Transport Protocols Reading: Sections 2.5, 5.1, and 5.2 COS

Transport Protocols Reading: Sections 2.5, 5.1, and 5.2 COS 461: Computer Networks Spring 2006 (MW 1:30-2:50 in Friend 109) Jennifer Rexford Teaching Assistant: Mike Wawrzoniak http://www.cs.princeton.edu/courses/archive/spring06/cos461/ 1

Goals for Today’s Lecture Principles underlying transport-layer services – (De)multiplexing – Detecting corruption – Reliable delivery – Flow control Transport-layer protocols in the Internet – User Datagram Protocol (UDP) – Transmission Control Protocol (TCP) 2

Role of Transport Layer Application layer – Communication for specific applications – E.g., HyperText Transfer Protocol (HTTP), File Transfer Protocol (FTP), Network News Transfer Protocol (NNTP) Transport layer – Communication between processes (e.g., socket) – Relies on network layer and serves the application layer – E.g., TCP and UDP Network layer – Logical communication between nodes – Hides details of the link technology – E.g., IP 3

Transport Protocols Provide logical communication between application processes running on different hosts network data link physical e al d -en nd network data link physical p ns tra ort Multiple transport protocol available to applications – Internet: TCP and UDP network data link physical ic log Run on end hosts – Sender: breaks application messages into segments, and passes to network layer – Receiver: reassembles segments into messages, passes to application layer application transport network data link physical network data link physical network data link physical application transport network data link physical 4

Internet Transport Protocols Datagram messaging service (UDP) – No-frills extension of “best-effort” IP Reliable, in-order delivery (TCP) – Connection set-up – Discarding of corrupted packets – Retransmission of lost packets – Flow control – Congestion control (next lecture) Other services not available – Delay guarantees – Bandwidth guarantees 5

Multiplexing and Demultiplexing Host receives IP datagrams – Each datagram has source and destination IP address, – Each datagram carries one transport-layer segment – Each segment has source and destination port number Host uses IP addresses and port numbers to direct the segment to appropriate socket 32 bits source port # dest port # other header fields application data (message) TCP/UDP segment format 6

Unreliable Message Delivery Service Lightweight communication between processes – Avoid overhead and delays of ordered, reliable delivery – Send messages to and receive them from a socket User Datagram Protocol (UDP) – IP plus port numbers to support (de)multiplexing – Optional error checking on the packet contents SRC port DST port checksum length DATA 7

Why Would Anyone Use UDP? Finer control over what data is sent and when – As soon as an application process writes into the socket – UDP will package the data and send the packet No delay for connection establishment – UDP just blasts away without any formal preliminaries – which avoids introducing any unnecessary delays No connection state – No allocation of buffers, parameters, sequence #s, etc. – making it easier to handle many active clients at once Small packet header overhead – UDP header is only eight-bytes long 8

Popular Applications That Use UDP Multimedia streaming – Retransmitting lost/corrupted packets is not worthwhile – By the time the packet is retransmitted, it’s too late – E.g., telephone calls, video conferencing, gaming Simple query protocols like Domain Name System – Overhead of connection establishment is overkill – Easier to have application retransmit if needed “Address for www.cnn.com?” “12.3.4.15” 9

Transmission Control Protocol (TCP) Connection oriented – Explicit set-up and tear-down of TCP session Stream-of-bytes service – Sends and receives a stream of bytes, not messages Reliable, in-order delivery – Checksums to detect corrupted data – Acknowledgments & retransmissions for reliable delivery – Sequence numbers to detect losses and reorder data Flow control – Prevent overflow of the receiver’s buffer space Congestion control – Adapt to network congestion for the greater good 10

An Analogy: Talking on a Cell Phone Alice and Bob on their cell phones – Both Alice and Bob are talking What if Alice couldn’t understand Bob? – Bob asks Alice to repeat what she said What if Bob hasn’t heard Alice for a while? – Is Alice just being quiet? – Or, have Bob and Alice lost reception? – How long should Bob just keep on talking? – Maybe Alice should periodically say “uh huh” – or Bob should ask “Can you hear me now?” 11

Some Take-Aways from the Example Acknowledgments from receiver – Positive: “okay” or “ACK” – Negative: “please repeat that” or “NACK” Timeout by the sender (“stop and wait”) – Don’t wait indefinitely without receiving some response – whether a positive or a negative acknowledgment Retransmission by the sender – After receiving a “NACK” from the receiver – After receiving no feedback from the receiver 12

Challenges of Reliable Data Transfer Over a perfectly reliable channel – All of the data arrives in order, just as it was sent – Simple: sender sends data, and receiver receives data Over a channel with bit errors – All of the data arrives in order, but some bits corrupted – Receiver detects errors and says “please repeat that” – Sender retransmits the data that were corrupted Over a lossy channel with bit errors – Some data are missing, and some bits are corrupted – Receiver detects errors but cannot always detect loss – Sender must wait for acknowledgment (“ACK” or “OK”) – and retransmit data after some time if no ACK arrives13

