Elements of Transport
Protocols
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The transport service is implemented by a
transport protocol used between the two
transport entities
Transport protocols resemble the data link
protocols
Both have to deal with error control, sequencing,
and flow control, among other issues.
Differences are due to major dissimilarities
between the environments in which the two
protocols operate
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Elements of Transport Protocols
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Addressing
Connection Establishment
Connection Release
Flow Control and Buffering
Multiplexing
Crash Recovery
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1.Addressing
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When an application (e.g., a user) process wishes
to set up a connection to a remote application
process, it must specify which one to connect to.
The method normally used is to define transport
addresses to which processes can listen for
connection requests. In the Internet, these end
points are called ports.
We will use the generic term TSAP, (Transport
Service Access Point).
The analogous end points in the network layer (i.e.,
network layer addresses) are then called NSAPs.
IP addresses are examples of NSAPs.
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2.Establishing a connection
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1.
2.
It would seem sufficient for one transport entity to just send a
CONNECTION REQUEST TPDU to the destination and wait for a
CONNECTION ACCEPTED reply.
The problem occurs when the network can lose, store, and
duplicate packets. This causes serious complications.
The crux of the problem is the existence of delayed duplicates.
It can be attacked in various ways
using throw-away transport addresses. In this approach, each
time a transport address is needed, a new one is generated.
When a connection is released, the address is discarded and
never used again.
give each connection a connection identifier. After each
connection is released, each transport entity could update a table
listing obsolete connections. Whenever a connection request
comes in, it could be checked against the table, to see if it
belonged to a previously-released connection.
this scheme has a basic flaw: it requires each transport entity to
maintain a certain amount of history information indefinitely. If a
machine crashes and loses its memory, it will no longer know
which connection identifiers
have already been used.
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Rather than allowing packets to live forever
within the subnet, devise a mechanism to kill off
aged packets that are still hobbling about. If no
packet lives longer than some known time, the
problem becomes somewhat more manageable.
Packet lifetime can be restricted to a known
maximum using one of the following techniques:
 Restricted subnet design.
 Putting a hop counter in each packet.
 Timestamping each packet.
3.
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Once both transport entities have agreed on the initial
sequence number, any sliding window protocol can be used
for data flow control.
Three-way handshake protocol is used
This protocol does not require both sides to begin sending
with the same sequence number.
Host 1 chooses a sequence number, x, and sends a
CONNECTION REQUEST TPDU containing it to host 2.
Host 2 replies with an ACK TPDU acknowledging x and
announcing its own initial sequence number, y.
Finally, host 1 acknowledges host 2's choice of an initial
sequence number in the first data TPDU that it sends.
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Connection Establishment
Three protocol scenarios for establishing a connection using a
three-way handshake. CR denotes CONNECTION REQUEST.
(a) Normal operation,
(b) Old CONNECTION REQUEST appearing out of nowhere.
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(c) Duplicate CONNECTION
REQUEST and duplicate ACK.
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Fig b: shows how the three-way handshake works in the
presence of delayed duplicate control TPDUs .
The first TPDU is a delayed duplicate CONNECTION
REQUEST from an old connection.
This TPDU arrives at host 2 without host 1's knowledge.
Host 2 reacts to this TPDU by sending host 1 an ACK TPDU,
in effect asking for verification that host 1 was indeed
trying to set up a new connection.
When Host 1 Rejects Host 2 ack, Then Host 2 realizes that
it was tricked by a delayed duplicate and abandons the
connection. In this way, a delayed duplicate does no
damage
The worst case is when both a delayed CONNECTION
REQUEST and an ACK are floating around in the subnet.
This case is shown in fig c
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host 2 gets a delayed CONNECTION REQUEST and replies to it.
host 2 has proposed using y as the initial sequence number for
host 2 to host 1 traffic, knowing full well that no TPDUs
containing sequence number y or acknowledgements to y are still
in existence.
When the second delayed TPDU arrives at host 2, the fact that z
has been acknowledged rather than y tells host 2 that this, too, is
an old duplicate.
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3. Releasing a connection
Are of 2 types :
 Asymmetric release
Like telephone s/m: when one party hangs
up, the connection is broken.
 Symmetric release
It treats the connection as 2 separate
unidirectional connections and requires
each one to be released separately.
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Asymmetric Release
Abrupt disconnection with loss of data.
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After the connection is established, host 1
