm source IP,port: A,9157 dest IP, port: B,80 TCP segm source IP,port: B,80 dest IP,port: A,9157 host: IP address A host: IP address C network P3 TCP segm source IP,port: C,5775 dest IP,port: B,80 TCP segm source IP,port: C,9157 dest IP,port: B,80 Note: during persisent HTTP, same server socket it used during non-persistent HTTP, new TCP connection (=> new socket) for every request/response server: IP address B P4 threaded server

15 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

16 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:  streaming multimedia (loss tolerant, rate sensitive)  DNS  Internet phone  real-time video conferencing  reliable transfer over UDP:  add reliability at application layer  application-specific error recovery!

17 Transport Layer 3-17 UDP: segment header source port #dest port # 4 x 16 bits = 4 x 2 bytes application data (payload) UDP segment format length checksum length: in bytes of UDP segment, including header; max. 16 bits => length of data field <2^16=65536 bytes (minus 8 bytes (UDP header), 20 bytes (IP header – not yet prepended!)  no connection establishment which can add delay (e.g. important for DNS)  simple: no connection state at sender, receiver (server can support more active clients)  small header size (UDP: 8 bytes, TCP: 20 bytes)  no congestion control: UDP can blast away as fast as desired why is there a UDP?

18 Transport Layer 3-18 UDP checksum sender:  add IP pseudo header (transport layer must inform network layer about destination => certain information known); later ‘real’ IP header is generated  treat segment contents, including header fields, as sequence of 16-bit integers  checksum: addition of segment contents and one’s complement  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 => pass damaged segment to app with warning or discard it  YES - no error detected. But maybe errors nonetheless? More later …. Goal: detect “errors” (e.g., flipped bits) in transmitted segment

19 Transport Layer 3-19 Internet checksum: example example: add two 16-bit integers wraparound sum checksum Note: when adding numbers, a carryout from the most significant bit needs to be added to the result RFC 768

20 Transport Layer 3-20 Internet checksum Why do we have to add the carry from the most significant bit? 4-Bit example: we want to add -5 and -3 in 1’s complement arithmetic +5 is 0101 => -5 is is 0011 => -3 is ’ 1100 = (ignoring the carry) 0110 (which is -9) = (with carry) 0111 (which is -8)

21 Transport Layer 3-21 Internet checksum Why is the checksum calculated like this? Use notation [a,b] for 16-bit integer a ⋅ 2 8 +b, where a and b are bytes checksum computation corresponds to [A,B] +' [C,D] +'... +' [Y,Z] where +' is1's complement addition 1’s complement addition has nice properties: - associative property: ( [A,B] +' [C,D] +'... +' [J,0] ) +' ( [0,K] +'... +' [Y,Z] ) - commutative property: ( [0,K] +'... +' [Y,Z] ) +’ ( [A,B] +' [C,D] +'... +' [J,0] ) - byte order independence: [B,A] +' [D,C] +'... +' [Z,Y] - allows parallel summation (efficient implementation: see RFC 1071)

22 Transport Layer 3-22 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

23 Transport Layer 3-23 Reliable data transfer protocols send side receive side rdt_send(): called from above, (e.g., by app.). Passes data to deliver to receiver at 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 (receive function of the abstract rdt protocol) deliver_data(): called by rdt to deliver data to upper imagine the following (abstract) data transfer protocol: some abstract upper layer

24 Transport Layer 3-24 we’ll:  incrementally develop sender and 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 behavior of sender and receiver state 1 state 2 event causing state transition actions taken on state transition state: when in this “state” next state uniquely determined by next event event actions Reliable data transfer: getting started

25 Transport Layer 3-25 rdt1.0: reliable transfer over a reliable channel  underlying channel perfectly reliable  no bit errors  no loss of packets  separate FSMs for sender and 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) extract (packet,data) deliver_data(data) Wait for call from below rdt_rcv(packet) sender receiver

26 Transport Layer 3-26  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  receiver feedback: control msgs (ACK,NAK) rcvr- >sender rdt2.0: channel with bit errors How do humans recover from “errors” during conversation?

27 Transport Layer 3-27  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 rdt2.0: channel with bit errors

28 Transport Layer 3-28 rdt2.0: FSM specification Wait for call from above sndpkt = 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 sender receiver rdt_send(data)  stop and wait sender sends one packet, then waits for receiver response

29 Transport Layer 3-29 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

30 Transport Layer 3-30 rdt2.1: sender handles corrupted 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) && notcorrupt(rcvpkt) && isACK(rcvpkt) udt_send(sndpkt) rdt_rcv(rcvpkt) && ( corrupt(rcvpkt) || isNAK(rcvpkt) ) rdt_rcv(rcvpkt) && notcorrupt(rcvpkt) && isACK(rcvpkt) Wait for call 1 from above Wait for ACK or NAK 1   sequence numbers 0 and 1

31 Transport Layer 3-31 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) && notcorrupt(rcvpkt) && has_seq1(rcvpkt) extract(rcvpkt,data) deliver_data(data) sndpkt = make_pkt(ACK, chksum) udt_send(sndpkt) Wait for 1 from below rdt_rcv(rcvpkt) && notcorrupt(rcvpkt) && has_seq0(rcvpkt) extract(rcvpkt,data) 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) rdt2.1: receiver, handles garbled ACK/NAKs

32 Transport Layer 3-32 rdt2.1: discussion sender:  seq # added to pkt  two seq. #’s (0,1) will suffice since receiver can distinguish detect retransmission  must check if received ACK/NAK corrupted  twice as many states  state must “remember” whether “expected” ACK is for 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

33 Transport Layer 3-33 rdt2.2: a NAK-free protocol  same functionality as rdt2.1, using ACKs only  instead of NAK, receiver sends ACK for last packet received OK (=> may ACK same paket several times)  receiver must explicitly include seq # of packet being ACKed  duplicate ACK at sender results in same action as NAK: retransmit current packet

