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learn about transport layer protocols in the Internet: UDP: connectionless transport TCP: connection-oriented transport TCP congestion control. Our goals: understand principles behind transport layer services: multiplexing/ demultiplexing reliable data transfer flow control
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learn about transport layer protocols in the Internet: • UDP: connectionless transport • TCP: connection-oriented transport • TCP congestion control Our goals: • understand principles behind transport layer services: • multiplexing/demultiplexing • reliable data transfer • flow control • congestion control Transport Layer
Transport Layer – Topics • Review: multiplexing, connection and connectionless transport, services provided by a transport layer • UDP • Reliable transport • Tools for reliable transport layer • Error detection, ACK/NACK, ARQ • Approaches to reliable transport • Go-Back-N • Selective repeat • TCP • Services • TCP: Connection setup, acks and seq num, timeout and triple-dup ack, slow-start, congestion avoidance.
application transport network link physical application transport network link physical application transport network link physical application transport network link physical application transport network link physical application transport network link physical Transport Layer messages Key transport layer service: Send messages between Apps Just specify the destination and the message and that’s it Web Browser App Google Server App Transport Transport Network Network Key service the transport layer requires: Network should attempt to deliver segements.
Transport layer • Transfers messages between application in hosts • For ftp you exchange files and directory information. • For http you exchange requests and replies/files • For smtp messages are exchanged • Services possibly provided • Reliability • Error detection/correction • Flow/congestion control • Multiplexing (support several messages being transported simultaneously)
Connection oriented / connectionless • TCP supports the idea of a connection • Once listen and connect complete, there is a logical connection between the hosts. • One can determine if the message was sent • UDP is connectionless • Packets are just sent. There is no concept (supported by the transport layer) of a connection • But the application can make a connection over UDP. So the application is each host will support the hand-shaking and monitoring the state of the “connection.” • There are other transport layer protocols such as SCTP besides TCP and UDP, but TCP and UDP are the most popular
Connection oriented Connections must be set up The state of the connection can be determined Flow/congestion control Limits congestion in the network and end hosts Control how fast data can be sent Larger Packet header Automatically retransmits lost packets and reports if the message was not successfully transmitted Check sum for error detection Connectionless Connections do not need to be set-up No feedback provided as to whether packets were successfully delivered No flow/congestion control Could cause excessive congestion and unfair usage Data can be sent exactly when it needs to be Low overhead Check sum for error detection TCP vs. UDP
Applications and Transport Protocols Application TCP or UDP? SMTP TCP Telnet TCP HTTP TCP FTP TCP NFS TCP or UDP Multimedia streaming via youtude TCP VoIP via Skype UDP DNS UDP
App Transport SP: 5775 SP: 9157 P3 P1 P1 P2 P4 P5 P6 DP: 80 DP: 80 Network Multiplexing with ports Transport layer packet headers always contain source and destination port IP headers have source and destination IPs When a message is sent, the destination port must be known. However, the source port could be selected by the OS. Client IP:B client IP: A server IP: C S-IP: B D-IP:C TCP SP: 9157 DP: 80 S-IP: A D-IP:C S-IP: B TCP D-IP:C TCP
About multiplexing • HTTP usually has port 80 as the destination, but you can make a web server listen on any port that is not already used by another application • ICANN registered ports (0-1024) • HTTP: 80 • HTTP over SSL: 443 • FTP: 21 • Telnet: 23 • DNS: 53 • Microsoft server: 3389 • … • Typically, only one application can listen on a port at a time (tools such as PCAP can be used to listen on ports that are already in use. Wireshark uses PCAP) • For TCP, you cannot control the source port; the OS sets it. For UDP, you can set the source port. • A connection is defined as a 5 tuple: source IP, source port, destination IP, and destination port, and transport protocol. • NATs make use to these five pieces of information. NATs are discussed in detail in Chapter 4, but they are dependent on transport layer • Since connections are defined by ports and addresses, there cross layer dependencies (the transport layer cannot demultiplex without knowledge of the IP addresses, with is a concept of a different layer.)
