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Transport Layer 3-1 08-Transport Layer: TCP. Transport Layer 3-2 TCP: Overview RFCs: 793, 1122, 1323, 2018, 2581 r full duplex data: m bi-directional.

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Presentation on theme: "Transport Layer 3-1 08-Transport Layer: TCP. Transport Layer 3-2 TCP: Overview RFCs: 793, 1122, 1323, 2018, 2581 r full duplex data: m bi-directional."— Presentation transcript:

1 Transport Layer 3-1 08-Transport Layer: TCP

2 Transport Layer 3-2 TCP: Overview RFCs: 793, 1122, 1323, 2018, 2581 r full duplex data: m bi-directional data flow in same connection r connection-oriented: m handshaking (exchange of control msgs) r flow control: m sender will not overwhelm receiver r congestion control r point-to-point: m one sender, one receiver r reliable, in-order byte stream: m no “message boundaries” r pipelined: m TCP congestion and flow control set window size r send & receive buffers

3 Transport Layer 3-3 TCP segment structure source port # dest port # 32 bits application data (variable length) sequence number acknowledgement number Receive window Urg data pnter checksum F SR PAU head len not used Options (variable length) URG: urgent data (generally not used) ACK: ACK # valid PSH: push data up 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)

4 Transport Layer 3-4 Data = 500,000 bytes MSS = 1000 bytes

5 TCP seq. #’s and ACKs Transport Layer 3-5 Seq. #’s: m byte stream “number” of first byte in segment’s data ACKs: m seq # of next byte expected from other side m cumulative ACK(as in GBN) m Buffers at receiver can hold out-of-order bytes, hence when ACK for “n” is not received, retransmit only “n” and not n+1 and all others.(as in SR) m Piggybacked ACKS

6 Transport Layer 3-6 TCP Round Trip Time and Timeout Q: how to set TCP timeout value? r longer than RTT m but RTT varies r too short: premature timeout m unnecessary retransmissions r too long: slow reaction to segment loss Q: how to estimate RTT?  SampleRTT : measured time from segment transmission until ACK receipt m ignore retransmissions  SampleRTT will vary, want estimated RTT “smoother”  average several recent measurements, not just current SampleRTT

7 Transport Layer 3-7 TCP Round Trip Time and Timeout EstimatedRTT = (1-  )*EstimatedRTT +  *SampleRTT r Exponential weighted moving average r influence of past sample decreases exponentially fast  typical value:  = 0.125

8 Transport Layer 3-8 Example RTT estimation:

9 Transport Layer 3-9 TCP Round Trip Time and Timeout Setting the timeout  EstimatedRTT plus “safety margin”  large variation in EstimatedRTT -> larger safety margin 1. Estimate how much SampleRTT deviates from EstimatedRTT: TimeoutInterval = EstimatedRTT + 4*DevRTT DevRTT = (1-  )*DevRTT +  *|SampleRTT-EstimatedRTT| (typically,  = 0.25) 2. Then set timeout interval:

10 Transport Layer 3-10 TCP reliable data transfer r TCP creates rdt service on top of IP’s unreliable service r Pipelined segments r Cumulative acks r TCP uses single retransmission timer r Retransmissions are triggered by: m timeout events m duplicate acks r Initially consider simplified TCP sender: m ignore duplicate acks m ignore flow control, congestion control

11 Transport Layer 3-11 TCP sender events: data rcvd from app: r Create segment with seq # r start timer if not already running (think of timer as for oldest unacked segment)  expiration interval: TimeOutInterval (as calculated earlier) timeout: r retransmit segment that caused timeout r restart timer Ack rcvd: r If acknowledges previously unacked segments m update what is known to be acked For example, if ACK=150, then assume that segments with sequence #s 0, 50 and100 have been received at the receiver. m restart timer if there are outstanding segments

12 Transport Layer 3-12 TCP: retransmission scenarios Host A Seq=100, 20 bytes data ACK=100 time premature timeout Host B Seq=92, 8 bytes data ACK=120 Seq=92, 8 bytes data Seq=92 timeout ACK=120 Host A Seq=92, 8 bytes data ACK=100 loss timeout lost ACK scenario Host B X Seq=92, 8 bytes data ACK=100 time Seq=92 timeout SendBase = 100 SendBase = 120 SendBase = 120 Sendbase = 100

13 Transport Layer 3-13 TCP retransmission scenarios (more) Host A Seq=92, 8 bytes data ACK=100 loss timeout Cumulative ACK scenario Host B X Seq=100, 20 bytes data ACK=120 time SendBase = 120

14 Transport Layer 3-14 Doubling the timeout interval r Timeout interval for a segment is doubled every time it has to be re-transmitted. r This is a limited form of congestion control m If the segments are being dropped or are taking longer to get through the network, then TCP eases the congestion in the network, by delaying the retransmission of segments. m Similar to CSMA/CD in Ethernet.