sends a TPDU that arrives properly at host
2.
 Then host 1 sends another TPDU.
 Unfortunately, host 2 issues a
DISCONNECT before the second TPDU
arrives.
 The result is that the connection is
released and data are lost.
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Symmetric Release
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A more sophisticated release protocol is needed
to avoid data loss.
One way is to use symmetric release, in which
each direction is released independently of the
other one.
Here, a host can continue to receive data even
after it has sent a DISCONNECT TPDU.
Symmetric release does the job when each
process has a fixed amount of data to send and
clearly knows when it has sent it.
One can envision a protocol in which host 1 says:
I am done. Are you done too? If host 2
responds: I am done too. Goodbye, the
connection can be safely released
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The Two-Army problem
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Imagine that a white army is encamped in a
valley, as shown in fig.
On both of the surrounding hillsides are blue
armies.
The white army is larger than either of the blue
armies alone, but together the blue armies are
larger than the white army.
If either blue army attacks by itself, it will be
defeated, but if the two blue armies attack
simultaneously, they will be victorious.
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The two-army problem.
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The blue armies want to synchronize their
attacks.
 However, their only communication
medium is to send messengers on foot
down into the valley, where they might be
captured and the message lost (i.e., they
have to use an unreliable communication
channel).
 Does a protocol exist that allows the blue
armies to win?
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Suppose that the commander of blue army #1
sends a message reading: ''I propose we attack
at dawn on March 29. How about it?''
Now suppose that the message arrives, the
commander of blue army #2 agrees, and his
reply gets safely back to blue army #1.
Will the attack happen? Probably not, because
commander #2 does not know if his reply got
through. If it did not, blue army #1 will not
attack, so it would be foolish for him to charge
into battle.
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Now improve the protocol by making it a threeway handshake.
The initiator of the original proposal must
acknowledge the response.
Assuming no messages are lost, blue army #2
will get the acknowledgement, but the
commander of blue army #1 will now hesitate.
After all, he does not know if his
acknowledgement got through, and if it did not,
he knows that blue army #2 will not attack.
We could make a four-way handshake protocol,
but that does not help either.
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To see the relevance of the two-army
problem to releasing connections, just
substitute ''disconnect'' for ''attack.'‘
 If neither side is prepared to disconnect
until it is convinced that the other side is
prepared to disconnect too, the
disconnection will never happen.
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Four protocol scenarios for releasing a connection.
(a) Normal case of a three-way handshake. (b)
final ACK lost.
6-14, a, b
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(c) Response lost. (d) Response lost
and subsequent DRs lost.
6-14, c,d
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While this protocol usually works, in theory it can
fail if the initial DR and N retransmissions are all
lost.
The sender will give up and release the
connection, while the other side knows nothing at
all about the attempts to disconnect and is still
fully active.
This situation results in a half-open connection.
We could avoid this problem by not allowing the
sender to give up after N retries but forcing it to
go on forever until it gets a response.
One way to kill off half-open connections is to
have a rule saying that if no TPDUs have arrived
for a certain number of seconds, the connection is
then automatically disconnected
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4.Flow Control and Buffering
How connections are managed when in use:
Flow control
 In some ways the flow control problem in the transport
layer is the same as in the data link layer, but in other
ways it is different.
 The basic similarity is that in both layers a sliding window
or other scheme is needed on each connection to keep a
fast transmitter from overrunning a slow receiver.
 The main difference is that a router usually has relatively
few lines, whereas a host may have numerous connections.
 This difference makes it impractical to implement the data
link buffering strategy in the transport layer.
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if the network service is unreliable, the sender must buffer
all TPDUs sent, just as in the data link layer.
with reliable network service, other trade-offs become
possible.
In particular, if the sender knows that the receiver always
has buffer space, it need not retain copies of the TPDUs it
sends.
if the receiver cannot guarantee that every incoming TPDU
will be accepted, the sender will have to buffer anyway.
In the latter case, the sender cannot trust the network
layer's acknowledgement, because the acknowledgement
means only that the TPDU arrived, not that it was accepted.
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Even if the receiver has agreed to do the buffering, there
still remains the question of the buffer size.
If most TPDUs are nearly the same size, it is natural to
organize the buffers as a pool of identically-sized buffers,