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 Chapter 3 outline
“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 Why is there a UDP? no connection establishment (which can add delay) simple: no connection state at sender, receiver small segment header no congestion control: UDP can blast away as fast as desired UDP: User Datagram Protocol [RFC 768]
often used for streaming multimedia apps loss tolerant rate sensitive other UDP uses DNS SNMP reliable transfer over UDP: add reliability at application layer application-specific error recovery! 32 bits source port # dest port # Length, in bytes of UDP segment, including header checksum length Application data (message) UDP segment format UDP: more
Sender: treat segment contents as sequence of 16-bit integers checksum: addition (1’s complement sum) of segment contents sender puts checksum value into UDP checksum field Receiver: compute checksum of received segment check if computed checksum equals checksum field value: NO - error detected YES - no error detected. But maybe errors nonetheless? More later …. UDP checksum Goal: detect “errors” (e.g., flipped bits) in transmitted segment
Internet Checksum Example • Note • When adding numbers, a carryout from the most significant bit needs to be added to the result • Example: add two 16-bit integers 1 1 1 1 0 0 1 1 0 0 1 1 0 0 1 1 0 1 1 1 0 1 0 1 0 1 0 1 0 1 0 1 0 1 1 1 0 1 1 1 0 1 1 1 0 1 1 1 0 1 1 1 1 0 1 1 1 0 1 1 1 0 1 1 1 1 0 0 1 0 1 0 0 0 1 0 0 0 1 0 0 0 0 1 1 wraparound sum checksum
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 Chapter 3 outline
rdt_send():called from above, (e.g., by app.). Passed data to deliver to receiver upper layer deliver_data():called by rdt to deliver data to upper udt_send():called by rdt, to transfer packet over unreliable channel to receiver rdt_rcv():called when packet arrives on rcv-side of channel Reliable data transfer: getting started send side receive side
Application implemented reliable data transfer Application Application Main App Main App Application Layer reliable channel communication communication Transport Layer unreliable channel UDP UDP Pros and cons of implementing a reliable transport protocol in the application • Cons • It is already done by the OS, why “reinvent the wheel.” • The OS might have higher priority than the application. • Pros • The OS’s TCP is designed to work in every scenario, but your app might only exist in specific scenarios • Network storage device • Mobile phone • Cloud app
We’ll: incrementally develop sender, receiver sides of reliable data transfer protocol (rdt) consider only unidirectional data transfer but control info will flow on both directions! use finite state machines (FSM) to specify sender, receiver event causing state transition actions taken on state transition state: when in this “state” next state uniquely determined by next event event state 1 state 2 actions Reliable data transfer: getting started
Assume that the underlying channel is perfectly reliable no bit errors no loss of packets Make separate FSMs for sender, receiver: sender sends data into underlying channel receiver read data from underlying channel rdt_send(data) rdt_rcv(segment) Wait for call from below Wait for call from above data = extract (segment) deliver_data(data) sender receiver Rdt1.0: reliable transfer over a reliable channel segment = make_pkt(data) udt_send(segment)
underlying channel may flip bits in packets checksum to detect bit errors the question: how to recover from errors: negative acknowledgements (NAKs): receiver explicitly tells sender that pkt had errors sender retransmits pkt on receipt of NAK acknowledgements (ACKs): receiver explicitly tells sender that pkt received OK new mechanisms in rdt2.0 (beyond rdt1.0): error detection receiver feedback: control msgs (ACK,NAK) rcvr->sender Rdt2.0: channel with bit errors
rdt2.0: FSM specification rdt_send(data) receiver snkpkt = make_pkt(data, checksum) udt_send(sndpkt) Wait for call from above Wait for ACK or NAK Wait for call from below sender rdt_rcv(rcvpkt) && notcorrupt(rcvpkt) extract(rcvpkt,data) deliver_data(data) udt_send(ACK)
Wait for call from below rdt2.0: FSM specification rdt_send(data) receiver snkpkt = make_pkt(data, checksum) udt_send(sndpkt) Wait for call from above Wait for ACK or NAK rdt_rcv(rcvpkt) && isACK(rcvpkt) sender rdt_rcv(rcvpkt) && notcorrupt(rcvpkt) data = extract(rcvpkt) deliver_data(data) udt_send(ACK)
Wait for call from below rdt2.0: FSM specification rdt_send(data) receiver snkpkt = make_pkt(data, checksum) udt_send(sndpkt) rdt_rcv(rcvpkt) && isNAK(rcvpkt) Wait for call from above Wait for ACK or NAK rdt_rcv(rcvpkt) && corrupt(rcvpkt) udt_send(sndpkt) udt_send(NAK) rdt_rcv(rcvpkt) && isACK(rcvpkt) sender rdt_rcv(rcvpkt) && notcorrupt(rcvpkt) data = extract(rcvpkt) deliver_data(data) udt_send(ACK)
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 garbled sender adds sequence number to each pkt receiver discards (doesn’t deliver up) duplicate pkt stop and wait Sender sends one packet, then waits for receiver response rdt2.0 has a fatal flaw!