TCP Support for Reliable Delivery Checksum – – Sequence numbers – – Used to detect corrupted data at the receiver leading the receiver to drop the packet Used to detect missing data . and for putting the data back in order Retransmission – – – Sender retransmits lost or corrupted data Timeout based on estimates of round-trip time Fast retransmit algorithm for rapid retransmission 14

TCP Segments 15

TCP “Stream of Bytes” Service Host A Byte 80 Byte 3 Byte 2 Byte 1 Byte 0 Host B Byte 80 Byte 3 Byte 2 Byte 1 Byte 0 16

Emulated Using TCP “Segments” Host A Byte 80 Byte 3 Byte 2 Byte 1 Byte 0 Segment sent when: TCP Data 1. 2. 3. Segment full (Max Segment Size), Not full, but times out, or “Pushed” by application. TCP Data Host B Byte 80 Byte 3 Byte 2 Byte 1 Byte 0 17

TCP Segment IP Data TCP Data (segment) TCP Hdr IP Hdr IP packet – No bigger than Maximum Transmission Unit (MTU) – E.g., up to 1500 bytes on an Ethernet TCP packet – IP packet with a TCP header and data inside – TCP header is typically 20 bytes long TCP segment – No more than Maximum Segment Size (MSS) bytes – E.g., up to 1460 consecutive bytes from the stream 18

Sequence Numbers Host A ISN (initial sequence number) Sequence number 1st byte TCP Data TCP HDR ACK sequence number next expected byte TCP Data TCP HDR Host B 19

Initial Sequence Number (ISN) Sequence number for the very first byte – E.g., Why not a de facto ISN of 0? Practical issue – IP addresses and port #s uniquely identify a connection – Eventually, though, these port #s do get used again – and there is a chance an old packet is still in flight – and might be associated with the new connection So, TCP requires changing the ISN over time – Set from a 32-bit clock that ticks every 4 microseconds – which only wraps around once every 4.55 hours! But, this means the hosts need to exchange ISNs 20

TCP Three-Way Handshake 21

Establishing a TCP Connection A SY N CK A N Y S ACK B Each host tells its ISN to the other host. Data Data Three-way handshake to establish connection – Host A sends a SYN (open) to the host B – Host B returns a SYN acknowledgment (SYN ACK) – Host A sends an ACK to acknowledge the SYN ACK 22

TCP Header Source port Destination port Sequence number Flags: SYN FIN RST PSH URG ACK Acknowledgment HdrLen 0 Flags Advertised window Checksum Urgent pointer Options (variable) Data 23

Step 1: A’s Initial SYN Packet A’s port B’s port A’s Initial Sequence Number Flags: SYN FIN RST PSH URG ACK Acknowledgment 20 Flags 0 Checksum Advertised window Urgent pointer Options (variable) A tells B it wants to open a connection 24

Step 2: B’s SYN-ACK Packet B’s port A’s port B’s Initial Sequence Number Flags: SYN FIN RST PSH URG ACK A’s ISN plus 1 20 Flags 0 Checksum Advertised window Urgent pointer Options (variable) B tells A it accepts, and is ready to hear the next byte upon receiving this packet, A can start sending data 25

Step 3: A’s ACK of the SYN-ACK A’s port B’s port Sequence number Flags: SYN FIN RST PSH URG ACK B’s ISN plus 1 20 Flags 0 Checksum Advertised window Urgent pointer Options (variable) A tells B it wants is okay to start sending upon receiving this packet, B can start sending data 26

What if the SYN Packet Gets Lost? Suppose the SYN packet gets lost – Packet is lost inside the network, or – Server rejects the packet (e.g., listen queue is full) Eventually, no SYN-ACK arrives – Sender sets a timer and wait for the SYN-ACK – and retransmits the SYN-ACK if needed How should the TCP sender set the timer? – Sender has no idea how far away the receiver is – Hard to guess a reasonable length of time to wait – Some TCPs use a default of 3 or 6 seconds 27

SYN Loss and Web Downloads User clicks on a hypertext link – Browser creates a socket and does a “connect” – The “connect” triggers the OS to transmit a SYN If the SYN is lost – The 3-6 seconds of delay may be very long – The user may get impatient – and click the hyperlink again, or click “reload” User triggers an “abort” of the “connect” – Browser creates a new socket and does a “connect” – Essentially, forces a faster send of a new SYN packet! – Sometimes very effective, and the page comes fast 28

TCP Retransmissions 29

Automatic Repeat reQuest (ARQ) Automatic Repeat Request – Receiver sends acknowledgment (ACK) when it receives packet – Sender waits for ACK and timeouts if it does not arrive within some time period Timeout Sender Receiver Packe t ACK Simplest ARQ protocol – Stop and wait – Send a packet, stop and wait until ACK arrives Time 30

AC K Packet lost ACK Packe t ACK ACK lost DUPLICATE PACKET Timeout Packe t Timeout Packe t Timeout Packe t Timeout Timeout Timeout Reasons for Retransmission Packe t K C A Packe t ACK Early timeout DUPLICATE PACKETS 31