with one TPDU per buffer, as in Fig (a).
if there is wide variation in TPDU size, a pool of fixed-sized
buffers presents problems.
If the buffer size is chosen equal to the largest possible
TPDU, space will be wasted whenever a short TPDU arrives.
If the buffer size is chosen less than the maximum TPDU
size, multiple buffers will be needed for long TPDUs, with
the attendant complexity.
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(a) Chained fixed-size buffers. (b) Chained variablesized buffers.
(c) One large circularwww.techstudent.co.cc
buffer per connection.
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Another approach to the buffer size problem is to
use variable-sized buffers, as in Fig(b).
The advantage here is better memory utilization,
at the price of more complicated buffer
management.
A third possibility is to dedicate a single large
circular buffer per connection, as in Fig. (c).
This system also makes good use of memory,
provided that all connections are heavily loaded,
but is poor if some connections are lightly loaded.
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5.Multiplexing
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In the transport layer the need for multiplexing can arise in a number of
ways.
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For Eg, if only one network address is available on a host, all transport
connections on that machine have to use it.
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When a TPDU comes in, some way is needed to tell which process to give it
to.
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This situation, called upward multiplexing, is shown in fig a.
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In this figure, 4 distinct transport connections all use the same network
connection (e.g., IP address) to the remote host.
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If a user needs more bandwidth than one virtual circuit can provide, a way
out is to open multiple network connections and distribute the traffic among
them on a round-robin basis, as indicated in fig b.
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This modus operandi is called downward multiplexing
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(a) Upward multiplexing.
(b) Downward multiplexing.
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THE INTERNET TRANSPORT
PROTOCOL
(TCP and UDP)
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The Internet has two main protocols in the
transport layer,
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connectionless protocol
connection-oriented protocol
connectionless protocol is UDP
connection-oriented protocol is TCP
UDP is basically just IP with a short header
added
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Introduction to TCP
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TCP (Transmission Control Protocol) was designed to provide a
reliable end-to-end byte stream over an unreliable internetwork
TCP was designed to dynamically adapt to properties of the
internetwork and to be robust in the face of many kinds of
failures.
Each machine supporting TCP has a TCP transport entity, either
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a library procedure
a user process or
part of the kernel
The IP layer gives no guarantee that datagrams will be
delivered properly, so it is up to TCP to time out and retransmit
them as need be.
Datagrams that do arrive may do so in the wrong order; it is
also up to TCP to reassemble them into messages in the proper
sequence.
TCP must furnish the reliability that most users want and that IP
does not provide.
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The TCP Service Model
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TCP service is obtained by both the sender and receiver
creating end points, called sockets.
Each socket has a socket number (address) consisting of
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the IP address of the host and
a 16-bit number local to that host, called a port.
A port is the TCP name for a TSAP.
For TCP service to be obtained, a connection must be
explicitly established between a socket on the sending
machine and a socket on the receiving machine.
A socket may be used for multiple connections at the same
time.
two or more connections may terminate at the same socket.
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Port numbers below 256 are called well-known ports and
are reserved for standard services.
For example, any process wishing to establish a
connection to a host to transfer a file using FTP can
connect to the destination host's port 21 to contact its FTP
daemon .
All TCP connections are full duplex and point-to-point.
Full duplex means that traffic can go in both directions at
the same time.
Point-to-point means that each connection has exactly two
end points.
TCP does not support multicasting or broadcasting.
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Some assigned ports.
Port
21
23
25
69
79
80
110
119
Protocol
FTP
Telnet
SMTP
TFTP
Finger
HTTP
POP-3
NNTP
Use
File transfer
Remote login
E-mail
Trivial File Transfer Protocol
Lookup info about a user
World Wide Web
Remote e-mail access
USENET news
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A TCP connection is a byte stream, not a
message stream.
Message boundaries are not preserved end to
end.
For example, if the sending process does four
512-byte writes to a TCP stream, these data may
be delivered to the receiving process as
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four 512-byte chunks,
two 1024-byte chunks, or
one 2048-byte chunk
There is no way for the receiver to detect the
units in which the data were written.