34 Transport Layer 3-34 rdt2.2: sender, receiver fragments Wait for call 0 from above sndpkt = make_pkt(0, data, checksum) udt_send(sndpkt) rdt_send(data) udt_send(sndpkt) rdt_rcv(rcvpkt) && ( corrupt(rcvpkt) || isACK(rcvpkt,1) ) rdt_rcv(rcvpkt) && notcorrupt(rcvpkt) && isACK(rcvpkt,0) Wait for ACK 0 sender FSM fragment rdt_rcv(rcvpkt) && notcorrupt(rcvpkt) && has_seq1(rcvpkt) extract(rcvpkt,data) deliver_data(data) sndpkt = make_pkt(ACK1, chksum) udt_send(sndpkt) Wait for 0 from below rdt_rcv(rcvpkt) && (corrupt(rcvpkt) || has_seq1(rcvpkt)) udt_send(sndpkt) receiver FSM fragment 

35 Transport Layer 3-35 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

36 Transport Layer 3-36 rdt3.0 sender sndpkt = make_pkt(0, data, checksum) 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, checksum) udt_send(sndpkt) start_timer rdt_send(data) rdt_rcv(rcvpkt) && notcorrupt(rcvpkt) && isACK(rcvpkt,0) rdt_rcv(rcvpkt) && ( corrupt(rcvpkt) || isACK(rcvpkt,0) ) rdt_rcv(rcvpkt) && notcorrupt(rcvpkt) && isACK(rcvpkt,1) stop_timer udt_send(sndpkt) start_timer timeout udt_send(sndpkt) start_timer timeout rdt_rcv(rcvpkt) Wait for call 0from above Wait for ACK1  rdt_rcv(rcvpkt)    receiver FSM is homework (do it at home and compare your solution to our solution during the tutorial)

37 Transport Layer 3-37 sender receiver rcv pkt1 rcv pkt0 send ack0 send ack1 send ack0 rcv ack0 send pkt0 send pkt1 rcv ack1 send pkt0 rcv pkt0 pkt0 pkt1 ack1 ack0 (a) no loss sender receiver rcv pkt1 rcv pkt0 send ack0 send ack1 send ack0 rcv ack0 send pkt0 send pkt1 rcv ack1 send pkt0 rcv pkt0 pkt0 ack1 ack0 (b) packet loss pkt1 X loss pkt1 timeout resend pkt1 rdt3.0 in action

38 Transport Layer 3-38 rdt3.0 in action (slide is repaired) rcv pkt1 send ack1 (detect duplicate) pkt1 sender receiver rcv pkt1 rcv pkt0 send ack0 send ack1 send ack0 rcv ack0 send pkt0 send pkt1 rcv ack1 send pkt0 rcv pkt0 pkt0 ack1 ack0 (c) ACK loss ack1 X loss pkt1 timeout resend pkt1 rcv pkt1 send ack1 (detect duplicate) pkt1 sender receiver rcv pkt1 send ack0 rcv ack0 send pkt1 send pkt0 rcv pkt0 pkt0 ack0 (d) premature timeout/ delayed ACK pkt1 timeout resend pkt1 ack1 send ack1 send pkt1 rcv ack0 pkt1 ack1 send pkt0 rcv ack1 pkt0 rcv pkt0 send ack0 ack0 rcv pkt1 send ack1

39 Transport Layer 3-39 Performance of rdt3.0  rdt3.0 is correct, but performance is slow  e.g.: 1 Gbps link (10 9 bits/sec), from US west to east coast 15 ms prop. delay (RRT=30 ms=30/1000 sec), 8000 bit packet: Calculate U sender : utilization – fraction of time sender busy sending D trans = L R 8000 bits 10 9 bits/sec ==.008 ms

40 Transport Layer 3-40 rdt3.0: stop-and-wait operation first packet bit transmitted, t = 0 senderreceiver RTT last packet bit transmitted, t = L / R first packet bit arrives last packet bit arrives, send ACK ACK arrives, send next packet, t = RTT + L / R assume no transmission delay for ACK percentage of time the server is busy (with transmitting)

41 Transport Layer 3-41 Performance of rdt3.0  rdt3.0 is correct, but performance is slow  e.g.: 1 Gbps link (10 9 bits/sec), from US west to east coast 15 ms prop. delay (RRT=30 ms=30/1000 sec), 8000 bit packet:  U sender : utilization – fraction of time sender busy sending  throughput = utilization x net bitrate = bits/sec  ≈ 34kB/sec throughput over 1 Gbps link  receiver has very slow download rate from that server! D trans = L R 8000 bits 10 9 bits/sec ==.008 ms

42 Transport Layer 3-42 Pipelined protocols pipelining: sender allows multiple, yet-to-be- acknowledged packets  range of sequence numbers must be increased  buffering at sender and/or receiver  two generic forms of pipelined protocols: go-Back-N, selective repeat

43 Transport Layer 3-43 Pipelining: increased utilization first packet bit transmitted, t = 0 senderreceiver RTT last bit transmitted, t = L / R first packet bit arrives last packet bit arrives, send ACK ACK arrives, send next packet, t = RTT + L / R last bit of 2 nd packet arrives, send ACK last bit of 3 rd packet arrives, send ACK 3-packet pipelining increases utilization by a factor of 3! maximal 3 unacknowledged packets in pipeline

44 Transport Layer 3-44 Pipelined protocols: overview Go-back-N:  sender can have up to N unack’ ed packets in pipeline  receiver only sends ‘cumulative’ ack  doesn’t ack packet if there’s a gap  sender has timer for oldest unack’ed 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

45 Transport Layer 3-45 Go-Back-N protocol: sender  k-bit seq # in packet header  “window” of up to N, consecutive unack’ed packets allowed  ACK(n): ACKs all pkts up to, including seq # n - “cumulative ACK”  may receive duplicate ACKs (see receiver)  monitor timer for oldest in-flight pkt  timeout(n): retransmit packet n and all higher seq # pkts in window

46 Transport Layer 3-46 GBN: sender extended FSM (slide is repaired) Wait restart_timer(base) udt_send(sndpkt[base]) restart_timer(base+1) udt_send(sndpkt[base+1]) … restart_timer(nextseqnum-1) udt_send(sndpkt[nextseqnum-1]) Timeout(base) rdt_send(data) if (nextseqnum < base+N) { sndpkt[nextseqnum] = make_pkt(nextseqnum,data,chksum) udt_send(sndpkt[nextseqnum]) start_timer(nextseqnum) nextseqnum++ } else refuse_data(data) base = getacknum(rcvpkt)+1 (drop all timers with # < base) rdt_rcv(rcvpkt) && notcorrupt(rcvpkt) base=1 nextseqnum=1 rdt_rcv(rcvpkt) && corrupt(rcvpkt)  