How Long Should Sender Wait? Sender sets a timeout to wait for an ACK – Too short: wasted retransmissions – Too long: excessive delays when packet lost TCP sets timeout as a function of the RTT – Expect ACK to arrive after an RTT – plus a fudge factor to account for queuing But, how does the sender know the RTT? – Can estimate the RTT by watching the ACKs – Smooth estimate: keep a running average of the RTT EstimatedRTT a * EstimatedRTT (1 –a ) * SampleRTT – Compute timeout: TimeOut 2 * EstimatedRTT 32

Example RTT Estimation RTT: gaia.cs.umass.edu to fantasia.eurecom.fr 350 RTT (milliseconds) 300 250 200 150 100 1 8 15 22 29 36 43 50 57 64 71 78 85 92 99 106 time (seconnds) SampleRTT Estimated RTT 33

A Flaw in This Approach An ACK doesn’t really acknowledge a transmission – Rather, it acknowledges receipt of the data Consider a retransmission of a lost packet – If you assume the ACK goes with the 1st transmission – the SampleRTT comes out way too large Consider a duplicate packet – If you assume the ACK goes with the 2nd transmission – the Sample RTT comes out way too small Simple solution in the Karn/Partridge algorithm – Only collect samples for segments sent one single time 34

Yet Another Limitation Doesn’t consider variance in the RTT – If variance is small, the EstimatedRTT is pretty accurate – but, if variance is large, the estimate isn’t all that good Better to directly consider the variance – Consider difference: SampleRTT – EstimatedRTT – Boost the estimate based on the difference Jacobson/Karels algorithm – See Section 5.2 of the Peterson/Davie book for details 35

TCP Sliding Window 36

Motivation for Sliding Window Stop-and-wait is inefficient – Only one TCP segment is “in flight” at a time – Especially bad when delay-bandwidth product is high Numerical example – 1.5 Mbps link with a 45 msec round-trip time (RTT) Delay-bandwidth product is 67.5 Kbits (or 8 KBytes) – But, sender can send at most one packet per RTT Assuming a segment size of 1 KB (8 Kbits) leads to 8 Kbits/segment / 45 msec/segment 182 Kbps That’s just one-eighth of the 1.5 Mbps link capacity 37

Sliding Window Allow a larger amount of data “in flight” – Allow sender to get ahead of the receiver – though not too far ahead Sending process TCP Last byte written Last byte ACKed Last byte sent Receiving process TCP Last byte read Next byte expected Last byte received 38

Receiver Buffering Window size – Amount that can be sent without acknowledgment – Receiver needs to be able to store this amount of data Receiver advertises the window to the receiver – Tells the receiver the amount of free space left – and the sender agrees not to exceed this amount Window Size Data ACK’d Outstanding Un-ack’d data Data OK to send Data not OK to send yet 39

TCP Header for Receiver Buffering Source port Destination port Sequence number Flags: SYN FIN RST PSH URG ACK Acknowledgment HdrLen 0 Flags Advertised window Checksum Urgent pointer Options (variable) Data 40

Fast Retransmission 41

Timeout is Inefficient Timeout-based retransmission – Sender transmits a packet and waits until timer expires – and then retransmits from the lost packet onward 42

Fast Retransmission Better solution possible under sliding window – Although packet n might have been lost – packets n 1, n 2, and so on might get through Idea: have the receiver send ACK packets – ACK says that receiver is still awaiting nth packet And repeated ACKs suggest later packets have arrived – Sender can view the “duplicate ACKs” as an early hint that the nth packet must have been lost and perform the retransmission early Fast retransmission – Sender retransmits data after the triple duplicate ACK 43

Effectiveness of Fast Retransmit When does Fast Retransmit work best? – Long data transfers High likelihood of many packets in flight – High window size High likelihood of many packets in flight – Low burstiness in packet losses Higher likelihood that later packets arrive successfully Implications for Web traffic – Most Web transfers are short (e.g., 10 packets) Short HTML files or small images – So, often there aren’t many packets in flight – making fast retransmit less likely to “kick in” – Forcing users to like “reload” more often 44

Tearing Down the Connection 45

Tearing Down the Connection ACK FI N Data ACK FIN C FIN A K K A ACK C SYN A SY N B time Closing the connection – Finish (FIN) to close and receive remaining bytes – And other host sends a FIN ACK to acknowledge – Reset (RST) to close and not receive remaining bytes 46

Sending/Receiving the FIN Packet Sending a FIN: close() – Process is done sending data via the socket – Process invokes “close()” to close the socket – Once TCP has sent all of the outstanding bytes – then TCP sends a FIN Receiving a FIN: EOF – Process is reading data from the socket – Eventually, the attempt to read returns an EOF 47

Conclusions Transport protocols – Multiplexing and demultiplexing – Sequence numbers – Window-based flow control – Timer-based retransmission – Checksum-based error detection Reading for this week – Sections 2.5, 5.1-5.2, and 6.1-6.4 Next lecture – Congestion control 48