Elements of Transport Protocols

1
Elements of Transport Protocols
2

• 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

3
Elements of Transport Protocols

• Addressing
• Connection Establishment
• Connection Release
• Flow Control and Buffering
• Multiplexing
• Crash Recovery

4
1.Addressing

• 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.

5
2.Establishing a connection

• 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.

6

• 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.

7

• 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.

8
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. (c) Duplicate CONNECTION
REQUEST and duplicate ACK.
9

• 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

10

• 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.

11
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.

12
Asymmetric Release

• Abrupt disconnection with loss of data.

13

• 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.

14
Symmetric Release

• 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

15
The Two-Army problem

• 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.

16

• The two-army problem.

17

• 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?

18

• 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.

19

• Now improve the protocol by making it a three-way
  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.

20

• 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.

21

• Four protocol scenarios for releasing a
  connection. (a) Normal case of a three-way
  handshake. (b) final ACK lost.

6-14, a, b
22

• (c) Response lost. (d) Response lost and
  subsequent DRs lost.

6-14, c,d
23

• 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

24
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.

25

• 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.

26

• 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
variable-sized buffers. (c) One large circular
buffer per connection.
28

• 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.

29
5.Multiplexing

• In the transport layer the need for multiplexing
  can arise in a number of ways.
• For Eg, if only one network address is available
  on a host, all transport connections on that
  machine have to use it.
• When a TPDU comes in, some way is needed to tell
  which process to give it to.
• This situation, called upward multiplexing, is
  shown in fig a.
• In this figure, 4 distinct transport connections
  all use the same network connection (e.g., IP
  address) to the remote host.
• 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.
• 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)

32

• The Internet has two main protocols in the
  transport layer,
• connectionless protocol
• connection-oriented protocol
• connectionless protocol is UDP
• connection-oriented protocol is TCP
• UDP is basically just IP with a short header added

33
Introduction to TCP

• 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
• 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.

34
The TCP Service Model

• TCP service is obtained by both the sender and
  receiver creating end points, called sockets.
• Each socket has a socket number (address)
  consisting of
• 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.

35

• 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.

36

• Some assigned ports.

Port
Protocol
Use
21
FTP
File transfer
23
Remote login
Telnet
E-mail
25
SMTP
69
Trivial File Transfer Protocol
TFTP
Finger
Lookup info about a user
79
80
World Wide Web
HTTP
POP-3
110
Remote e-mail access
USENET news
119
NNTP
37

• 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
• 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.

38

• (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.

39
The TCP Protocol

• 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 65,515-byte 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

40

• The basic protocol used by TCP entities is the
  sliding window protocol.
• When a sender transmits a segment, it also starts
  a timer.
• 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.
• If the sender's timer goes off before the
  acknowledgement is received, the sender transmits
  the segment again.

41
The TCP Segment Header

• Every segment begins with a fixed-format 20-byte
  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
• first 20 refer to the IP header and
• second to the TCP header.
• Segments without any data are legal and are
  commonly used for
• acknowledgements and
• control messages.

42

• TCP Header.

43

• 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 .

44

• 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 problem on your hands.

45

• The SYN bit is used to establish connections.
• The connection request has SYN 1 and ACK 0 to
  indicate that the piggyback acknowledgement field
  is not in use.
• The connection reply bears an acknowledgement it
  has SYN 1 and ACK 1.
• 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.

46

• 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.

47
ATM AAL LAYER PROTOCOLS
48

• 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
• 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.

49

• When ITU began defining AAL, it realized that
  different applications had different
  requirements, so it organized the service space
  along 3 axes
• 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

50
Original service classes supported by AAL (now
obsolete)
B
D
A
C
Real Time
Real Time
Real Time
Real Time
None
None
None
None
Timing
Bit Rate
Constant
Variable
Constant
Variable
Connection oriented
Connection less
Mode
51
Structure of the AAL
Convergence sublayer (service specific part)
AAL
Convergence sublayer (common part)
Segmentation Reassembly sublayer
ATM layer
Physical layer
52

• 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
• 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

53

• 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

54
AAL1

• Is the protocol used for transmiting class A
  traffic, that is
• 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 stop-and-wait are not used because the delays
  introduced by timeouts and retransmition are
  unacceptable.
• Missing cells are reported to the application

55

• AAL1 uses a Convergence Sublayer and an SAR
  sublayer.
• The Convergence Sublayer (CS)
• 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.

56
The AAL 1 CELL FORMAT
0
SN
SNP
47-byte Payload
Non-P
Even Parity
1
SN
SNP
46-byte Payload
Parity
P
48 bytes
SAR header
57
AAL1 SAR

• It has 1 byte headercontaining a
• 3 bit cell seq number SN
• to detect missing or misinserted cells
• 3 bit Sequence Number Protection SNP (like check
  sum)
• Allows correction of single errors detection of
  double errors in seq no. field
• One bit for parity even bit parity
• P cells used when message boundaries must be
  preserved
• Pointer field -1 byte
• Used to give offset of start of next message
• Higher order bit is reserved for future use

58
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 problem-AAL 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.

59

• For these reason another protocol AAL2 is used.
• As in AAL 1 , the CS sublayer does not have a
  protocol but the SAR sublayer does.
• It has a 1-byte header and a 2-byte trailer-45
  byte data also.

SN
IT
45-Byte Payload
LI
CRC

• The SN (sequence Number) is used for numbering
  cells in order to detect missing or misinserted
  cells.
• The IT field (InformationType) is used to
  indicate that the cell is the start, middle or
  end of a msg.
• The LI( Length Indicator) field tells how big the
  payload is, in bytes(lt45 bytes).
• The CRC is a checksum over the entire cell-so
  errors can be detected.
• field sizes not included in std

60
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 VC and be separated
  at the destn.

61
AAL 3/4 CS msg format
Bits
CS header
CS Trailer

• 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.
• These 2 bytes must be same incremented by 1 on
  every new msg sent
• The BA size is used for buffer allocation.
• Tells receiver how much buffer space to be
  allocated for message in advance of its arrival
• The Length field gives the payload length.
• In msg mode length must be equal to BA size..
• Trailer has 1 unused byte

62
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
4
10
6
10
SN
ST
MID
44-byte payload
LI
CRC
00 middle 01 End 10 beginning 11 single cell msg
1-44
63

• ST (Segment Type)- for msg framing
• 00 middle (continuation of message COM)
• 01 end of message (EOM)
• 10 beginning of message (BOM)
• 11 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

64
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
1
2
4
Payload (1-65,535 bytes)
UU
Length
CRC
AAL 5 Convergence Sublayer format
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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.