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Transport Layer

Transport Layer. Moving Segments. Transport Layer Protocols. Provide a logical communication link between processes running on different hosts as if directly connected Implemented in end systems but not in network routers

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Transport Layer

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  1. Transport Layer Moving Segments

  2. Transport Layer Protocols • Provide a logical communication link between processes running on different hosts as if directly connected • Implemented in end systems but not in network routers • (Possibly) break messages into smaller units, adding transport layer header to create segment • We have two protocols: TCP and UDP

  3. An Analogy • Ann, her brothers and sisters on West Coast; Bill and family on East Coast • Application messages = letters in envelopes • Processes = cousins • Hosts (end systems) = houses • Transport Layer Protocol = Ann and Bill • Network Layer Protocol = Postal Service

  4. UDP • User Datagram Protocol • Provides an unreliable, connectionless service to a process • Provides integrity checking by including error detection fields in segment header

  5. TCP • Transmission Control Protocol • Provides reliable data transfer • Flow control • Sequence numbers • Acknowledgments • Timers • Provides congestion control • Provides integrity through error checking

  6. Multiplexing and Demultiplexing • Multi- is the job of gathering data chunks thru sockets and creating segments • Demulti- is delivering data chunks (segments minus Transport header) to correct socket

  7. Segment Identification • UDP: destination IP address and port number • TCP: source IP, source port, destination IP and destination port

  8. TCP Handshake • Server application has a “welcome socket” that waits for connection requests • Client generates a connection-establishment request (includes source IP and port at client) • Server creates new port (socket) for client • Both sides allocate resources for connection

  9. UDP • Defined in RFC 768 • Does about as little as a transport protocol can do • Attaches source and destination port numbers and passes segment to network layer • No handshaking before segment is sent • DNS uses UDP

  10. Why use UDP? • Finer application-level control over what data is sent and when • No connection establishment (thus no delays) • No connection state information • Only 8 bytes of packet overhead • Out of order receipt can be discarded • Lack of congestion control can lead to high loss rates if network is busy

  11. UDP Checksum • For error detection, can’t fix error(s) • Add (with wrap-around) 16-bit words • Take 1’s compliment (invert 0/1) • Send this value • At receiver, all words are added (including checksum) and result should be 11111111111111111

  12. Principles of Reliable Data Transfer • No transferred data bits are corrupted • All are delivered in the order sent • This gets complicated because lower layer (IP) is a best-effort (no guarantees) delivery service

  13. Stop and Wait protocol • Sender sends packet • Receiver gets packet, checks for accuracy • Receiver sends acknowledgement back • If sender times out, presume NAK and resend packet • Use sequence number to identify packets sent/resent

  14. A little math • West coast to East coast transfer; RTT = 30msec; Channel with 1GHz rate; packet size of 1000 bytes (8000 bits) • Time needed to send packet is 8 microsec • 15.008 msec for packet to get to East coast • Ack packet back to sender after 30.008msec • Utilization is .00027; effective throughput is 267 kbps P 215

  15. Pipelining • Range of sequence numbers must increase • May have to buffer packets on both sides of link • Error response either Go-Back-N or Selective Repeat

  16. Go-Back-N (GBN) • Sender allowed to transmit multiple packets but is constrained to have no more than some maximum, N, not ACK’ed • N is window size and GBN is sliding window protocol

  17. Selective Repeat • Avoids unnecessary retransmission by having the sender retransmit only those packets that it suspects were in error • Big difference is that we will buffer (keep) out-of-order packets

  18. TCP • Client first sends a TCP segment • Server responds with a second segment • Client responds with a third segment (that can optionally have message data) • Connection is point-to-point, not one to many • Can be full duplex

  19. TCP Timer • We need to know when data is lost • We can measure round trip time • Timer expiration could be due to congestion in the network, so… • If timeout, we double timeout value next interval and go back to original value when ACK received.

  20. Flow Control • We have receive buffer. If application is slow to read, we can overflow buffer • Receiver sends value of receive window to sender (with each ACK) • Sender makes sure un-ACK’ed data does not exceed receive window size.

  21. Closing a connection • Client issues a close command (FIN=1) • Server sends ACK • Server sends a close command (FIN=1) • Client sends ACK • Resources are now deallocated

  22. Congestion Control • Theory: As supply (feed) rate increases, output increases to limit of output line and then levels off • T2: As feed rate increases, delay grows exponentially • As feed rate grows, we start loosing packets at router – forcing retransmission • TCP has to infer that this is congestion

  23. TCP Congestion Control • Additive-increase, multiplicative-decrease • Slow start • Reaction to timeout events

  24. Speed Control • Congestion window = amount of data “in the pipeline” • With congestion (lost packet) we halve the window for each occurrence • With ACKs, we increase window by set amount (Maximum Segment Size)

  25. Slow Start • Start at one MSS – send one packet • Double that value each time an ACK comes back – send two, then four, then …

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