Transport Layer Protocol (continued)-part-2

Transport Layer Protocol (continued)

TCP connection is a duplex connection. That means there is no difference between two sides once the connection is established.
Salient Features of TCP

• Piggybacking of acknowledments:The ACK for the last received packet need not be sent as a new packet, but gets a free ride on the next outgoing data frame(using the ACK field in the frame header). The technique is temporarily delaying outgoing ACKs so that they can be hooked on the next outgoing data frame is known as piggybacking. But ACK can't be delayed for a long time if receiver(of the packet to be acknowledged) does not have any data to send.
• Flow and congestion control:TCP takes care of flow control by ensuring that both ends have enough resources and both can handle the speed of data transfer of each other so that none of them gets overloaded with data. The term congestion control is used in almost the same context except that resources and speed of each router is also taken care of. The main concern is network resources in the latter case.
• Multiplexing / Demultiplexing: Many application can be sending/receiving data at the same time. Data from all of them has to be multiplexed together. On receiving some data from lower layer, TCP has to decide which application is the recipient. This is called demultiplexing. TCP uses the concept of port number to do this.

TCP segment header:

Explanation of header fields:

• Source and destination port :These fields identify the local endpoint of the connection. Each host may decide for itself how to allocate its own ports starting at 1024. The source and destination socket numbers together identify the connection.
• Sequence and ACK number : This field is used to give a sequence number to each and every byte transferred. This has an advantage over giving the sequence numbers to every packet because data of many small packets can be combined into one at the time of retransmission, if needed. The ACK signifies the next byte expected from the source and not the last byte received. The ACKs are cumulative instead of selective.Sequence number space is as large as 32-bit although 17 bits would have been enough if the packets were delivered in order. If packets reach in order, then according to the following formula:
  (sender's window size) + (receiver's window size) < (sequence number space)
  the sequence number space should be 17-bits. But packets may take different routes and reach out of order. So, we need a larger sequence number space. And for optimisation, this is 32-bits.
• Header length :This field tells how many 32-bit words are contained in the TCP header. This is needed because the options field is of variable length.
• Flags : There are six one-bit flags.
  1. URG : This bit indicates whether the urgent pointer field in this packet is being used.
  2. ACK :This bit is set to indicate the ACK number field in this packet is valid.
  3. PSH : This bit indicates PUSHed data. The receiver is requested to deliver the data to the application upon arrival and not buffer it until a full buffer has been received.
  4. RST : This flag is 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. This causes an abrupt end to the connection, if it existed.
  5. SYN : This bit is used to establish connections. The connection request(1st packet in 3-way handshake) has SYN=1 and ACK=0. The connection reply (2nd packet in 3-way handshake) has SYN=1 and ACK=1.
  6. FIN : This bit is used to release a connection. It specifies that the sender has no more fresh data to transmit. However, it will retransmit any lost or delayed packet. Also, it will continue to receive data from other side. Since SYN and FIN packets have to be acknowledged, they must have a sequence number even if they do not contain any data.
• Window Size : Flow control in TCP is handled using a variable-size sliding window. The Window Size field tells how many bytes may be sent starting at the byte acknowledged. Sender can send the bytes with sequence number between (ACK#) to (ACK# + window size - 1) A window size of zero is legal and says that the bytes up to and including ACK# -1 have been received, but the receiver would like no more data for the moment. Permission to send can be granted later by sending a segment with the same ACK number and a nonzero Window Size field.
• Checksum : This is provided for extreme reliability. It checksums the header, the data, and the conceptual pseudoheader. The pseudoheader contains the 32-bit IP address of the source and destination machines, the protocol number for TCP(6), and the byte count for the TCP segment (including the header).Including the pseudoheader in TCP checksum computation helps detect misdelivered packets, but doing so violates the protocol hierarchy since the IP addresses in it belong to the IP layer, not the TCP layer.
• Urgent Pointer : Indicates a byte offset from the current sequence number at which urgent data are to be found. Urgent data continues till the end of the segment. This is not used in practice. The same effect can be had by using two TCP connections, one for transferring urgent data.
• Options : Provides a way to add extra facilities not covered by the regular header. eg,
  • Maximum TCP payload that sender is willing to handle. The maximum size of segment is called MSS (Maximum Segment Size). At the time of handshake, both parties inform each other about their capacity. Minimum of the two is honoured. This information is sent in the options of the SYN packets of the three way handshake.
  • Window scale option can be used to increase the window size. It can be specified by telling the receiver that the window size should be interpreted by shifting it left by specified number of bits. This header option allows window size up to 230.
• Data : This can be of variable size. TCP knows its size by looking at the IP size header.