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(a) Four 512-byte segments sent as separate IP
datagrams.
(b) The 2048 bytes of data delivered to the application in
a single READ CALL.
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The TCP Protocol
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1.
2.
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Every byte on a TCP connection has its own 32-bit sequence
number.
Separate 32-bit sequence numbers are used for acknowledgements
and for the window mechanism.
The sending and receiving TCP entities exchange data in the form of
segments.
A TCP segment consists of a fixed 20-byte header (plus an optional
part) followed by zero or more data bytes.
The TCP software decides how big segments should be.
Two limits restrict the segment size.
Each segment, including the TCP header, must fit in the 65,515byte IP payload.
Each network has a Maximum Transfer Unit, or MTU, and each
segment must fit in the MTU.
In practice, the MTU is generally 1500 bytes (the Ethernet payload
size) and thus defines the upper bound on segment size.
A segment that is too large for a n/w can be broken into multiple
segments by a router
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The basic protocol used by TCP entities is the sliding
window protocol.
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When a sender transmits a segment, it also starts a
timer.
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When the segment arrives at the destination, the
receiving TCP entity sends back a segment (with data
if any exist, otherwise without data) bearing an
acknowledgement number equal to the next sequence
number it expects to receive.
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If the sender's timer goes off before the
acknowledgement is received, the sender transmits
the segment again.
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The TCP Segment Header
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Every segment begins with a fixed-format 20byte header.
The fixed header may be followed by header
options.
After the options, if any, up to 65,535 - 20 - 20 =
65,495 data bytes may follow, where the
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first 20 refer to the IP header and
second to the TCP header.
Segments without any data are legal and are
commonly used for
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acknowledgements and
control messages.
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TCP Header.
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The Source port and Destination port fields identify the
local end points of the connection.
A port plus its host's IP address forms a 48-bit unique end
point (TSAP).
The Sequence number and Acknowledgement number
fields perform their usual functions.
Acknowledgement number specifies the next byte
expected, not the last byte correctly received.
Both are 32 bits long
The TCP header length tells how many 32-bit words are
contained in the TCP header.
This information is needed because the Options field is of
variable length, so the header is, too.
Next comes a 6-bit field that is not used .
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1.
2.
3.
4.
Now come six 1-bit flags.
URG is set to 1 if the Urgent pointer is in use.
 The Urgent pointer is used to indicate a byte offset
from the current sequence number at which urgent data
are to be found.
The ACK bit
 set to 1 to indicate that the Acknowledgement number is
valid.
 If ACK is 0, the segment does not contain an
acknowledgement so the Acknowledgement number
field is ignored.
The PSH bit indicates PUSHed data.
 Applications can use the PUSH flag, which tells TCP not
to delay the transmission.
The RST bit
 used to reset a connection that has become confused
due to a host crash or some other reason.
 It is also used to reject an invalid segment or refuse an
attempt to open a connection.
 if you get a segment
with the RST bit on, you have a
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problem on your hands.
5.
6.
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The SYN bit is used to establish connections.
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The connection request has SYN = 1 and ACK = 0 to
indicate that the piggyback acknowledgement field is
not in use.
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The connection reply bears an acknowledgement it has
SYN = 1 and ACK = 1.
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In essence the SYN bit is used to denote CONNECTION
REQUEST and CONNECTION ACCEPTED, with the ACK
bit used to distinguish between those two possibilities.
The FIN bit is used to release a connection. It specifies
that the sender has no more data to transmit.
Both SYN and FIN segments have sequence numbers and
are thus guaranteed to be processed in the correct order.
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Flow control in TCP is handled using a
variable-sized sliding window.
The Window size field tells how many bytes
may be sent starting at the byte
acknowledged.
A Checksum is also provided for extra reliability.
The Options field provides a way to add extra
facilities not covered by the regular header.
The most important option is the one that allows
each host to specify the maximum TCP payload it
is willing to accept.
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ATM AAL LAYER
PROTOCOLS
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The AAL layer in ATM is radically different than TCP.
This is mainly because ATM is primarily used for
transmitting voice and video streams, in which rapid
delivery is more important than accurate delivery.
ATM layer outputs 53-byte cells one after another.