Wait for ACK or NAK 1 rdt2.1: sender, handles garbled ACK/NAKs rdt_send(data) sndpkt = make_pkt(0, data, checksum) udt_send(sndpkt) rdt_rcv(rcvpkt) && (corrupt(rcvpkt) || isNAK(rcvpkt) ) Wait for ACK or NAK 0 Wait for call 0 from above udt_send(sndpkt) rdt_rcv(rcvpkt) && notcorrupt(rcvpkt) && isACK(rcvpkt) rdt_rcv(rcvpkt) && notcorrupt(rcvpkt) && isACK(rcvpkt) Wait for call 1 from above rdt_rcv(rcvpkt) && ( corrupt(rcvpkt) || isNAK(rcvpkt) ) rdt_send(data) sndpkt = make_pkt(1, data, checksum) udt_send(sndpkt) udt_send(sndpkt)
rdt2.1: receiver, handles garbled ACK/NAKs rdt_rcv(rcvpkt) && !corrupt(rcvpkt) && has_seq0(rcvpkt) extract(rcvpkt,data) deliver_data(data) sndpkt = make_pkt(ACK, chksum) udt_send(sndpkt) Wait for 0 from below Wait for 1 from below
rdt2.1: receiver, handles garbled ACK/NAKs rdt_rcv(rcvpkt) && !corrupt(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(NAK, chksum) udt_send(sndpkt) Wait for 0 from below Wait for 1 from below rdt_rcv(rcvpkt) && ! corrupt(rcvpkt) && seqnum(rcvpkt)==1 sndpkt = make_pkt(ACK, chksum) udt_send(sndpkt)
rdt2.1: receiver, handles garbled ACK/NAKs rdt_rcv(rcvpkt) && !corrupt(rcvpkt) && has_seq0(rcvpkt) extract(rcvpkt,data) deliver_data(data) sndpkt = make_pkt(ACK, chksum) udt_send(sndpkt) rdt_rcv(rcvpkt) && (corrupt(rcvpkt) rdt_rcv(rcvpkt) && (corrupt(rcvpkt) sndpkt = make_pkt(NAK, chksum) udt_send(sndpkt) sndpkt = make_pkt(NAK, chksum) udt_send(sndpkt) Wait for 0 from below Wait for 1 from below rdt_rcv(rcvpkt) && ! corrupt(rcvpkt) && seqnum(rcvpkt)==1 rdt_rcv(rcvpkt) && not corrupt(rcvpkt) && has_seq0(rcvpkt) sndpkt = make_pkt(ACK, chksum) udt_send(sndpkt) sndpkt = make_pkt(ACK, chksum) udt_send(sndpkt) rdt_rcv(rcvpkt) && !corrupt(rcvpkt) && has_seq1(rcvpkt) extract(rcvpkt,data) deliver_data(data) sndpkt = make_pkt(ACK, chksum) udt_send(sndpkt)
Wait for ACK or NAK 0 Wait for call 1 from above Wait for ACK or NAK 1 rdt2.1: sender, handles garbled ACK/NAKs rdt_send(data) sndpkt = make_pkt(0, data, checksum) udt_send(sndpkt) rdt_rcv(rcvpkt) && ( corrupt(rcvpkt) || isNAK(rcvpkt) ) Wait for call 0 from above udt_send(sndpkt) rdt_rcv(rcvpkt) && notcorrupt(rcvpkt) && isACK(rcvpkt) rdt_rcv(rcvpkt) && notcorrupt(rcvpkt) && isACK(rcvpkt) L L rdt_rcv(rcvpkt) && ( corrupt(rcvpkt) || isNAK(rcvpkt) ) rdt_send(data) sndpkt = make_pkt(1, data, checksum) udt_send(sndpkt) udt_send(sndpkt)
Wait for 0 from below Wait for 1 from below rdt2.1: receiver, handles garbled ACK/NAKs 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) rdt_rcv(rcvpkt) && (corrupt(rcvpkt) sndpkt = make_pkt(NAK, chksum) udt_send(sndpkt) sndpkt = make_pkt(NAK, chksum) udt_send(sndpkt) rdt_rcv(rcvpkt) && not corrupt(rcvpkt) && has_seq1(rcvpkt) rdt_rcv(rcvpkt) && not corrupt(rcvpkt) && has_seq0(rcvpkt) sndpkt = make_pkt(ACK, chksum) udt_send(sndpkt) sndpkt = make_pkt(ACK, chksum) udt_send(sndpkt) rdt_rcv(rcvpkt) && notcorrupt(rcvpkt) && has_seq1(rcvpkt) extract(rcvpkt,data) deliver_data(data) sndpkt = make_pkt(ACK, chksum) udt_send(sndpkt)