Topics to be Discussed relating TCP

1. Maximum Segment Size : It refers to the maximum size of segment ( MSS ) that is acceptable to both ends of the connection. TCP negotiates for MSS using OPTION field. In Internet environment MSS is to be selected optimally. An arbitrarily small segment size will result in poor bandwith utilization since Data to Overhead ratio remains low. On the other hand extremely large segment size will necessitate large IP Datagrams which require fragmentation. As there are finite chances of a fragment getting lost, segment size above "fragmentation threshold " decrease the Throughput. Theoretically an optimum segment size is the size that results in largest IP Datagram, which do not require fragmentation anywhere enroute from source to destination. However it is very difficult to find such an optimum segmet size. In system V a simple technique is used to identify MSS. If H1 and H2 are on the same network use MSS=1024. If on different networks then MSS=5000.
2. Flow Control : TCP uses Sliding Window mechanism at octet level. The window size can be variable over time. This is achieved by utilizing the concept of "Window Advertisement" based on :
   1. Buffer availabilty at the receiver
   2. Network conditions ( traffic load etc.)
   In the former case receiver varies its window size depending upon the space available in its buffers. The window is referred as RECEIVE WINDOW (Recv_Win). When receiver buffer begin to fill it advertises a small Recv_Win so that the sender does'nt send more data than it can accept. If all buffers are full receiver sends a "Zero" size advertisement. It stops all transmission. When buffers become available receiver advertises a Non Zero widow to resume retransmission. The sender also periodically probes the "Zero" window to avoid any deadlock if the Non Zero Window advertisement from receiver is lost. The Variable size Recv_Win provides efficient end to end flow control.
   The second case arises when some intermediate node ( e.g. a router ) controls the source to reduce transmission rate. Here another window referred as COGESTION WINDOW (C_Win) is utilized. Advertisement of C_Win helps to check and avoid congestion.
3. Congestion Control : Congestion is a condition of severe delay caused by an overload of datagrams at any intermediate node on the Internet. If unchecked it may feed on itself and finally the node may start dropping arriving datagrams.This can further aggravate congestion in the network resulting in congestion collapse. TCP uses two techniques to check congestion.
   1. Slow Start : At the time of start of a connection no information about network conditios is available. A Recv_Win size can be agreed upon however C_Win size is not known. Any arbitrary C_Win size can not be used because it may lead to congestion. TCP acts as if the window size is equal to the minimum of ( Recv_Win & C_Win). So following algorithm is used.
      1. Recv_Win=X
      2. SET C_Win=1
      3. for every ACK received C_Win++
   2. Multiplicative decrease : This scheme is used when congestion is encountered ( ie. when a segment is lost ). It works as follows. Reduce the congestion window by half if a segment is lost and exponentially backoff the timer ( double it ) for the segments within the reduced window. If the next segment also gets lost continue the above process. For successive losses this scheme reduces traffic into the connection exponentially thus allowing the intermediate nodes to clear their queues. Once congestion ends SLOW START is used to scale up the transmission.
4. Congestion Avoidance : This procedure is used at the onset of congestion to minimize its effect on the network. When transmission is to be scaled up it should be done in such a way that it does'nt lead to congestion again. Following algorithm is used .
   1. At loss of a segment SET C_Win=1
   2. SET SLOW START THRESHOLD (SST) = Send_Win / 2
   3. Send segment
   4. If ACK Received, C_Win++ till C_Win <= SST
   5. else for each ACK C_Win += 1 / C_Win
5. Time out and Retransmission : Following two schemes are used :
   1. Fast Retransmit
   2. Fast Recovery
   When a source sends a segment TCP sets a timer. If this value is set too low it will result in many unnecessary treransmissions. If set too high it results in wastage of banwidth and hence lower throughput. In Fast Retransmit scheme the timer value is set fairly higher than the RTT. The sender can therefore detect segment loss before the timer expires. This scheme presumes that the sender will get repeated ACK for a lost packet.
6. Round Trip Time (RTT) : In Internet environment the segments may travel across different intermediate networks and through multiple routers. The networks and routers may have different delays, which may vary over time. The RTT therefore is also variable. It makes difficult to set timers. TCP allows varying timers by using an adaptive retransmission algorithm. It works as follows.
   1. Note the time (t1) when a segment is sent and the time (t2) when its ACK is received.
   2. Compute RTT(sample) = (t 2 - t 1 )
   3. Again Compute RTT(new) for next segment.
   4. Compute Average RTT by weighted average of old and new values of RTT
   5. RTT(est) = a *RTT(old) + (1-a) * RTT (new) where 0 < a < 1
      A high value of 'a' makes the estimated RTT insensitive to changes that last for a short time and RTT relies on the history of the network. A low value makes it sensitive to current state of the network. A typical value of 'a' is 0.75
   6. Compute Time Out = b * RTT(est) where b> 1
      A low value of 'b' will ensure quick detection of a packet loss. Any small delay will however cause unnecessary retransmission. A typical value of 'b' is kept at .2

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Image References

• http://plato.acadiau.ca/courses/comp/Eberbach/comp4343/lectures/transport/Com-TCP/f20_6.gif

Transport Layer Protocol- Implementation Issues

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Transport Layer Protocol- Implementation Issues

In this class we discussed about the TCP from the implementation point of view and addressed various issues like state diagram and other details which TCP Standard does not define but supported by commercial implementations.

State Diagram

The state diagram approach to view the TCP connection establishment and closing simplifies the design of TCP implementation. The idea is to represent the TCP connection state, which progresses from one state to other as various messages are exchanged. To simplify the matter, we considered two state diagrams, viz., for TCP connection establishment and TCP connection closing.
Fig 1 shows the state diagram for the TCP connection establishment and associated table briefly explains each state. TCP Connection establishment The table gives brief description of each state of the above diagram.

State Description Table 1.

Listen	Represents the state when waiting for connection request from any remote host and port. This specifically applies to a Server.  From this state, the server can close the service or actively open a connection by sending SYN.
Syn-Sent	Represents waiting for a matching for a connection request after having sent a connection request. This applies to both server and client side. Even though server is considered as the one with passive open, it can also send a SYN packet actively.
Syn_Rcvd	Represents waiting for a confirmation connection request acknowledgment after having both received and sent connection request.
Estab	Represents an open connection. Data transfer can take place from this point onwards.

  After the connection has been established, two end-points will exchange useful information and terminate the connection. Fig. 2 shows the state diagram for terminating an active connection.
   Fig 2.  TCP Connection termination  
 
 

State Description Table 2

FIN-WAIT-1	Represents connection termination request from the remote TCP peer, or an acknowledgment of the connection termination request previously  sent.  This state is entered when server issues close call.
FIN-WAIT-2	Represents waiting for a connection termination request from the remote TCP.
CLOSING	Represents connection termination request acknowledgment from the remote TCP.
TIME_WAIT	This represents waiting time enough for the packets to reach their destination. This waiting time is usually 4 min.
CLOSE_WAIT	Represents a state when the server receives a FIN from the remote TCP , sends ACK and issues close call sending FIN
LAST_ACK	Represents waiting for an ACK for the previously sent FIN-ACK to the remote TCP
CLOSE	Represents a closed TCP connection having received all the ACKs

 

Other implementation details

Quite Time

It might happen that a host currently in communication crashes and reboots. At startup time, all the data structures and timers will be reset to an initial value. To make sure that earlier connection packets are gracefully rejected, the local host is not allowed to make any new connection for a small period at startup. This time will be set in accordance with reboot time of the operating system.

Initial Sequence number :

Initial sequence number used in the TCP communication will be  initialized at boot time randomly, rather than to 0. This is to ensure that packets from old connection should not interfere with a new connection. So the recommended method is to

• Initialize the ISN at boot time by a random number
• For every 500 ms, increment ISN by 64K
• With every SYN received, increment ISN by 64K

Maximum Request backlog at server

As we have seen in Unix Networking programming, listen(sd,n),  sets a maximum to the number of requests to be obliged by the server at any time. So if there are already n requests for connection, and n+1 request comes, two things can be done.

• Drop the packet silently
• Ask the peer to send the request later.

The first option is recommended here because, the assumption is that this queue for request is a coincident and some time later, the server should be free to process the new request. Hence if we drop the packet, the client will go through the time-out and retransmission and server will be free to process it. Also, Standard TCP does not define any strategy/option of knowing who requested the connection. Only Solaris 2.2 supports this option.

Delayed Acknowledgment

TCP will piggyback the acknowledgment with its data. But if the peer does not have the any data to send at that moment, the acknowledgment should not be delayed too long. Hence a timer for 200 ms will be used. At every 200 ms, TCP will check for any acknowledgment to be sent and send them as individual packets.