15 Transport Layer 3-15 Fast Retransmit r Time-out period often relatively long: m long delay before resending lost packet r Detect lost segments via duplicate ACKs. m Sender often sends many segments back-to- back m If segment is lost, there will likely be many duplicate ACKs. r If sender receives 3 ACKs for the same data, it assumes that segment after ACKed data was lost: m fast retransmit

16 Transport Layer 3-16 TCP Flow Control r receive side of TCP connection has a receive buffer: r speed-matching service: matching the send rate to the receiving app’s drain rate r app process may be slow at reading from buffer sender won’t overflow receiver’s buffer by transmitting too much, too fast flow control

17 Transport Layer 3-17 TCP Flow control: how it works (Suppose TCP receiver discards out-of-order segments)  spare room in buffer RcvWindow = RcvBuffer-[LastByteRcvd - LastByteRead] e.g. RCVWindow = 1000 – [150 – 50]  Rcvr advertises spare room by including value of RcvWindow in segments  Sender limits unACKed data to RcvWindow m guarantees receive buffer doesn’t overflow m Maintain: LastByteSent-LastByteAcked <= RecvWindow

18 Transport Layer 3-18 TCP Connection Management Recall: TCP sender, receiver establish “connection” before exchanging data segments r initialize TCP variables: m seq. #s Each host randomly picks an initial value for the sequence number. Following this, sequence number is updated based on number of bytes of data is sent.  buffers, flow control info (e.g. RcvWindow ) Three way handshake: Step 1: client host sends TCP SYN segment to server m specifies initial seq # (client_isn) m SYN = 1 m no data Step 2: server host receives SYN, replies with SYNACK segment m server allocates buffers m specifies server initial seq. # (server_seq) m SYN = 1, ACK = client_isn + 1 Step 3: client receives SYNACK, replies with ACK segment, which may contain data. m Client allocates buffers m SYN = 0, ACK = server_isn +1

19 Transport Layer 3-19 RST and FIN r FIN is the bit set when a client is ready to close the connection. r RST is used by the server to indicate that the port and IP address do not match any sockets on the server.

20 Transport Layer 3-20 Principles of Congestion Control Congestion: r informally: “too many sources sending too much data too fast for network to handle” r different from flow control!

21 Transport Layer 3-21 Costs of congestion: scenario 1 r two senders, two receivers r one router, infinite buffers r no retransmission unlimited shared output link buffers Host A in : original data Host B out As the link capacity R is approached, the delay in the system increases.

22 Transport Layer 3-22 Costs of congestion: scenario 2 r one router, finite buffers, reliable data transfer r sender retransmission of lost packet finite shared output link buffers Host A in : original data Host B out ' in : original data, plus retransmitted data (offered load)

23 Transport Layer 3-23 Costs of congestion: scenario 2 r Reduced transmission rate m With finite buffers, packets will be dropped and re- transmissions are required m Depending on the length of timeout interval, retransmissions may occur for packets that are delayed (not lost) and hence the transmission rate decreases.

24 Transport Layer 3-24 Causes/costs of congestion: scenario 3 r four senders r multihop paths r timeout/retransmit finite shared output link buffers Host A Host B Consider what happens when 2 senders compete Host C Host D

25 Transport Layer 3-25 Costs of congestion: scenario 3 r when packet dropped, any “upstream" transmission capacity used for that packet was wasted!

26 Transport Layer 3-26 Approaches towards congestion control End-to-end congestion control: r no explicit feedback from network r congestion inferred from end-system observed loss, delay r approach taken by TCP Network-assisted congestion control: r routers provide feedback to end systems m bits indicating congestion m explicit rate sender should send at Two broad approaches towards congestion control:

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