Transcript Voice over Mobile IP

Ch 3. Transport Layer
Myungchul Kim
[email protected]
Transport-Layer Service
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A transport-layer protocol: logical communications between
application processes running on different hosts.
Processes vs hosts
A transport protocol can offer reliable data transfer service to an
application even when the underlying ntwork protocol is
unreliable.
IP, a best-effort delivery service: logical communications between
hosts, unreliable service
Extending host-to-host delivery to process-to-process delivery is
called transport-layer multiplexing and demultiplexing.
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Multiplexing and demultiplexing
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Demultiplexing: delivering the data in a transport-layer segment
to the correct socket
Multiplexing: passing the segments to the network layer
16 bit number for port number including 1 to 1023 for well-known
port numbers
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Each socket in the host could be assigned a port number.
A UDP socket is identified by a destination IP address
and destination port number
A TCP socket is identifed by a four-tuple
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Connectionless Transport: UDP
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Multiplexing/demultiplexing and light error checking
No handshaking
Finer application-level control over what data is sent, and when
No connection establishment
No connection state
Small packet header overhead: TCP 20 bytes of header and
UDP 8 bytes of overhead
TCP is increasingly being used for streaming media transport
The lack of congesting control in UDP?
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UDP checksum: the 1’s complement of the sum of all the 16-bit
words in the segment at the sender -> if no errors are introduced
into the packet, the sum at the receiver will be all 1’s.
Errors in a link-layer or a router’s memory
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Principles of reliable data transfer
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Unidirection data transfer
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Reliable data transfer over a perfectly reliable channel rdt1.0
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Reliable data transfer over a channel with bit errors rdt2.0
– Rdt1.0: perfectly reliable channel, no limit of receiving speed
– Error detection and Acknowledgement (ack, nak)
– Automatic Repeat reQuest (ARQ) protocols; error detection,
receiver feedback and retransmission
– Stop-and-wait protocol
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Reliable data transfer over a channel with bit errors rdt2.1
– Rdt 2.0: ack or nak packet could be corrupted
– Sequence number of 0 or 1
– Duplicate acks: a sender that receives two ACKs for the same
packet knows that receivers did not correctly receive the packets
following the duplicated acked packet.
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Reliable data transfer over a channel with bit errors rdt2.2
– a NAK-free protocol
– 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
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Reliable data transfer over a lossy channel with bit errors rdt3.0
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Rdt 2.0: packet loss
Countdown timer
Alternating-bit protocol
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Pipelined reliable data transfer protocols
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Rdt3.0: performance problem due to a stop-and-wait protocol
Transmission delay = L/R = 8000 bits/packet / 10 9 bits/sec
= 8 microseconds.
where R= 1 Gbps, L = 1,000 bytes
Utilization of sender U sender = L/R / (RTT + L/R)
= .008 / 30.008 = 0.00027
where RTT = 30 milliseconds
Effective throughput of only 267kbps for a 1 Gbps link
Pipelining: the sender is allowed to send multiple packets without
waiting for acknowledgements.
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Go-Back-N (GBN)
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Window size
A sliding window protocol
Receipt of an ack at the sender: Cumulative acknowledgement
The receiver discards out-of-order packets -> no buffers at the
receiver
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Selective Repeat (SR)
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The receiver individually acknowledge correctly received packets.
The sender and receiver will not always have an identical view of
what has been received correctly and what has not.
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Connection-oriented Transport: TCP
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Full-duplex service
Point-to-point
Three-way handshake
Maximum segment size (MSS)
Maximum transmission unit (MTU)
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Cumulative acknowledgements
The acknowledgement number that Host A puts in its segment is
the sequence number of the next byte Host A is expecting from
Host B.
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Round-trip time estimation and timeout
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EstimatedRTT = (1-α) * EstimatedRTT + α * SampleRTT
α = 0.125
DevRTT = (1 -β)*DevRTT + β * |SampleRTT – EstimatedRTT|
TimeoutInterval = EstimatedRTT + 4* DevRTT
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Reliable data transfer
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A single retransmission timer; associate with the oldest
unacknowledged segment
TCP uses cumulative acknowledgement
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Doubling the Timeout Interval
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Fast retransmit
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Each time TCP retransmits, it sets the next timeout interval to
twice the previous value.
A duplicate ACK is an ACK that reacknowledgements a segment
for which the sender has already received an earlier
acknowledgement.
Three duplicate ACKs -> TCP sender performs a fast retransmit,
retransmitting the missing segment before that segment’s timer
expires.
why three?
TCP is a hybrid of GBN and SR protocols.
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Flow control
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Eliminate the possibility of the sender overflowing the receiver’s
buffer
Sender maintains a variable called the receive window
LastByteRead: the number of the last byte in the data stream
read from the buffer by the application process in B
LastByteRcvd: the number of the last byte in the data stream that
has arrived from the network and has been placed in the receive
buffer at B
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LastByteRcvd – LastByteRead ≤ RcvBuffer
RcvWindow = RcvBuffer – [ LastByteRcvd – LastByteRead]
No overflow at the receive buffer : LastByteSent – LastByteAcked ≤
RcvWindow (at the sender)
Host A continues to send segments with one data byte when B’s receive
window is zero.
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TCP connection management
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Three-way handshake
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Principles of congestion control
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Scenario 1
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Scenario 2
Offered load to the network
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Scenario 3
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Approaches to congestion control
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End-to-end congestion control
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The network layer provides no explicit support to the transport
layer for congestion control purpose.
Congestion is inferred on observed network behavior such
packet loss and delay.
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Network-assisted congestion control
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ATM Available bit-rate (ABR) congestion control
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Network-assisted congestion-control
example: ATM ABR congestion control
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Explicitly signaling to the sender to reduce its rate when the
switch becomes congested.
Resource-management cells (RM cells)
Signaling congestion-related information
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Explicit forward congestion indication (EFCI) bit in data cells
Congestion indication (CI) and No increase (NI) bits in RM cells
Explict rate setting
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TCP congestion control
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Congestion window
LastByteSent – LastByteAcked ≤ min{CongWin, RcvWindow}
At the beginning: sending rate = CongWin / RTT
Additive-Increase, Multiplicative-Decrease (AIMD)
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Halving the current value of CongWin after a loss event
Not allowed to drop below 1 MSS
Increase CongWin by 1 MSS every round-trip time
Congestion avoidance
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Slow start
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A TCP sender increases its rate exponentially by doubling its
value of CongWin every RTT
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Reaction to timeout events
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A triple duplicate ACK -> fast recovery: ½ CongWin
A timeout -> a slow start
Vegas: the longer the RTT of the packets, the greater the
congestion in the routers
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Fairness
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Fairness and UDP
Fairness and parallel TCP connections
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