Small packets

TCP implementation discourages small packets. Especially if a previous relatively large packet has been sent and no acknowledgment has been received so far, then this small packet will be stored in the buffer until the situation improves. But there are some applications for which delayed data is worse than bad data. For example, in telnet, each key stroke will be processed  by the server and hence no delay should be introduced. As we have seen in Unix Networking programming, options for the socket can be set as NO_DELAY, so that small packets are not discouraged.

ICMP Source Quench

We have seen in ICMP that ICMP Source Quench message will be send for the peer to slow down. Some implementations discard this message, but few set the current window size to 1.
But this is not a very good idea.

Retransmission Timeout

In some implementation (E.g.. Linux), RTO = RTT + 4 * delay variance is used to instead of constant 2. Also instead of calculating RTT(est) from the scratch, cache will be used to store the history from which new values are calculated as discussed in the previous classes.
Standard values for Maximum Segment Life (MSL)  will be between 0.5 to 2 minutes and Time wait state = f(MSL)

Keep Alive Time

Another important timer in TCP is keep alive timer. It is basically used by a TCP peer  to check whether the other end is up or down. It periodically checks this connection. If  the other end did not respond, then that connection will be closed.

Persist Timer

As we saw in TCP window management, when source sends one full window of packets, it will  set its window size to 0 and expects an ACK from remote TCP  to increase its window size. Suppose such an ACK has been sent and is lost. Hence source will have current window size = 0 and cannot send & destination is expecting next byte. To avoid such a deadlock, a Persist Timer will be used. When this timer goes off, the source will send the last one byte again. So we hope that situation has improved and an ACK to increase the current window size will be received.

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References

• http://www.ssfnet.org/Exchange/tcp/tcpTutorialNotes.html

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IEEE 802.3 and Ethernet-Notes

IEEE 802.3 and Ethernet

• Very popular LAN standard.
• Ethernet and IEEE 802.3 are distinct standards but as they are very similar to one another these words are used interchangeably.
• A standard for a 1-persistent CSMA/CD LAN.
• It covers the physical layer and MAC sublayer protocol.

Ethernet Physical Layer

A Comparison of Various Ethernet and IEEE 802.3 Physical-Layer Specifications

Characteristic	Ethernet Value	IEEE 802.3 Values
		10Base5	10Base2	10BaseT	10BaseF	10 Base -TX	100BaseT4
Data rate (Mbps)	10	10	10	10	10	100	100
Signaling method	Baseband	Baseband	Baseband	Baseband	Baseband	Baseband	Baseband
Maximum segment length (m)	500	500	185	100	2,000	100	100
Media	50-ohm coax (thick)	50-ohm coax (thick)	50-ohm coax (thin)	Unshielded twisted-pair cable	Fiber-optic	Cat 5 UTP	Unshielded twisted-pair cable
Nodes/segment	100	100	30	1024	1024
Topology	Bus	Bus	Bus	Star	Point-to-point	Bus	Bus

10Base5 means it operates at 10 Mbps, uses baseband signaling and can support segments of up to 500 meters. The 10Base5 cabling is popularly called the Thick Ethernet. Vampire taps are used for their connections where a pin is carefully forced halfway into the co-axial cable's core as shown in the figure below. The 10Base2 or Thin Ethernet bends easily and is connected using standard BNC connectors to form T junctions (shown in the figure below). In the 10Base-T scheme a different kind of wiring pattern is followed in which all stations have a twisted-pair cable running to a central hub (see below). The difference between the different physical connections is shown below:

(a) 10Base5 (b)10Base2 (c)10Base-T
All 802.3 baseband systems use Manchester encoding , which is a way for receivers to unambiguously determine the start, end or middle of each bit without reference to an external clock. There is a restriction on the minimum node spacing (segment length between two nodes) in 10Base5 and 10Base2 and that is 2.5 meter and 0.5 meter respectively. The reason is that if two nodes are closer than the specified limit then there will be very high current which may cause trouble in detection of signal at the receiver end. Connections from station to cable of 10Base5 (i.e. Thick Ethernet) are generally made using vampire taps and to 10Base2 (i.e. Thin Ethernet) are made using industry standard BNC connectors to form T junctions. To allow larger networks, multiple segments can be connected by repeaters as shown. A repeater is a physical layer device. It receives, amplifies and retransmits signals in either direction. Note: To connect multiple segments, amplifier is not used because amplifier also amplifies the noise in the signal, whereas repeater regenerates signal after removing the noise.

IEEE 802.3 Frame Structure

Preamble (7 bytes)	Start of Frame Delimiter (1 byte)	Dest. Address (2/6 bytes)	Source Address (2/6 bytes)	Length (2 bytes)	802.2 Header+Data (46-1500 bytes)	Frame Checksum (4 bytes)

A brief description of each of the fields

• Preamble :Each frame starts with a preamble of 7 bytes, each byte containing the bit pattern 10101010. Manchester encoding is employed here and this enables the receiver's clock to synchronize with the sender's and initialise itself.
• Start of Frame Delimiter :This field containing a byte sequence 10101011 denotes the start of the frame itself.
• Dest. Address :The standard allows 2-byte and 6-byte addresses. Note that the 2-byte addresses are always local addresses while the 6-byte ones can be local or global.
  
  2-Byte Address    - Manually assigned address

  Individual(0)/Group(1) (1 bit)	Address of the machine (15 bits)

  
  6-Byte Address    - Every Ethernet card with globally unique address

  Individual(0)/Group(1) (1 bit)	Universal(0)/Local(1) (1 bit)	Address of the machine (46 bits)

  
  Multicast : Sending to group of stations. This is ensured by setting the first bit in either 2-byte/6-byte addresses to 1.
  Broadcast : Sending to all stations. This can be done by setting all bits in the address field to 1.All Ethernet cards(Nodes) are a member of this group.
• Source Address :Refer to Dest. Address. Same holds true over here.
• Length : The Length field tells how many bytes are present in the data field, from a minimum of 0 to a maximum of 1500. The Data and padding together can be from 46bytes to 1500 bytes as the valid frames must be at least 64 bytes long, thus if data is less than 46 bytes the amount of padding can be found out by length field.
• Data :Actually this field can be split up into two parts - Data(0-1500 bytes) and Padding(0-46 bytes).
  Reasons for having a minimum length frame :
  1. To prevent a station from completing the transmission of a short frame before the first bit has even reached the far end of the cable, where it may collide with another frame. Note that the transmission time ought to be greater than twice the propagation time between two farthest nodes. transmission time for frame > 2*propagation time between two farthest nodes
  2. When a transceiver detects a collision, it truncates the current frame, which implies that stray bits and pieces of frames appear on the cable all the time. Hence to distinguish between valid frames from garbage, 802.3 states that the minimum length of valid frames ought to be 64 bytes (from Dest. Address to Frame Checksum).
• Frame Checksum : It is a 32-bit hash code of the data. If some bits are erroneously received by the destination (due to noise on the cable), the checksum computed by the destination wouldn't match with the checksum sent and therefore the error will be detected. The checksum algorithm is a cyclic redundancy checksum (CRC) kind. The checksum includes the packet from Dest. Address to Data field.