47 Transport Layer 3-47 Wait udt_send(sndpkt) default rdt_rcv(rcvpkt) && notcurrupt(rcvpkt) && hasseqnum(rcvpkt,expectedseqnum) extract(rcvpkt,data) deliver_data(data) sndpkt = make_pkt(expectedseqnum,ACK,chksum) udt_send(sndpkt) expectedseqnum++ expectedseqnum=1 sndpkt = make_pkt(0,ACK,chksum)  GBN: receiver extended FSM

48 Transport Layer 3-48 Wait udt_send(sndpkt) default rdt_rcv(rcvpkt) && notcurrupt(rcvpkt) && hasseqnum(rcvpkt,expectedseqnum) extract(rcvpkt,data) deliver_data(data) sndpkt = make_pkt(expectedseqnum,ACK,chksum) udt_send(sndpkt) expectedseqnum++ expectedseqnum=1 sndpkt = make_pkt(0,ACK,chksum)  GBN: receiver extended FSM initialize: counter for expected sequence number make initial packet ready for sending but use seqnum 0 in case that 1 st packet is corrupt or has wrong number

49 Transport Layer 3-49 Wait udt_send(sndpkt) default rdt_rcv(rcvpkt) && notcorrupt(rcvpkt) && hasseqnum(rcvpkt,expectedseqnum) extract(rcvpkt,data) deliver_data(data) sndpkt = make_pkt(expectedseqnum,ACK,chksum) udt_send(sndpkt) expectedseqnum++ expectedseqnum=1 sndpkt = make_pkt(0,ACK,chksum)  GBN: receiver extended FSM if packet arrives that is not corrupt and has the expected seq number: - extract and deliver data to layer above -send ack -increase seq number

50 Transport Layer 3-50 Wait udt_send(sndpkt) default rdt_rcv(rcvpkt) && notcurrupt(rcvpkt) && hasseqnum(rcvpkt,expectedseqnum) extract(rcvpkt,data) deliver_data(data) sndpkt = make_pkt(expectedseqnum,ACK,chksum) udt_send(sndpkt) expectedseqnum++ expectedseqnum=1 sndpkt = make_pkt(0,ACK,chksum)  GBN: receiver extended FSM if packet arrives that is corrupt or has the wrong seq number: resend ack for last correctly received packet

51 Transport Layer 3-51 ACK-only: always send ACK for correctly-received pkt with highest in-order seq #  may generate duplicate ACKs (e.g. if there is a gap)  need only remember expectedseqnum  out-of-order pkt:  discard (don’t buffer): no receiver buffering!  re-ACK packet with highest in-order seq # Wait udt_send(sndpkt) default rdt_rcv(rcvpkt) && notcurrupt(rcvpkt) && hasseqnum(rcvpkt,expectedseqnum) extract(rcvpkt,data) deliver_data(data) sndpkt = make_pkt(expectedseqnum,ACK,chksum) udt_send(sndpkt) expectedseqnum++ expectedseqnum=1 sndpkt = make_pkt(0,ACK,chksum)  GBN: receiver extended FSM

52 Transport Layer 3-52 GBN in action send pkt1 send pkt2 send pkt3 send pkt4 (wait) sender receiver receive pkt1, send ack1 receive pkt2, send ack2 receive pkt4, discard, (re)send ack2 rcv ack1, send pkt5 rcv ack2, send pkt6 pkt 3 timeout send pkt3 send pkt4 send pkt5 send pkt6 X loss receive pkt5, discard, (re)send ack2 receive pkt6, discard, (re)send ack2 rcv pkt3, deliver, send ack3 rcv pkt4, deliver, send ack4 rcv pkt5, deliver, send ack5 rcv pkt6, deliver, send ack6 ignore duplicate ACK sender window (N=4) remark: packet numbers start at 1 (different to applet)

53 Transport Layer 3-53 GBN in action (variant 1) send pkt1 send pkt2 send pkt3 send pkt4 (wait) sender receiver receive pkt1, send ack1 receive pkt2, send ack2 receive pkt4, discard, (re)send ack2 rcv ack1, send pkt5 rcv ack2, send pkt6 pkt3 timeout send pkt3 send pkt4 send pkt5 send pkt6 X loss receive pkt5, discard, (re)send ack2 receive pkt6, discard, (re)send ack2 rcv pkt3, deliver, send ack3 rcv pkt4, deliver, send ack4 rcv pkt5, deliver, send ack5 rcv pkt6, deliver, send ack6 ignore duplicate ACK sender window (N=4) remark: packet numbers start at 1 (different to applet)

54 Transport Layer 3-54 GBN in action (variant 2) send pkt1 send pkt2 send pkt3 send pkt4 (wait) sender receiver receive pkt1, send ack1 receive pkt2, send ack2 receive pkt4, discard, (re)send ack2 rcv ack1, send pkt5 rcv ack2, send pkt6 timeout send pkt3 send pkt4 send pkt5 send pkt6 X loss receive pkt5, discard, (re)send ack2 receive pkt6, discard, (re)send ack2 rcv pkt3, deliver, send ack3 rcv pkt4, deliver, send ack4 rcv pkt5, deliver, send ack5 rcv pkt6, deliver, send ack6 ignore duplicate ACK sender window (N=4) remark: packet numbers start at 1 (different to applet) increase base and restart timer

55 Transport Layer 3-55 Selective repeat  improves GBN by not retransmitting a large number of packets  receiver individually acknowledges all correctly received packets  receiver buffers packets, as needed, for eventual in- order delivery to upper layer  sender only resends packets for which ACK not received  sender has timer for each unACKed packet (since only a single packet will be retransmitted on timeout)  sender window  N consecutive seq #’s  limits seq #s of sent, unACKed packets

56 Transport Layer 3-56 Selective repeat: sender, receiver windows windows are not synchronized!!

57 Transport Layer 3-57 Selective repeat data from above:  if next available seq # in window, then send packet timeout(n):  resend packet n, restart timer of packet n ACK(n) in [sendbase,sendbase+N-1]:  mark packet n as received  if n smallest unACKed packet, then advance window base to next unACKed seq # sender packet 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 packet n in [rcvbase-N,rcvbase-1]  ACK(n) since previous ack got lost receiver

58 Transport Layer 3-58 Selective repeat in action send pkt0 send pkt1 send pkt2 send pkt3 (wait) sender receiver receive pkt0, send ack0 receive pkt1, send ack1 receive pkt3, buffer, send ack3 rcv ack0, send pkt4 rcv ack1, send pkt5 pkt 2 timeout resend pkt2 X loss receive pkt4, buffer, send ack4 receive pkt5, buffer, send ack5 rcv pkt2; deliver pkt2, pkt3, pkt4, pkt5; send ack2 record ack3 arrived sender window (N=4) record ack4 arrived record ack5 arrived Q: what happens when ack2 arrives?