It has no error control, no flow control and no other
control.
To bridge this gap , ITU defined an end- to-end layer on
top of the ATM layer.
This layer is called AAL(ATM Adaptation Layer).
The goal of AAL is
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to provide useful services to application programs and
to shield them from the mechanics of chopping data up
into cells at the src and reassembling them at the desn.
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When ITU began defining AAL, it realized that
different applications had different
requirements, so it organized the service space
along 3 axes:
1.
2.
3.
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Real-time service VS Non real-time services.
Constant bit rate services VS variable bit rate services.
Connection-oriented service VS connection less service.
ITU felt only 4 of these were of any use and
named them class A,B,C and D.
To handle these 4 classes of service, ITU
defined 4 protocols, AAL1 thru AAL4
respectively.
Technical requirements for classes C & D were
similar
So combined AAL3 & AAL4 to form AAL3/4
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Original service classes supported by
AAL (now obsolete)
B
A
Timing Real
Time
None
Real
Time
Bit
Rate
Constant
Mode
Connection oriented
D
C
None
Variable
Real
Time
None
Constant
Real
Time
Variable
Connection less
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None
Structure of the AAL
Convergence sublayer (service specific part)
AAL
Convergence sublayer (common part)
Segmentation Reassembly sublayer
ATM layer
Physical layer
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The AAL is divided into 2 major parts.
 The upper part of AAL is called Convergence
Sublayer.
 Its job is to provide interface to the application.
 It consists of 2 subparts that is
1.
2.
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common to all applications and
an application specific sub part
The functions of each of these parts are protocol
dependent but can include msg framing and error
detection.
It is also responsible for accepting bit streams and
breaking them up into 44-48 bytes for transmition .
Message boundaries are preserved when present
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The lower layer of AAL is called SAR (Segmentation
And Reassembly sublayer.
It can add headers and trailers to the data units given
to it by the CS to form cell payloads.
These payloads are then given to the ATM layer for
transmition.
At the destn the SAR sublayer reassembles the cells
into msgs.
The SAR sublayer is basically concerned with cells, but
the CS sublayer is concerned with msgs.
It has some additional functions for some service
classes
It sometime handles error detection & multiplexing
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AAL1
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Is the protocol used for transmiting class A traffic, that is
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real-time,
constant bit-rate ,
connection-oriented traffic- eg: uncompressed audio
and video.
Bits are fed in by the application at a constant rate and
must be delivered at destn at the same constant rate ,
with a min. of delay and overhead.
The input is a stream of bits, with no msg boundaries.
For this traffic, error-detecting protocols such as stopand-wait are not used because the delays introduced by
timeouts and retransmition are unacceptable.
Missing cells are reported to the application
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AAL1 uses a Convergence Sublayer and an SAR
sublayer.
The Convergence Sublayer (CS)
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detects lost and misinserted cells.
smoothes out incoming traffic to provide delivery of cells at
a constant rate.
breaks up the inputp msg or stream into 46 or 47 byte
units that r given for SAR for txn.
At the other end it extracts these and reconstructs the
original i/p.
The AAL1 CS does not have any protocol headers of its
own.
But the AAL1 SAR sublayer does have.
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The AAL 1 CELL FORMAT
0
SN
SNP
47-byte Payload
Non-P
Even Parity
1 SN
SNP
46-byte Payload
Parity
48 bytes
SAR header
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P
AAL1 SAR
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It has 1 byte headercontaining a
3 bit cell seq number SN
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to detect missing or misinserted cells
3 bit Sequence Number Protection SNP (like
check sum)
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Allows correction of single errors & detection of double
errors in seq no. field
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One bit for parity –even bit parity
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P cells used when message boundaries must be
preserved
Pointer field -1 byte
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Used to give offset of start of next message
Higher order bit is reserved for future use
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AAL 2
For pure uncompressed audio/video , or any other data stream in
which having a few garbled bits once in a while is not a problemAAL 1 is adequate.
 For compressed audio or video, the rate can vary strongly in
time.
Eg:
 Many compression schemes transmit a full video stream
periodically, and then send only the differences betwn
subsequent frames and the last full frame for several frames.
 When the camera is stationary and nothing is moving , the
differenz betwn frames are small.
 But when the camera is panning rapidly, they r large.
 Also msg boundaries must be preserved so that the start of the
next full frame can be recognized , even in the presence of lost
cells or bad data.