Ethernet Frame Structure

Preamble (8 bytes)	Dest. Address (2/6 bytes)	Source Address (2/6 bytes)	Type (2 bytes)	Data (46-1500 bytes)	Frame Checksum (4 bytes)

A brief description of the fields which differ from IEEE 802.3

• Preamble :The Preamble and Start of Frame Delimiter are merged into one in Ethernet standard. However, the contents of the first 8 bytes remains the same in both.
• Type :The length field of IEEE 802.3 is replaced by Type field, which denotes the type of packet being sent viz. IP, ARP, RARP, etc. If the field indicates a value less than 1500 bytes then it is length field of 802.3 else it is the type field of Ethernet packet.

Truncated Binary Exponential Back off

In case of collision the node transmitting backs off by a random number of slots , each slot time being equal to transmission time of 512 bits (64 Byte- minimum size of a packet) in the following fashion: No of Collision                            Random No of slots
1st                                                0-1
2nd                                               0-3
3rd                                                0-7
|                                                        |
|                                                        |
10th                                                0-1023
---------------------------------------------
11th                                                0-1023
12th                                                0-1023
|                                                        |
16th                                                0-1023
In general after i collisions a random number between 0-2^i-1 is chosen , and that number of slots is skipped. However, after 10 collisions have been reached the randomization interval is frozen at maximum of 1023 slots. After 16 collisions the controller reports failure back to the computer.

5-4-3 Rule

Each version of 802.3 has a maximum cable length per segment because long propagation time leads to difficulty in collision detection. To compensate for this the transmission time has to be increased which can be achieved by slowing down the transmission rate or increasing the packet size, neither of which is desirable. Hence to allow for large networks, multiple cables are connected via repeaters. Between any two nodes on an Ethernet network, there can be at most five segments, four repeaters and three populated segments (non-populated segments are those which do not have any machine connected between the two repeaters). This is known as the 5-4-3 Rule.

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Image References:

• http://homepages.ius.edu/rwisman/b438/Html/4-14.jpg

CSMA with Collision Avoidance-Notes

CSMA with Collision Avoidance

We have observed that CSMA/CD would break down in wireless networks because of hidden node and exposed nodes problems. We will have a quick recap of these two problems through examples.

Hidden Node Problem

In the case of wireless network it is possible that A is sending a message to B, but C is out of its range and hence while "listening" on the network it will find the network to be free and might try to send packets to B at the same time as A. So, there will be a collision at B. The problem can be looked upon as if A and C are hidden from each other. Hence it is called the "hidden node problem".

Exposed Node Problem

If C is transmitting a message to D and B wants to transmit a message to A, B will find the network to be busy as B hears C trnasmitting. Even if B would have transmitted to A, it would not have been a problem at A or D. CSMA/CD would not allow it to transmit message to A, while the two transmissions could have gone in parallel.

Addressing hidden node problem (CSMA/CA)

Consider the figure above.Suppose A wants to send a packet to B. Then it will first send a small packet to B called "Request to Send" (RTS). In response, B sends a small packet to A called "Clear to Send" (CTS). Only after A receives a CTS, it transmits the actual data. Now, any of the nodes which can hear either CTS or RTS assume the network to be busy. Hence even if some other node which is out of range of both A and B sends an RTS to C (which can hear at least one of the RTS or CTS between A and B), C would not send a CTS to it and hence the communication would not be established between C and D. One issue that needs to be addressed is how long the rest of the nodes should wait before they can transmit data over the network. The answer is that the RTS and CTS would carry some information about the size of the data that B intends to transfer. So, they can calculate time that would be required for the transmission to be over and assume the network to be free after that.Another interesting issue is what a node should do if it hears RTS but not a corresponding CTS. One possibility is that it assumes the recipient node has not responded and hence no transmission is going on, but there is a catch in this. It is possible that the node hearing RTS is just on the boundary of the node sending CTS. Hence, it does hear CTS but the signal is so deteriorated that it fails to recognize it as a CTS. Hence to be on the safer side, a node will not start transmission if it hears either of an RTS or a CTS.
The assumption made in this whole discussion is that if a node X can send packets to a node Y, it can also receive a packet from Y, which is a fair enough assumption given the fact that we are talking of a local network where standard instruments would be used. If that is not the case additional complexities would get introduced in the system.

Does CSMA/CD work universally in the wired networks ?

The problem of range is there in wired networks as well in the form of deterioration of signals. Normally to counter this, we use repeaters, which can regenerate the original signal from a deteriorated one. But does that mean that we can build as long networks as we want with repeaters. The answer, unfortunately, is NO! The reason is the beyond a certain length CSMA/CD will break down. The mechanism of collision detection which CSMA/CD follows is through listening while talking. What this means is so long as a node is transmitting the packet, it is listening on the cable. If the data it listens to is different from the data it is transmitting it assumes a collision. Once it has stopped transmitting the packet, and has not detected collision while transmission was going on, it assumes that the transmission was successful. The problem arises when the distance between the two nodes is too large. Suppose A wants to transmit some packet to B which is at a very large distance from B. Data can travel on cable only at a finite speed (usually 2/3c, c being the speed of light). So, it is possible that the packet has been transmitted by A onto the cable but the first bit of the packet has not yet reached B. In that case, if a collision occurs, A would be unaware of it occurring. Therefore there is problem in too long a network.
Let us try to parametrize the above problem. Suppose "t" is the time taken for the node A to transmit the packet on the cable and "T" is the time , the packet takes to reach from A to B. Suppose transmission at A starts at time t0. In the worst case the collision takes place just when the first packet is to reach B. Say it is at t0+T-e (e being very small). Then the collision information will take T-e time to propagate back to A. So, at t0+2(T-e) A should still be transmitting. Hence, for the correct detection of collision (ignoring e)

t > 2T

t increases with the number of bits to be transferred and decreases with the rate of transfer (bits per second). T increases with the distance between the nodes and decreases with the speed of the signal (usually 2/3c). We need to either keep t large enough or T as small. We do not want to live with lower rate of bit transfer and hence slow networks. We can not do anything about the speed of the signal. So what we can rely on is the minimum size of the packet and the distance between the two nodes. Therefore, we fix some minimum size of the packet and if the size is smaller than that, we put in some extra bits to make it reach the minimum size. Accordingly we fix the maximum distance between the nodes. Here too, there is a tradeoff to be made. We do not want the minimum size of the packets to be too large since that wastes lots of resources on cable. At the same time we do not want the distance between the nodes to be too small. Typical minimum packet size is 64 bytes and the corresponding distance is 2-5 kilometers.