59 Transport Layer 3-59 Selective repeat: short range of seqnum example:  seq #’s: 0, 1, 2, 3  window size=3 receiver window (after receipt) sender window (after receipt) pkt0 pkt1 pkt pkt0 timeout retransmit pkt X X X will accept packet with seq number 0 (b) oops! pkt0 pkt1 pkt pkt X will accept packet with seq number pkt3 (a) no problem receiver can’t see sender side. receiver behavior identical in both cases! something’s (very) wrong!  receiver does not know whether pkt0 is retransmitted or not  duplicate data accepted as new in (b)? Q: what relationship between seq # range and window size to avoid problem in (b)?

60 Transport Layer 3-60 Packet Reordering  assume now that packets my be reordered during transmission (since we do not have a single channel but a network of links)  receiver may get packets with seqnum n where n is neither in sender’s nor in receiver’s window (e.g. delayed ACK => retransmission => windows move further)  duplicate or new data?????  sender must be ‘sure’ that when reusing seqnum n, a packet with this number is no longer on the way to receiver Solution: assume that packet cannot ‘live’ in network longer than maximum amount of time (e.g. 3 min for TCP)

61 Transport Layer 3-61 Summary of Reliable Data Transfer MechanismUse, Comments ChecksumUsed to detect bit errors in transmitted packet. TimerUsed to retransmit if packet or ACK was lost (or delayed). Sequence number Sequential numbering of packet flow. Use gaps to detect lost packets, use duplicate numbers to detect duplicate packets. Acknow- ledgment Receiver confirms correctly received packets; contains sequence number; may be cumulative or individual. (Negative ack- nowledgment) Receiver tells sender that packet has not been received. Window, pipelining Sending of packets only in restricted range => increases utilization of sender compared to stop-and-wait.

62 Transport Layer 3-62 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

63 Transport Layer 3-63 TCP: Overview RFCs: 793,1122,1323, 2018, 2581  full duplex data:  bi-directional data flow in same connection  MSS: maximum segment size (max amount of data placed in segment); usually about 1460 B since Ethernet frames typically have 1500 Bytes (TCP/IP header: 40 B)  connection-oriented:  handshaking (exchange of control msgs) inits sender, receiver state before data exchange  point-to-point:  one sender, one receiver  reliable  mechanisms for message loss/ corrupted msgs  pipelined:  TCP congestion and flow control set window size  flow controlled:  sender will not overwhelm receiver

64 Transport Layer 3-64 TCP segment structure source port # dest port # 32 bits application data (variable length) sequence number acknowledgement number receive window Urg data pointer checksum F SR PAU head len not used options (variable length) used for (de- )multiplexing data to/from upper-layer applications

65 Transport Layer 3-65 TCP segment structure source port # dest port # 32 bits application data (variable length) sequence number acknowledgement number receive window Urg data pointer checksum F SR PAU head len not used options (variable length) error detection

66 Transport Layer 3-66 TCP segment structure source port # dest port # 32 bits application data (variable length) sequence number acknowledgement number receive window Urg data pointer checksum F SR PAU head len not used options (variable length) counting by bytes of data (not segments!); used for implementing a reliable data transfer service; more details later

67 Transport Layer 3-67 TCP segment structure source port # dest port # 32 bits application data (variable length) sequence number acknowledgement number receive window Urg data pointer checksum F SR PAU head len not used options (variable length) used for flow control; indicates number of bytes that receiver is willing to accept (more details later)

68 Transport Layer 3-68 TCP segment structure source port # dest port # 32 bits = 4 bytes application data (variable length) sequence number acknowledgement number receive window Urg data pointer checksum F SR PAU head len not used options (variable length) TCP header length (4 bits); because of options field length can be greater than 20 bytes 5 rows

69 Transport Layer 3-69 TCP segment structure source port # dest port # 32 bits application data (variable length) sequence number acknowledgement number receive window Urg data pointer checksum F SR PAU head len not used options (variable length) flag fields (6 bits): A = ACK field acknowledgement number is valid (this msg is an ACK; used for reliable data transfer!)

70 Transport Layer 3-70 TCP segment structure source port # dest port # 32 bits application data (variable length) sequence number acknowledgement number receive window Urg data pointer checksum F SR PAU head len not used options (variable length) flag fields (6 bits): R = RST S = SYN F = FIN used for connection setup and teardown (more details later)

71 Transport Layer 3-71 TCP segment structure source port # dest port # 32 bits application data (variable length) sequence number acknowledgement number receive window Urg data pointer checksum F SR PAU head len not used options (variable length) flag fields (6 bits): P = PSH field TCP push function for cases where data needs to be sent immediately (no buffering! sender TCP does not wait for more data) Examples: - Telnet: want each keystroke to be sent immed. -HTTP GET request: client has no further data to add and request should be sent to web daemon immediately (do not wait until segment “filled”) -packet containing last bytes of requested file (for now sender has no further data to transmit)

72 Transport Layer 3-72 TCP segment structure source port # dest port # 32 bits application data (variable length) sequence number acknowledgement number receive window Urg data pointer checksum F SR PAU head len not used options (variable length) flag fields (6 bits): U = URG field segment contains data that sending site marked as urgent => urgent data pointer (16 bit) points to last byte of urgent part in segment (isn’t employed much; usually in comb. with PSH) receiver TCP forwards the urgent data to the process with an indication that the data is marked as urgent by the sender

73 Transport Layer 3-73 TCP segment structure source port # dest port # 32 bits application data (variable length) sequence number acknowledgement number receive window Urg data pointer checksum F SR PAU head len not used 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 willing to accept counting by bytes of data (not segments!) Internet checksum (as in UDP)