Sender: seq # added to pkt two seq. #’s (0,1) will suffice. Why? must check if received ACK/NAK corrupted twice as many states state must “remember” whether “current” pkt has 0 or 1 seq. # 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 rdt2.1: discussion
same functionality as rdt2.1, using ACKs only instead of NAK, receiver sends ACK for last pkt received OK receiver must explicitly include seq # of pkt being ACKed duplicate ACK at sender results in same action as NAK: retransmit current pkt rdt2.2: a NAK-free protocol
Wait for call 0 from above Wait for ACK 0 Wait for 0 from below rdt2.2: sender, receiver fragments rdt_send(data) sndpkt = make_pkt(0, data, checksum) udt_send(sndpkt) rdt_rcv(rcvpkt) && ( corrupt(rcvpkt) || isACK(rcvpkt,1) ) udt_send(sndpkt) sender FSM fragment rdt_rcv(rcvpkt) && notcorrupt(rcvpkt) && isACK(rcvpkt,0) rdt_rcv(rcvpkt) && (corrupt(rcvpkt) || has_seq1(rcvpkt)) L receiver FSM fragment udt_send(sndpkt) rdt_rcv(rcvpkt) && notcorrupt(rcvpkt) && has_seq1(rcvpkt) extract(rcvpkt,data) deliver_data(data) sndpkt = make_pkt(ACK1, chksum) udt_send(sndpkt) What happens if a pkt is duplicated?
New assumption: underlying channel can also lose packets (data or ACKs) checksum, seq. #, ACKs, retransmissions will be of help, but not enough Approach: sender waits “reasonable” amount of time for ACK retransmits if no ACK received in this time if pkt (or ACK) just delayed (not lost): retransmission will be duplicate, but use of seq. #’s already handles this receiver must specify seq # of pkt being ACKed requires countdown timer rdt3.0: channels with errors and loss
Wait for ACK0 Wait for call 1 from above Wait for call 0from above rdt3.0 sender rdt_send(data) rdt_rcv(rcvpkt) && ( corrupt(rcvpkt) || isACK(rcvpkt,1) ) sndpkt = make_pkt(0, data, checksum) udt_send(sndpkt) start_timer timeout udt_send(sndpkt) start_timer rdt_rcv(rcvpkt) && notcorrupt(rcvpkt) && isACK(rcvpkt,0) stop_timer
Wait for ACK0 Wait for ACK1 Wait for call 1 from above Wait for call 0from above rdt3.0 sender rdt_send(data) rdt_rcv(rcvpkt) && ( corrupt(rcvpkt) || isACK(rcvpkt,1) ) sndpkt = make_pkt(0, data, checksum) udt_send(sndpkt) start_timer rdt_rcv(rcvpkt) timeout udt_send(sndpkt) start_timer rdt_rcv(rcvpkt) && notcorrupt(rcvpkt) && isACK(rcvpkt,1) rdt_rcv(rcvpkt) && notcorrupt(rcvpkt) && isACK(rcvpkt,0) stop_timer stop_timer timeout udt_send(sndpkt) start_timer rdt_rcv(rcvpkt) rdt_send(data) rdt_rcv(rcvpkt) && ( corrupt(rcvpkt) || isACK(rcvpkt,0) ) sndpkt = make_pkt(1, data, checksum) udt_send(sndpkt) start_timer
rdt3.0 in action receiver sender receiver sender send pkt0 send pkt0 rec pkt0 send ack0 rec ack0 rec pkt0 send pkt1 send ack0 rec ack0 TO send pkt1 rec pkt1 resend pkt1 send ack1 rec ack1 rec pkt1 send pkt1 send ack1 rec ack1 rec pkt1 send pkt2 time rec pkt2 time
rdt3.0 in action receiver sender send pkt0 receiver sender rec pkt0 send pkt0 send ack0 rec ack0 rec pkt0 send pkt1 send ack0 rec pkt1 TO rec ack0 send ack1 send pkt1 send pkt1 rec pkt1 TO send ack1 rec ack1 rec pkt1 send pkt2 send ack1 send pkt1 rec ack1 rec pkt2 rec pkt1 send no pkt (dupACK) send pkt? send ack2 send ack1 rec ack1 rec ack2 send pkt2 send pkt2 time time