Collision Free Protocols

Although collisions do not occur with CSMA/CD once a station has unambigously seized the channel, they can still occur during the contention period. These collisions adversely affect the efficiency of transmission. Hence some protocols have been developed which are contention free.

Bit-Map Method

In this method, there N slots. If node 0 has a frame to send, it transmit a 1 bit during the first slot. No other node is allowed to transmit during this period. Next node 1 gets a chance to transmit 1 bit if it has something to send, regardless of what node 0 had transmitted. This is done for all the nodes. In general node j may declare the fact that it has a frsme to send by inserting a 1 into slot j. Hence after all nodes have passed, each node has complete knowledge of who wants to send a frame. Now they begin transmitting in numerical order. Since everyone knows who is transmitting and when, there could never be any collision. The basic problem with this protocol is its inefficiency during low load. If a node has to transmit and no other node needs to do so, even then it has to wait for the bitmap to finish. Hence the bitmap will be repeated over and over again if very few nodes want to send wasting valuable bandwidth.

Binary Countdown

In this protocol, a node which wants to signal that it has a frame to send does so by writing its address into the header as a binary number. The arbitration is such that as soon as a node sees that a higher bit position that is 0 in its address has been overwritten with a 1, it gives up. The final result is the address of the node which is allowed to send. After the node has transmitted the whole process is repeated all over again. Given below is an example situation.

Nodes	Addresses
A	0010
B	0101
C	1010
D	1001
	----
	1010

Node C having higher priority gets to transmit. The problem with this protocol is that the nodes with higher address always wins. Hence this creates a priority which is highly unfair and hence undesirable.

Limited Contention Protocols

Both the type of protocols described above - Contention based and Contention - free has their own problems. Under conditions of light load, contention is preferable due to its low delay. As the load increases, contention becomes increasingly less attractive, because the overload associated with channel arbitration becomes greater. Just the reverse is true for contention - free protocols. At low load, they have high delay, but as the load increases , the channel efficiency improves rather than getting worse as it does for contention protocols. Obviously it would be better if one could combine the best properties of the contention and contention - free protocols, that is, protocol which used contention at low loads to provide low delay, but used a cotention-free technique at high load to provide good channel efficiency. Such protocols do exist and are called Limited contention protocols.
It is obvious that the probablity of some station aquiring the channel could only be increased by decreasing the amount of competition. The limited contention protocols do exactly that. They first divide the stations up into ( not necessarily disjoint ) groups. Only the members of group 0 are permitted to compete for slot 0. The competition for aquiring the slot within a group is contention based. If one of the members of that group succeeds, it aquires the channel and transmits a frame. If there is collision or no node of a particular group wants to send then the members of the next group compete for the next slot. The probablity of a particular node is set to a particular value ( optimum ).

Adaptive Tree Walk Protocol

The following is the method of adaptive tree protocol. Initially all the nodes are allowed to try to aquire the channel. If it is able to aquire the channel, it sends its frame. If there is collision then the nodes are divided into two equal groups and only one of these groups compete for slot 1. If one of its member aquires the channel then the next slot is reserved for the other group. On the other hand, if there is a collision then that group is again subdivided and the same process is followed. This can be better understood if the nodes are thought of as being organised in a binary tree as shown in the following figure.

Many improvements could be made to the algorithm. For example, consider the case of nodes G and H being the only ones wanting to transmit. At slot 1 a collision will be detected and so 2 will be tried and it will be found to be idle. Hence it is pointless to probe 3 and one should directly go to 6,7.

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Image References:

• http://www.ccs.neu.edu/course/csg150/Solutions-III_files/image002.gif

Multiplexing,Network Topologies,Aloha Protocols,Carrier Sense Mutiple Access Protocols(CSMA)-NOTES

Multiplexing

When two communicating nodes are connected through a media, it generally happens that bandwidth of media is several times greater than that of the communicating nodes. Transfer of a single signal at a time is both slow and expensive. The whole capacity of the link is not being utilized in this case. This link can be further exploited by sending several signals combined into one. This combining of signals into one is called multiplexing.

1. Frequency Division Multiplexing (FDM): This is possible in the case where transmission media has a bandwidth than the required bandwidth of signals to be transmitted. A number of signals can be transmitted at the same time. Each source is allotted a frequency range in which it can transfer it's signals, and a suitable frequency gap is given between two adjescent signals to avoid overlapping. This is type of multiplexing is commonly seen in the cable TV networks.
   
   
2. Time Division Multiplexing (TDM): This is possible when data transmission rate of the media is much higher than that of the data rate of the source. Multiple signals can be transmitted if each signal is allowed to be transmitted for a definite amount of time. These time slots are so small that all transmissions appear to be in parallel.
   1. Synchronous TDM: Time slots are preassigned and are fixed. Each source is given it's time slot at every turn due to it. This turn may be once per cycle, or several turns per cycle ,if it has a high data transfer rate, or may be once in a no. of cycles if it is slow. This slot is given even if the source is not ready with data. So this slot is transmitted empty.
      
      
   2. Asynchronous TDM: In this method, slots are not fixed. They are allotted dynamically depending on speed of sources, and whether they are ready for transmission.
      
      

Network Topologies

A network topology is the basic design of a computer network. It is very much like a map of a road. It details how key network components such as nodes and links are interconnected. A network's topology is comparable to the blueprints of a new home in which components such as the electrical system, heating and air conditioning system, and plumbing are integrated into the overall design. Taken from the Greek work "Topos" meaning "Place," Topology, in relation to networking, describes the configuration of the network; including the location of the workstations and wiring connections. Basically it provides a definition of the components of a Local Area Network (LAN). A topology, which is a pattern of interconnections among nodes, influences a network's cost and performance. There are three primary types of network topologies which refer to the physical and logical layout of the Network cabling. They are:

1. Star Topology: All devices connected with a Star setup communicate through a central Hub by cable segments. Signals are transmitted and received through the Hub. It is the simplest and the oldest and all the telephone switches are based on this. In a star topology, each network device has a home run of cabling back to a network hub, giving each device a separate connection to the network. So, there can be multiple connections in parallel.
   