74 Transport Layer 3-74 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 (=> acks bytes up to first missing byte in stream)  TCP spec doesn’t say how to handle out-of-order segments, - up to implementor (in practise: receiver buffers and waits for missing bytes to fill gaps) source port # dest port # sequence number acknowledgement number checksum rwnd urg pointer incoming segment to sender A sent ACKed sent, not- yet ACKed (“in-flight”) usable but not yet sent not usable window size N sender sequence number space source port # dest port # sequence number acknowledgement number checksum rwnd urg pointer outgoing segment from sender

75 Transport Layer 3-75 TCP seq. numbers, ACKs Example 1: Host A has received bytes numbered from Host B  Host A waits for  puts 536 in ACK number field when it sends next segment to B Example 2: Host A has received bytes numbered from Host B AND bytes (has not yet received 536 – 899)  Host A waits for  puts 536 in ACK number field when it sends next segment to B

76 Transport Layer 3-76 Telnet example User types ‘C’ host ACKs receipt of echoed ‘C’ host ACKs receipt of ‘C’, echoes back ‘C’ each character typed by user (at client host A) is sent to sever (host B) and back to be displayed at Telnet’s user screen (echo back) => user sees what has already been processed on remote site Host B Host A Seq=42, ACK=79, data = ‘C’ Seq=79, ACK=43, data = ‘C’ Seq=43, ACK=80 seqnum at A starts at 42 (waiting for byte 79) seqnum at B starts at 79 (waiting for byte 42)

77 Transport Layer 3-77 Telnet example User types ‘C’ host ACKs receipt of echoed ‘C’ host ACKs receipt of ‘C’, echoes back ‘C’ each character typed by user (at client host A) is sent to sever (host B) and back to be displayed at Telnet’s user screen (echo back) => user sees what has already been processed on remote site Host B Host A Seq=42, ACK=79, data = ‘C’ Seq=79, ACK=43, data = ‘C’ Seq=43, ACK=80 seqnum at A starts at 42 (waiting for byte 79) seqnum at B starts at 79 (waiting for byte 42) because data is 1 byte (header is not counted)

78 Transport Layer 3-78 Telnet example User types ‘C’ host ACKs receipt of echoed ‘C’ host ACKs receipt of ‘C’, echoes back ‘C’ each character typed by user (at client host A) is sent to sever (host B) and back to be displayed at Telnet’s user screen (echo back) => user sees what has already been processed on remote site Host B Host A Seq=42, ACK=79, data = ‘C’ Seq=79, ACK=43, data = ‘C’ Seq=43, ACK=80 seqnum at A starts at 42 (waiting for byte 79) seqnum at B starts at 79 (waiting for byte 42) send data and ack together (ack is piggybacked)

79 Transport Layer 3-79 Telnet example User types ‘C’ host ACKs receipt of echoed ‘C’ host ACKs receipt of ‘C’, echoes back ‘C’ each character typed by user (at client host A) is sent to sever (host B) and back to be displayed at Telnet’s user screen (echo back) => user sees what has already been processed on remote site Host B Host A Seq=42, ACK=79, data = ‘C’ Seq=79, ACK=43, data = ‘C’ Seq=43, ACK=80 seqnum at A starts at 42 (waiting for byte 79) seqnum at B starts at 79 (waiting for byte 42)

80 Transport Layer 3-80 HTTP example SYN bit = 1 to establish connection (more details later) => increase sequence number by one even though no bytes are sent. Note: we use relative seq num here. When connection is established a random initial number is chosen.

81 Transport Layer 3-81 TCP round trip time, timeout Q: how to set TCP timeout value?  longer than connection’s round- trip time (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 (it is ambiguous whether the reply was for the first instance of the packet or a later instance)  SampleRTT will vary => use average  average several recent measurements, not just current SampleRTT TCP uses a timeout/retransmission mechanism (as rdt protocol considered earlier)

82 Transport Layer 3-82 EstimatedRTT = (1-  )*EstimatedRTT +  *SampleRTT  exponential weighted moving average  influence of past sample decreases exponentially fast (why?)  typical value:  = TCP round trip time, timeout RTT (milliseconds) RTT: gaia.cs.umass.edu to fantasia.eurecom.fr sampleRTT EstimatedRTT time (seconds)

83 Transport Layer 3-83 EstimatedRTT = (1-  )*EstimatedRTT +  *SampleRTT  exponential weighted moving average  influence of past sample decreases exponentially fast (why?)  typical value:  = TCP round trip time, timeout RTT (milliseconds) RTT: gaia.cs.umass.edu to fantasia.eurecom.fr sampleRTT EstimatedRTT time (seconds) α close to 1 => weighted average immune to changes that last a short time (e.g., a single segment that encounters long delay) α close to 0 => weighted average respond to changes in delay very quickly

84 Transport Layer 3-84  timeout interval: EstimatedRTT plus “safety margin”  large variation in EstimatedRTT -> larger safety margin  estimate SampleRTT deviation from EstimatedRTT: DevRTT = (1-  )*DevRTT +  *|SampleRTT-EstimatedRTT| TCP round trip time, timeout (typically,  = 0.25) TimeoutInterval = EstimatedRTT + 4*DevRTT estimated RTT “safety margin” RFC 2988

85 Transport Layer 3-85 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

86 Transport Layer 3-86 TCP reliable data transfer  TCP creates rdt service on top of IP’s unreliable service  pipelined segments  cumulative acks  single retransmission timer (not one for each segm)  retransmissions triggered by:  timeout events  duplicate acks let’s initially consider simplified TCP sender:  data transfer in one direction only  ignore duplicate acks  ignore flow control, congestion control  assume that data from above is less than MSS

87 Transport Layer 3-87 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 (slide window to the right)  (re)start timer if there are still unacked segments

88 Transport Layer 3-88 TCP sender (simplified) wait for event NextSeqNum = InitialSeqNum SendBase = InitialSeqNum  create segment, seq. #: NextSeqNum pass 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 } ACK received, with ACK field value y RFC 2988, page 3 hope that gap is just a single segment

89 Transport Layer 3-89 TCP sender (simplified) wait for event NextSeqNum = InitialSeqNum SendBase = InitialSeqNum  create segment, seq. #: NextSeqNum pass 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 } ACK received, with ACK field value y RFC 2988, page 4