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AAL 3/4
For classes C and D (connection-oriented & connectionless
service)
ITU combined 3 and 4 together to form AAL 3 /4.
It can operate in 2 modes: stream or message.
In msg mode
each call from the application to AAL 3 /4 injects 1 msg
into the n/w.
The msg is delivered as such and boundaries are
preserved.
In stream mode the boundaries are not preserved.
It provides MULTIPLEXING.
 AAL 3/4 allows multiple sessions from a single host to
travel along the same www.techstudent.co.cc
VC and be separated at the destn.
AAL 3/4 CS msg format
Bits
8
8
16
CPI
Btag
BASize
< 64 KB
User data
0– 24 8
Pad
0
8
16
Etag
Len
CS header


CPI-Common Part Indicator, gives the msg type and the
counting unit for the BA size and Length fields.
Btag and Etag are used to frame msgs.


Tells receiver how much buffer space to be allocated for
message in advance of its arrival
The Length field gives the payload length.


These 2 bytes must be same & incremented by 1 on every new
msg sent
The BA size is used for buffer allocation.


CS Trailer
In msg mode length must be equal to BA size..
Trailer has 1 unused byte
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AAL3/4 SAR format
After the CS has constructed and added a header and
trailer to the msg, it passes the msg to the SAR
sublayer, which chops the msg up into 44-byte chunks.
The SAR sublayer inserts 44-byte chunk into the
payload of a cell whose format is shown below.


bits 2
S
T
4
S
N
10
MID
6
44-byte payload
00 middle
1-44
01 End
10 beginning
11 single cell msg
LI
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10
CRC

ST (Segment Type)- for msg framing









00
01
10
11
–middle (continuation of message COM)
–end of message (EOM)
–beginning of message (BOM)
–single segment message (SSM)
SN (Sequence Number) for detecting missing and
misinserted cells.
The MID (Multiplexing Identification) is used to keep
track of which cell belongs to which session.
Trailer consist of
LI (Lenth Indicator)- indicates payload length
CRC –cell checksum www.techstudent.co.cc
AAL 5





The AAL 1 thru AAL 3/4 protocols were largely
designed by the Tele Communications industry and
standardized by ITU without a lot of i/p from the
computer industry.
For computer industry a new protocol is invented
and it was called SEAL (Simple Efficient Adaptation
Layer).
Later it is renamed as AAL 5.
AAL 5 offers both reliable and unreliable services.
It supports both unicast and multicast

For multicast guaranteed delivery is not provided
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




Like AAL 3/4 ,AAL 5 supports both msg mode and
stream mode.
In msg mode an application can pass a datagram
of length 1-65,535 bytes to the AAL layer and have
it delivered to the destn, either on a guaranteed or
a best effort basis.
Upon arrival in the CS ,a msg is padded out and a
trailer added.
The amount of padding( 0-47 bytes) is chosen to
make the entire msg be a multiple of 48 bytes.
AAL 5 does not have a CS header , just an 8 byte
trailer.
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1
Bytes
Payload (1-65,535 bytes)
1
UU
AAL 5 Convergence Sublayer format
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2
Length
4
CRC
AAL5 CS format

UU- User to User



field is not used by the AAL itself.
It is available for a higher layer for its own purpose , like
sequencing or muxing.
Length- tells how long the true payload is-in bytes
(without padding).

Value =0 used to abort current message in midstream

CRC –is a std 32-bit checksum

The msg is txed by passing it to the SAR sublayer, which
does not add any headers or trailers.
Instead it breaks the msg into 48-byte units and passes
each of these to the ATM layer for txn.
The main advtg of AAL 5 over AAL 3/4 is the much greater
efficiency.


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
AAL 5 has a slightly large trailer/msg (8 bytes).

The lack of the seq. number is compensated for by the
longer checksum, which an detect lost, misinserted , or
missing cells without using seq. numbers.

Within the Internet Community , it is expected that normal
way of interfacing to ATM n/ws will be to transport IP
packets with the AAL 5 payload field.
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