   Advantages
   
   • Network administration and error detection is easier because problem is isolated to central node
   • Networks runs even if one host fails
   • Expansion becomes easier and scalability of the network increases
   • More suited for larger networks
   Disadvantages
   
   • Broadcasting and multicasting is not easy because some extra functionality needs to be provided to the central hub
   • If the central node fails, the whole network goes down; thus making the switch some kind of a bottleneck
   • Installation costs are high because each node needs to be connected to the central switch
2. Bus Topology: The simplest and one of the most common of all topologies, Bus consists of a single cable, called a Backbone, that connects all workstations on the network using a single line. All transmissions must pass through each of the connected devices to complete the desired request. Each workstation has its own individual signal that identifies it and allows for the requested data to be returned to the correct originator. In the Bus Network, messages are sent in both directions from a single point and are read by the node (computer or peripheral on the network) identified by the code with the message. Most Local Area Networks (LANs) are Bus Networks because the network will continue to function even if one computer is down. This topology works equally well for either peer to peer or client server.
   
   The purpose of the terminators at either end of the network is to stop the signal being reflected back. Advantages
   
   • Broadcasting and multicasting is much simpler
   • Network is redundant in the sense that failure of one node doesn't effect the network. The other part may still function properly
   • Least expensive since less amount of cabling is required and no network switches are required
   • Good for smaller networks not requiring higher speeds
   Disadvantages
   
   • Trouble shooting and error detection becomes a problem because, logically, all nodes are equal
   • Less secure because sniffing is easier
   • Limited in size and speed
3. Ring Topology: All the nodes in a Ring Network are connected in a closed circle of cable. Messages that are transmitted travel around the ring until they reach the computer that they are addressed to, the signal being refreshed by each node. In a ring topology, the network signal is passed through each network card of each device and passed on to the next device. Each device processes and retransmits the signal, so it is capable of supporting many devices in a somewhat slow but very orderly fashion. There is a very nice feature that everybody gets a chance to send a packet and it is guaranteed that every node gets to send a packet in a finite amount of time.
   Advantages
   
   • Broadcasting and multicasting is simple since you just need to send out one message
   • Less expensive since less cable footage is required
   • It is guaranteed that each host will be able to transmit within a finite time interval
   • Very orderly network where every device has access to the token and the opportunity to transmit
   • Performs better than a star network under heavy network load
   Disadvantages
   
   • Failure of one node brings the whole network down
   • Error detection and network administration becomes difficult
   • Moves, adds and changes of devices can effect the network
   • It is slower than star topology under normal load

Generally, a BUS architecture is preferred over the other topologies - ofcourse, this is a very subjective opinion and the final design depends on the requirements of the network more than anything else. Lately, most networks are shifting towards the STAR topology. Ideally we would like to design networks, which physically resemble the STAR topology, but behave like BUS or RING topology.

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Data Link Layer

Data link layer can be characterized by two types of layers:

1. Medium Access Layer (MAL)
2. Logical Link Layer

Aloha Protocols

History

The Aloha protocol was designed as part of a project at the University of Hawaii. It provided data transmission between computers on several of the Hawaiian Islands using radio transmissions.

• Communications was typically between remote stations and a central sited named Menehune or vice versa.
• All message to the Menehune were sent using the same frequency.
• When it received a message intact, the Menehune would broadcast an ack on a distinct outgoing frequency.
• The outgoing frequency was also used for messages from the central site to remote computers.
• All stations listened for message on this second frequency.

Pure Aloha

Pure Aloha is an unslotted, fully-decentralized protocol. It is extremely simple and trivial to implement. The ground rule is - "when you want to talk, just talk!". So, a node which wants to transmits, will go ahead and send the packet on its broadcast channel, with no consideration whatsoever as to anybody else is transmitting or not.

One serious drawback here is that, you dont know whether what you are sending has been received properly or not (so as to say, "whether you've been heard and understood?"). To resolve this, in Pure Aloha, when one node finishes speaking, it expects an acknowledgement in a finite amount of time - otherwise it simply retransmits the data. This scheme works well in small networks where the load is not high. But in large, load intensive networks where many nodes may want to transmit at the same time, this scheme fails miserably. This led to the development of Slotted Aloha.

Slotted Aloha

This is quite similar to Pure Aloha, differing only in the way transmissions take place. Instead of transmitting right at demand time, the sender waits for some time. This delay is specified as follows - the timeline is divided into equal slots and then it is required that transmission should take place only at slot boundaries. To be more precise, the slotted-Aloha makes the following assumptions:

• All frames consist of exactly L bits.
• Time is divided into slots of size L/R seconds (i.e., a slot equals the time to transmit one frame).
• Nodes start to transmit frames only at the beginnings of slots.
• The nodes are synchronized so that each node knows when the slots begin.
• If two or more frames collide in a slot, then all the nodes detect the collision event before the slot ends.

In this way, the number of collisions that can possibly take place is reduced by a huge margin. And hence, the performance become much better compared to Pure Aloha. collisions may only take place with nodes that are ready to speak at the same time. But nevertheless, this is a substantial reduction.

Carrier Sense Mutiple Access Protocols

In both slotted and pure ALOHA, a node's decision to transmit is made independently of the activity of the other nodes attached to the broadcast channel. In particular, a node neither pays attention to whether another node happens to be transmitting when it begins to transmit, nor stops transmitting if another node begins to interfere with its transmission. As humans, we have human protocols that allow allows us to not only behave with more civility, but also to decrease the amount of time spent "colliding" with each other in conversation and consequently increasing the amount of data we exchange in our conversations. Specifically, there are two important rules for polite human conversation:

1. Listen before speaking: If someone else is speaking, wait until they are done. In the networking world, this is termed carrier sensing - a node listens to the channel before transmitting. If a frame from another node is currently being transmitted into the channel, a node then waits ("backs off") a random amount of time and then again senses the channel. If the channel is sensed to be idle, the node then begins frame transmission. Otherwise, the node waits another random amount of time and repeats this process.
2. If someone else begins talking at the same time, stop talking. In the networking world, this is termed collision detection - a transmitting node listens to the channel while it is transmitting. If it detects that another node is transmitting an interfering frame, it stops transmitting and uses some protocol to determine when it should next attempt to transmit.

It is evident that the end-to-end channel propagation delay of a broadcast channel - the time it takes for a signal to propagate from one of the the channel to another - will play a crucial role in determining its performance. The longer this propagation delay, the larger the chance that a carrier-sensing node is not yet able to sense a transmission that has already begun at another node in the network.