90 Transport Layer 3-90 TCP: retransmission scenarios lost ACK scenario Host B Host A Seq=92, 8 bytes of data ACK=100 Seq=92, 8 bytes of data X timeout ACK=100 premature timeout Host B Host A Seq=92, 8 bytes of data ACK=100 Seq=92, 8 bytes of data timeout ACK=120 Seq=100, 20 bytes of data ACK=120 SendBase=100 SendBase=120 SendBase=92 since nextseqnum at A is 100

91 Transport Layer 3-91 TCP: retransmission scenarios X cumulative ACK Host B Host A Seq=92, 8 bytes of data ACK=100 Seq=120, 15 bytes of data timeout Seq=100, 20 bytes of data ACK=120

92 Transport Layer 3-92 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 receiver is allowed to delay 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 duplicate ACK: an ACK that has been sent before maybe cumulative ACK possible => increases performance

93 Transport Layer 3-93 TCP fast retransmit  time-out period often relatively long:  long delay before resending lost packet  if segment is lost, there will likely be many duplicate ACKs. if sender receives 3 ACKs for same data (“triple duplicate ACKs”), immediately resend unacked segment with smallest seq #  likely that unacked segment lost, so don’t wait for timer to expire TCP fast retransmit

94 Transport Layer 3-94 X fast retransmit after sender receipt of triple duplicate ACK Host B Host A Seq=92, 8 bytes of data ACK=100 timeout ACK=100 TCP fast retransmit Seq=100, 20 bytes of data

95 Transport Layer 3-95 TCP sender (simplified) wait for event NextSeqNum = InitialSeqNum SendBase = InitialSeqNum  create segment, seq. #: NextSeqNum pass 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 if (there are currently not-yet-acked segments) start timer } else { /* duplicate ACK received */ increment number of dupl. ACKs received for y if (number of dupl. ACKs received for y==3) resend segment with seq. number y } ACK received, with ACK field value y

96 Transport Layer 3-96 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

97 Transport Layer 3-97 TCP flow control receiver controls sender such that sender won’t overflow receiver’s buffer by transmitting too much, too fast  sender maintains variable called receive window (rwnd)  gives sender an idea of how much buffer space is available at receiver  since TCP is full-duplex, both sides have receive window variable LastByteRead: number of last byte read by app (on rcv site) LastByteRcvd: number of last byte in received data stream rwnd = RcvBuffer – [LastByteRcvd - LastByteRead] currently being buffered on receiver site

98 Transport Layer 3-98 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)  sender limits amount of unacked (“in-flight”) data to receiver’s rwnd value  sender guarantees that receive buffer will not overflow: receiver-side buffering LastByteSent – LastByteAcked ≤ rwnd applet:

99 Transport Layer 3-99 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

100 Transport Layer Connection Management before exchanging data, sender/receiver do “three-way handshake”: => agree to establish connection (each knowing the other willing to establish connection) and synchronize seq. numbers Step 1: client send special TCP segment with SYN flag =1 and initial seq. num. (client_isn) in seq. num. field Step 2: SYN segm. arrives at server => sever allocates TCP buffers and variables and sends SYNACK segment with (a)SYN and ACK flag =1 (b)ack number field is equal to client_isn+1 (c)puts own initial seq. num. (server_isn) in seq. num. field Step 3: Client receives SYNACK segm., allocates buffers & vars, and sends ACK with ack num (server_isn+1) (which may already contain app data)

101 Transport Layer TCP 3-way handshake SYNbit=1, Seq=x choose init seq num, x send TCP SYN msg ESTAB SYNbit=1, Seq=y ACKbit=1; ACKnum=x+1 choose init seq num, y send TCP SYNACK msg, acking SYN ACKbit=1, ACKnum=y+1 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 SYNSENT ESTAB SYN RCVD client state LISTEN server state LISTEN

102 Transport Layer TCP: closing a connection  client or server can start closing of connection  host A sends TCP segment with FIN bit = 1  host B responds to received FIN with ACK  host B also sends FIN  (on receiving FIN, ACK can be combined with own FIN)  host A receives FIN and sends final ACK  host A waits certain time in case that last ACK got lost and FIN is resent (=> A will ack again)

103 Transport Layer FIN_WAIT_2 CLOSE_WAIT FINbit=1, seq=y ACKbit=1; ACKnum=y+1 ACKbit=1; ACKnum=x+1 wait for server close can still send data can no longer send data LAST_ACK CLOSED TIMED_WAIT timed wait (usually 30 sec, 1 min or 2 min) CLOSED TCP: closing a connection FIN_WAIT_1 FINbit=1, seq=x can no longer send but can receive data clientSocket.close() client state server state ESTAB

104 Transport Layer Possible TCP states of client CLOSED SYN_SENT ESTAB FIN_WAIT_1 FIN_WAIT_2 TIME _WAIT send SYN receive SYNACK send ACK send FIN receive ACK (send nothing) receive FIN send ACK wait X seconds

105 Transport Layer Possible TCP states of server CLOSED LISTEN SYN_RCVD ESTAB CLOSE_WAIT LAST_ACK create listen socket receive SYN send SYNACK receive ACK (send nothing) receive FIN send ACK send FIN receive ACK (send nothing) serverPort = serverSocket = socket(AF_INET,SOCK_STREAM) serverSocket.bind((‘’,serverPort)) serverSocket.listen(1)

106 Transport Layer 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

107 Transport Layer congestion:  informally: “too many sources sending too much data too fast for network to handle”  different from flow control!  problems related to congestion:  lost packets (buffer overflow at routers)  long delays (queueing in router buffers)  important problem in data networks! Principles of congestion control

108 Transport Layer 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 buffer Host A original data: in Host B throughput:  out R/2 out in R/2 packet delay in  large delays as arrival rate, in, approaches capacity

109 Transport Layer  one router, finite buffer  sender retransmission of timed-out packet  application-layer sends at rate in  transport-layer includes retransmissions :  in ’ ≥ in finite shared output link buffers Host A in : original data Host B out ' in : original data, plus retransmitted data Causes/costs of congestion: scenario 2

110 Transport Layer idealization: perfect knowledge  sender sends only when router buffers available (=> no retransmissions) finite shared output link buffers in : original data out ' in : original data, plus retransmitted data copy free buffer space! R/2 out in  ' in Causes/costs of congestion: scenario 2 Host B A

111 Transport Layer in : original data out ' in : original data, plus retransmitted data copy no buffer space! Idealization: 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 A Host B