Wait for ACK0 Wait for ACK1 Wait for call 1 from above Wait for call 0from above rdt3.0 sender rdt_send(data) rdt_rcv(rcvpkt) && ( corrupt(rcvpkt) || isACK(rcvpkt,1) ) sndpkt = make_pkt(0, data, checksum) udt_send(sndpkt) start_timer rdt_rcv(rcvpkt) timeout udt_send(sndpkt) start_timer rdt_rcv(rcvpkt) && notcorrupt(rcvpkt) && isACK(rcvpkt,1) rdt_rcv(rcvpkt) && notcorrupt(rcvpkt) && isACK(rcvpkt,0) stop_timer stop_timer timeout udt_send(sndpkt) start_timer rdt_rcv(rcvpkt) rdt_send(data) rdt_rcv(rcvpkt) && ( corrupt(rcvpkt) || isACK(rcvpkt,0) ) sndpkt = make_pkt(1, data, checksum) udt_send(sndpkt) start_timer
rdt3.0 works, but performance stinks ex: 1 Gbps link, 15 ms prop. delay, 8000 bit packet and 100bit ACK: What is the total delay Data transmission delay 8000/109 = 810-6 ACK Transmission delay 100/109 = 10-7 sec Total Delay 215ms + .008 + .0001=30.0081ms Utilization Time transmitting / total time .008 / 30.0081 = 0.00027 This is one pkt every 30msec or 33 kB/sec over a 1 Gbps link! Performance of rdt3.0
rdt3.0: stop-and-wait operation sender receiver first packet bit transmitted, t = 0 last packet bit transmitted, t = L / R first packet bit arrives RTT last packet bit arrives, send ACK ACK arrives, send next packet, t = RTT + L / R
Pipelining: sender allows multiple, “in-flight”, yet-to-be-acknowledged pkts range of sequence numbers must be increased buffering at sender and/or receiver Two generic forms of pipelined protocols: go-Back-N, selective repeat Pipelined protocols
Pipelining: increased utilization sender receiver first packet bit transmitted, t = 0 last bit transmitted, t = L / R first packet bit arrives RTT last packet bit arrives, send ACK last bit of 2nd packet arrives, send ACK last bit of 3rd packet arrives, send ACK ACK arrives, send next packet, t = RTT + L / R Increase utilization by a factor of 3!
Go-back-N: big pic Sender can have up to N unacked packets in pipeline Rcvr only sends cumulative acks Doesn’t ack packet if there’s a gap Sender has timer for oldest unacked packet If timer expires, retransmit all unacked packets Selective Repeat: big pic Sender can have up to N unacked packets in pipeline Rcvracks individual packets Sender maintains timer for each unacked packet When timer expires, retransmit only unack packet Pipelining Protocols
Sender: k-bit seq # in pkt header “window” of up to N, unack’ed pkts allowed Go-Back-N • ACK(n): ACKs all pkts up to, including seq # n - “cumulative ACK” • may receive duplicate ACKs (see receiver) • timer for each in-flight pkt • timeout(n): retransmit pkt n and all higher seq # pkts in window
State of pkts pkts start 0 unACKed pkts window N=12 1 unACKed pkts window Next pkt to be sent N unACKed pkts window N-1 unACKed pkts window Sliding window N unACKed pkts Go-Back-N unACKed pkt Pkt that could be sent ACKed pkt Unused pkt send pkt send pkts ACK arrives Send pkt window N=12
N-1 unACKed pkts window N unACKed pkts window N unACKed pkts window 0 unACKed pkts window unACKed pkt Pkt that could be sent Go-Back-N ACKed pkt Unused pkt N unACKed pkts window ACK arrives Send pkt No ACK arrives …. timeout