CSMA- Carrier Sense Multiple Access

This is the simplest version CSMA protocol as described above. It does not specify any collision detection or handling. So collisions might and WILL occur and clearly then, this is not a very good protocol for large, load intensive networks. So, we need an improvement over CSMA - this led to the development of CSMA/CD.

CSMA/CD- CSMA with Collision Detection

In this protocol, while transmitting the data, the sender simultaneously tries to receive it. So, as soon as it detects a collission (it doesn't receive its own data) it stops transmitting. Thereafter, the node waits for some time interval before attempting to transmit again. Simply put, "listen while you talk". But, how long should one wait for the carrier to be freed? There are three schemes to handle this:

1. 1-Persistent: In this scheme, transmission proceeds immediately if the carrier is idle. However, if the carrier is busy, then sender continues to sense the carrier until it becomes idle. The main problem here is that, if more than one transmitters are ready to send, a collision is GUARANTEED!!
2. Non-Persistent: In this scheme, the broadcast channel is not monitored continuously. The sender polls it at random time intervals and transmits whenever the carrier is idle. This decreases the probability of collisions. But, it is not efficient in a low load situation, where number of collisions are anyway small. The problems it entails are:
   • If back-off time is too long, the idle time of carrier is wasted in some sense
   • It may result in long access delays
3. p-Persistent: Even if a sender finds the carrier to be idle, it uses a probabilistic distribution to determine whether to transmit or not. Put simply, "toss a coin to decide". If the carrier is idle, then transmission takes place with a probability p and the sender waits with a probability 1-p. This scheme is a good trade off between the Non-persistent and 1-persistent schemes. So, for low load situations, p is high (example: 1-persistent); and for high load situations, p may be lower. Clearly, the value of p plays an important role in determining the performance of this protocol. Also the same p is likely to provide different performance at different loads.

CSMA/CD doesn't work in some wireless scenarios called "hidden node" problems. Consider a situation, where there are 3 nodes - A, B and C communicating with each other using a wireless protocol. Morover, B can communicate with both A and C, but A and C lie outside each other's range and hence can't communicate directly with each other. Now, suppose both A and C want to communicate with B simultaneously. They both will sense the carrier to be idle and hence will begin transmission, and even if there is a collision, neither A nor C will ever detect it. B on the other hand will receive 2 packets at the same time and might not be able to understand either of them. To get around this problem, a better version called CSMA/CA was developed, specially for wireless applications.

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Image References:

• http://williams.comp.ncat.edu/Networks/multip10.gif
• http://williams.comp.ncat.edu/Networks/multip5.gif
• http://williams.comp.ncat.edu/Networks/multip13.gif
• http://www.e-networks.org/images/startopology.gif
• http://www.delmar.edu/Courses/CIS306/Primer/primf05.gif
• http://www.e-networks.org/images/ringtopology.gif
• http://www.laynetworks.com/images/aloha14.gif
• http://www.laynetworks.com/images/aloha15.gif

Computer Networks-Architecture Cont-part-II

 Session Layer
Session Layer It deals with the concept of Sessions i.e. when a user logins to a remote server he should be authenticated before getting access to the files and application programs. Another job of session layer is to establish and maintain sessions. If during the transfer of data between two machines the session breaks down, it is the session layer which re-establishes the connection. It also ensures that the data transfer starts from where it breaks keeping it transparent to the end user. e.g. In case of a session with a database server, this layer introduces check points at various places so that in case the connectoin is broken and reestablished, the transition running on the database is not lost even if the user has not committed. This activity is called Synchronization. Another function of this layer is Dialogue Control which determines whose turn is it to speak in a session. It is useful in video conferencing.

Presentation Layer

This layer is concerned with the syntax and semantics of the information transmitted. In order to make it possible for computers with different data representations to communicate data structures to be exchanged can be defined in abstract way alongwith standard encoding. It also manages these abstract data structres and allows higher level of data structres to be defined an exchange. It encodes the data in standard agreed way(network format). Suppose there are two machines A and B one follows 'Big Endian' and other 'Little Endian' for data representation. This layer ensures that the data transmitted by one gets converted in the form compatibale to othe machine. This layer is concerned with the syntax and semantics of the information transmitted.In order to make it possible for computers with different data representations to communicate data structures to be exchanged canbe defined in abstract way alongwith standard encoding. It also manages these abstract data structres and allows higher level of data structres to be defined an exchange. Other functions include compression, encryption etc.

Application Layer

The seventh layer contains the application protocols with which the user gains access to the network. The choice of which specific protocols and their associated functions are to be used at the application level is up to the individual user. Thus the boundary between the presentation layer and the application layer represents a separation of the protocols imposed by the network designers from those being selected and implemented by the network users.For example commonly used protocols are HTTP(for web browsing), FTP(for file transfer) etc.

Network Layers as in Practice

In most of the networks today, we do not follow the OSI model of seven layers. What is actually implemented is as follows. The functionality of Application layer and Presentation layer is merged into one and is called as the Application Layer. Functionalities of Session Layer is not implemented in most networks today. Also, the Data Link layer is split theoretically into MAC (Medium Access Control) Layer and LLC (Link Layer Control). But again in practice, the LLC layer is not implemented by most networks. So as of today, the network architecture is of 5 layers only.

Network Layers in Internet Today

Some Related Links on OSI Model and TCP Model

• http://en.wikipedia.org/wiki/OSI_model
• http://www.tcpipguide.com/free/t_OSIReferenceModelLayers.htm
• http://www.geocities.com/SiliconValley/Monitor/3131/ne/osimodel.html
• http://www.tech-faq.com/osi-model.shtml
• http://www.networkdictionary.com/protocols/osimodel.php

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Physical Layer

Physical layer is concerned with transmitting raw bits over a communication channel. The design issues have to do with making sure that when one side sends a 1 bit, it is recieved by the other side as 1 bit and not as 0 bit. In physical layer we deal with the communication medium used for transmission.

Types of Medium

Medium can be classified into 2 categories.

1. Guided Media : Guided media means that signals is guided  by the prescence of physical media i.e. signals are under control and remains in the physical wire. For eg. copper wire.
2. Unguided Media : Unguided Media means that there is no physical path for the signal to propogate. Unguided media are essentially electro-magnetic waves. There is no control on flow of signal. For eg. radio waves.