112 Transport Layer in : original data out ' in : original data, plus retransmitted data free buffer space! 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 in out when sending at R/2, some packets are retransmissions but asymptotic goodput is still R/2 (only original data leaves router) A Host B

113 Transport Layer A in out ' in copy free buffer space! timeout R/2 in out when sending at R/2, some packets are retransmissions including duplicates that are delivered! 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

114 Transport Layer R/2 out when sending at R/2, some packets are retransmissions including duplicates that are delivered! “costs” of congestion:  more work (retrans) for given “goodput”  unneeded retransmissions: link carries multiple copies of pkt  decreasing goodput R/2 in 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

115 Transport Layer  four senders  multihop paths  timeout/retransmit  R: capa. of router links Q: what happens as in and in ’ increase in A-C connection ? finite shared output link buffers Host A out Causes/costs of congestion: scenario 3 Host B Host C Host D in : original data ' in : original data, plus retransmitted data A: as red in ’ increases, all arriving blue pkts at upper queue are dropped, blue throughput  0 blue in-rate at upper router is at most R<< in ‘ ; router is ‘flooded’ with red packets => blue packets get lost

116 Transport Layer Causes/costs of congestion: scenario 3 R/2 out in ’ buffer overflows rare for small in ‘ throughput increases as long as in ‘ is not too large (buffer overflows still rare, more original data arrives)

117 Transport Layer another “cost” of congestion:  when packets are dropped, any upstream transmission capacity (left router!) used for that packet was wasted! Causes/costs of congestion: scenario 3 R/2 out in ’ in large for all connections: consider D-B conn. at upper router => most blue packets lost

118 Transport Layer Approaches towards congestion control two broad 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  explicit rate for sender to send at

119 Transport Layer Case study: ATM ABR congestion control Some remarks about ATM networks:  Asynchronous Transfer Mode (ATM) (also called cell relay)  originally designed to carry both voice and data traffic over WANs  only used in some backbone networks of providers (mostly because of its high costs compared to e.g. Ethernet)  Price of 100Mbps Enet NIC: < $10  Price of 155Mbps ATM NIC: > $500  fixed-sized packets called cells (Internet has variable sized packets)  In order to interconnect with the TCP/IP world, an ATM gateway is used that converts TCP/IP and Ethernet frames into ATM cells and then converts them back once they have reached their destination network.

120 Transport Layer Case study: ATM ABR congestion control Some remarks about ATM networks:  establish virtual-circuit (VC) between two endpoints before data exchange => all cells (of a fixed connection) take same path via certain routers (here called switches)  data is delivered in correct order  switches can track state of VC => know average transmission rate of sender  switches can signal sender to reduce rate in case of congestion => network-assisted congestion control

121 Transport Layer Case study: ATM ABR congestion control RM (resource management) cells:  sent by sender, interspersed with data cells (default: every 32 data cells)  travel along the data path to the destination and sent back  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 (probably modified by receiver and switches) ABR: available bit rate:  transmisson rate is adjusted by sender based on returned RM cells  if sender’s path “underloaded”:  sender should use available bandwidth  if sender’s path congested:  sender throttled to minimum guaranteed rate

122 Transport Layer Case study: ATM ABR congestion control  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  Explicit Forward Congestion Indication (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 RM celldata cell

123 Transport Layer Case study: ATM ABR congestion control source of illustration: older switch -> does not support RM cells

124 Transport Layer Case study: ATM ABR congestion control  detailed ABR flow control mechanism is complex and ATM switches allow for many different configurations  => ABR is not very prevalent today  nice overview article: “ABR Service on ATM Networks: What is ?“ by R. Jain (1995)

125 Transport Layer 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

126 Transport Layer TCP congestion control algorithm:  approach: use congestion window variable cwnd  choose sending rate accordingly  goal: LastByteSent – LastByteAcked ≤ min{cwnd, rwnd} what sender knows receive window (free buffer space at receiver) let’s assume for simplicity that rwnd is large and sender has always data to send => last byte ACKed sent, not- yet ACKed (“in-flight”) last byte sent cwnd sender sequence number space

127 Transport Layer TCP Congestion Control: details  cwnd is dynamic, function of perceived network congestion  cwnd indirectly limits sending rate 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 sender sequence number space rate ~ ~ cwnd RTT bytes/sec  idea: sender tries to find maximal rate at which no losses occur = bandwidth probing

128 Transport Layer TCP congestion control:  approach: sender increases window size (=> transmission rate) probing for usable bandwidth, until loss occurs  additive increase: increase cwnd by 1 MSS for every received ACK until loss detected  corresponds to doubling of cwnd every RTT since each segment is of size 1 MSS (or less) and # of received ACKs is old-cwnd-value  loss detection: timeout event or three duplicate ACKS

129 Transport Layer TCP Slow Start (not really slow)  when connection begins, increase rate 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  slow start ends when congestion avoidance mode starts (later) of when loss occurs Host A one segment RTT Host B time two segments four segments

130 Transport Layer TCP: detecting, reacting to loss  loss indicated by timeout:  before resetting cwnd store slow start threshold: ssthresh:= cwnd/2  cwnd set to 1 MSS;  window then grows exponentially (as in slow start) until threshold ssthres, then enter congestion avoidance mode, where cwnd increased by 1 MSS every RRT (linear growth)  loss indicated by 3 duplicate ACKs:  dup ACKs indicate network capable of delivering some segments  cwnd is cut in half window then grows linearly  Earlier TCP versions: always set cwnd to 1 (timeout or 3 duplicate acks)

131 TCP: switching from slow start to congestion avoidance Transport Layer Q: when should the exponential increase (slow start) 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

132 TCP: additive increase, multiplicative decrease (AIMD) algorithm Transport Layer (Ignoring slow start and timeouts.) - we have additive increase in congestion avoidance mode: - we have multiplicative decrease in case of 3 duplicate ACKs (after RTT time): after RTT time: cwnd = cwnd + 1 MSS (linear growth) cwnd = 0.5 * cwnd

133 Transport Layer  TCP Tahoe: early version of TCP; cut cong. window to 1 MSS in both cases (timeout AND triple dupl. ACK)  TCP Reno: newer version; uses ‘fast recovery’ (increase cwnd by 1 MSS for every duplicate ACK) Congestion Window: Reno vs Tahoe Tahoe and Reno curve is identical actually, it should start at 6+3=9!

134 Transport Layer Summary: TCP Congestion Control timeout ssthresh = cwnd/2 cwnd = 1 MSS dupACKcount = 0 retransmit missing segment  cwnd > 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 retransmit missing segment dupACKcount == 3 timeout ssthresh = cwnd/2 cwnd = 1 dupACKcount = 0 retransmit missing segment ssthresh= cwnd/2 cwnd = ssthresh + 3 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  cwnd = 1 MSS ssthresh = 64 KiB dupACKcount = 0 New ACK! New ACK! New ACK!

135 Transport Layer Summary: TCP Congestion Control 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  cwnd = 1 MSS ssthresh = 64 KB dupACKcount = 0 increase cwnd by 1 MSS for every ACK  if cwnd/MSS segements are sent in time interval [0,RTT] then cwnd is twice as large after receiving cwnd/MSS ACKs Example: 1 MSS = 1460 Bytes cwnd = 14,600 Bytes, then 10 segments are being sent within at RTT

136 Transport Layer Summary: TCP Congestion Control  cwnd > ssthresh 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  cwnd = 1 MSS ssthresh = 64 KB dupACKcount = 0 congestion avoidance cwnd = cwnd + MSS (MSS/cwnd) dupACKcount = 0 transmit new segment(s), as allowed new ACK. dupACKcount++ duplicate ACK timeout ssthresh = cwnd/2 cwnd = 1 MSS dupACKcount = 0 retransmit missing segment increase cwnd by 1 MSS for every RTT  receiving cwnd/MSS ACKs per RTT means we add cwnd/MSS times the factor MSS*(MSS/cwnd)  add 1 MSS per RTT  example: cwnd/MSS = 10 segments

137 Transport Layer Summary: TCP Congestion Control timeout ssthresh = cwnd/2 cwnd = 1 MSS dupACKcount = 0 retransmit missing segment  cwnd > ssthresh congestion avoidance cwnd = cwnd + MSS (MSS/cwnd) dupACKcount = 0 transmit new segment(s), as allowed new ACK. dupACKcount++ duplicate ACK ssthresh= cwnd/2 cwnd = ssthresh + 3 retransmit missing segment dupACKcount == 3 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  cwnd = 1 MSS ssthresh = 64 KB dupACKcount = 0 set cwnd to cwnd/2 since it was too high  try to find ‘optimal’ value for cwnd

138 Transport Layer Summary: TCP Congestion Control timeout ssthresh = cwnd/2 cwnd = 1 MSS dupACKcount = 0 retransmit missing segment  cwnd > 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 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  cwnd = 1 MSS ssthresh = 64 KB dupACKcount = 0 - increase cwnd on every dupl. ACK (note that dupl. ACK means that certain packets get through) - won’t have many dupl. ACK since we retransmit missing packet - additionally we can transmit further segments (cwnd increases quickly) - see RFC 2581

139 Transport Layer Summary: TCP Congestion Control timeout ssthresh = cwnd/2 cwnd = 1 MSS dupACKcount = 0 retransmit missing segment  cwnd > 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 retransmit missing segment dupACKcount == 3 timeout ssthresh = cwnd/2 cwnd = 1 dupACKcount = 0 retransmit missing segment ssthresh= cwnd/2 cwnd = ssthresh + 3 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  cwnd = 1 MSS ssthresh = 64 KB dupACKcount = 0 New ACK! New ACK! New ACK! applet

140 Transport Layer TCP throughput  ‘saw tooth’ behavior (if slow start is ignored)  compute avg. TCP throughput as function of window size and RTT using some simplifying assumptions  ignore slow start, assume always data to send  fixed window size W (measured in bytes) when loss occurs  transmission rate ranges from 0.5W/RTT to W/RTT  avg. window size (# in-flight bytes) is ¾ W W W/2 avg TCP throughput = 3 4 W RTT Bytes/sec

141 Transport Layer High-Speed TCP Connections  example: 1500 Byte segments, 100ms RTT, want 10 Gbps throughput (≈ / (1500*8) = 1/12 * 10 7 segments/sec)  => W = thr * RTT * 4/3 = 1/12 * 10 7 * 1/10 * 4/3 = 1/9 * 10 6 ≈ 10 5 in-flight segments (!!!!) This is a lot!!! avg TCP throughput = 3 4 W RTT Bytes/sec

142 Transport Layer High-Speed TCP Connections  What fraction of packets could be lost so that we still have a throughput of 10Gbps?  throughput in terms of segment loss probability, L : ➜ to achieve 10 Gbps throughput, need a loss rate of L ≈ 2 · – very small loss rate (one loss every 5 billion segm)  new versions of TCP for high-speed (RFC 3649) TCP throughput = MSS RTT L homework Remark: one cannot simply increase the MSS since this will lead to a lot of IP fragmentation (=> less efficient and in case of fragment loss whole segment is lost) default is 536 B => plus 40 B header gives MTU for IP networks local networks use larger MSS (e.g Bytes for Ethernet) modern LANs can use Jumbo frames (MTU up to 9000 B)

143 Transport Layer fairness goal: if K TCP sessions share same bottleneck link of bandwidth R, each should have average rate of R/K TCP connection 1 bottleneck router capacity R TCP Fairness TCP connection 2

144 Transport Layer Why TCP is fair (Intuition) two competing sessions (same MSS, RTT, no other traffic):  below full utilization, increase cwnd by 1 MSS per RTT in congestion avoidance mode (ignore slow start phase)  both detect losses during increase => halve their windows R R equal bandwidth share Connection 1 throughput Connection 2 throughput congestion avoidance: additive increase loss: decrease window by factor of 2 congestion avoidance: additive increase loss: decrease window by factor of 2 full bandwidth utilization => realized bandwidth converges to equal bandwidth share!

145 Transport Layer 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  => crowd out TCP traffic  need for congestion control for UDP Fairness, parallel TCP connections  application can open multiple parallel connections between two hosts => larger fraction of bandwidth at congested link (e.g. multiple tabs in browser)  unfair!  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 11* R/20

146 Transport Layer Chapter 3: summary  principles behind transport layer services:  multiplexing, demultiplexing  reliable data transfer  flow control  congestion control  their instantiation, implementation in the Internet  UDP  TCP Remark: TCP is actually more complex than what we have described (variety of patches, fixes improvements) next:  leaving the network “edge” (application, transport layers)  into the network “core”


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