Communication Links

In a nework nodes are connected through links. The communication through links can be classified as

1. Simplex : Communication can take place only in one direction. eg. T.V broadcasting.
2. Half-duplex : Communication can take place in one direction at a time. Suppose node A and B are connected then half-duplex communication means that at a time data can flow from A to B or from B to A but not simultaneously. eg. two persons talking to each other such that when speaks the other listens and vice versa.
3. Full-duplex : Communication can take place simultaneously in both directions. eg. A discussion in a group without discipline.

Links can be further classified as

1. Point to Point : In this communication only two nodes are connected to each other. When a node sends a packet then it can be recieved only by the node on the other side and none else.
2. Multipoint : It is a kind of sharing communication, in which signal can be recieved by all nodes. This is also called broadcast.

Generally two kind of problems are associated in transmission of signals.

1. Attenuation : When a signal transmitts in a network then the quality of signal degrades as the signal travels longer distances in the wire. This is called attenuation. To improve quality of signal amplifiers are used at regular distances.
2. Noise : In a communication channel many signals transmits simultaneously, certain random signals are also present in the medium. Due to interference of these signals our signal gets disrupted a bit.

Bandwidth

Bandwidth simply means how many bits can be transmitted per second in the communication channel. In technical terms it indicates the width of frequency spectrum.

Transmission Media

Guided Transmission Media
In Guided transmission media generally two kind of materials are used.

1. Copper
   • Coaxial Cable
   • Twisted Pair
2. Optical Fiber

1. Coaxial Cable: Coaxial cable consists of an inner conductor and an outer conductor which are seperated by an insulator. The inner conductor is usually copper. The outer conductor is covered by a plastic jacket. It is named coaxial because the two conductors are coaxial. Typical diameter of coaxial cable lies between 0.4 inch to 1 inch. The most application of coaxial cable is cable T.V. The coaxial cable has high bandwidth, attenuation is less.
2. Twisted Pair: A Twisted pair consists of two insulated copper wires, typically 1mm thick. The wires are twisted togather in a helical form the purpose of twisting is to reduce cross talk interference between several pairs. Twisted Pair is much cheaper then coaxial cable but it is susceptible to noise and electromagnetic interference and attenuation is large.
   Twisted Pair can be further classified in two categories:
   Unshielded twisted pair: In this no insulation is provided, hence they are susceptible to interference.
   Shielded twisted pair: In this a protective thick insulation is provided but shielded twisted pair is expensive and not commonly used. The most common application of twisted pair is the telephone system. Nearly all telephones are connected to the telephone company office by a twisted pair. Twisted pair can run several kilometers without amplification, but for longer distances repeaters are needed. Twisted pairs can be used for both analog and digital transmission. The bandwidth depends on the thickness of wire and the distance travelled. Twisted pairs are generally limited in distance, bandwidth and data rate.
3. Optical Fiber: In optical fiber light is used to send data. In general terms prescence of light is taken as bit 1 and its absence as bit 0. Optical fiber consists of inner core of either glass or plastic. Core is surrounded by cladding of the same material but of different refrective index. This cladding is surrounded by a plastic jacket which prevents optical fiber from electromagnetic interferrence and harshy environments. It uses the principle of total internal reflection to transfer data over optical fibers. Optical fiber is much better in bandwidth as compared to copper wire, since there is hardly any attenuation or electromagnetic interference in optical wires. Hence there is less requirement to improve quality of signal, in long distance transmission. Disadvantage of optical fiber is that end points are fairly expensive. (eg. switches) Differences between different kinds of optical fibers:
   
   1. Depending on material
      • Made of glass
      • Made of plastic.
   2. Depending on radius
      • Thin optical fiber
      • Thick optical fiber
   3. Depending on light source
      • LED (for low bandwidth)
      • Injection lased diode (for high bandwidth)

Wireless Transmission

1. Radio: Radio is a general term that is used for any kind of frequency. But higher frequencies are usually termed as microwave and the lower frequency band comes under radio frequency. There are many application of radio. For eg. cordless keyboard, wireless LAN, wireless ethernet. but it is limited in range to only a few hundred meters. Depending on frequency radio offers different bandwidths.
2. Terrestrial microwave: In terrestrial microwave two antennas are used for communication. A focused beam emerges from an antenna and is recieved by the other antenna, provided that antennas should be facing each other with no obstacle in between. For this reason antennas are situated on high towers. Due to curvature of earth terristial microwave can be used for long distance communication with high bandwidth. Telecom department is also using this for long distance communication. An advantage of wireless communication is that it is not required to lay down wires in the city hence no permissions are required.
3. Satellite communication: Satellite acts as a switch in sky. On earth VSAT(Very Small Aperture Terminal) are used to transmit and recieve data from satellite. Generally one station on earth transmitts signal to satellite and it is recieved by many stations on earth. Satellite communication is generally used in those places where it is very difficult to obtain line of sight i.e. in highly irregular terristial regions. In terms of noise wireless media is not as good as the wired media. There are frequency band in wireless communication and two stations should not be allowed to transmit simultaneously in a frequency band. The most promising advantage of satellite is broadcasting. If satellites are used for point to point communication then they are expensive as compared to wired media.

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References of Images

• http://ai3.asti.dost.gov.ph/sat/levels.jpg
• http://www.ll.mit.edu/Image-Lib/photos/color-Satellite.jpg
• http://oldsite.vislab.usyd.edu.au/photonics/revolution/technology/images/twisted_pair.jpg
• http://www.radio-electronics.com/info/antennas/coax/cross_section_thro_coax.gif

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Computer Network-ISO-OSI 7-Layer Network Architecture

ISO-OSI 7-Layer Network Architecture

This lecture introduces the ISO-OSI layered architecture of Networks. According to the ISO standards, networks have been divided into 7 layers depending on the complexity of the fucntionality each of these layers provide. The detailed description of each of these layers is given in the notes below. We will first list the layers as defined by the standard in the increasing order of function complexity:

1. Physical Layer
2. Data Link Layer
3. Network Layer
4. Transport Layer
5. Session Layer
6. Presentation Layer
7. Application Layer

Physical Layer

This layer is the lowest layer in the OSI model. It helps in the transmission of data between two machines that are communicating through a physical medium, which can be optical fibres,copper wire or wireless etc. The following are the main functions of the physical layer:

1. Hardware Specification: The details of the physical cables, network interface cards, wireless radios, etc are a part of this layer.
   

   Coaxial Cable	Hybrid Cable	Wireless Card	Network Card

   
   
2. Encoding and Signalling: How are the bits encoded in the medium is also decided by this layer. For example, on the coppar wire medium, we can use differnet voltage levels for a certain time interval to represent '0' and '1'. We may use +5mV for 1nsec to represent '1' and -5mV for 1nsec to represent '0'. All the issues of modulation is dealt with in this layer. eg, we may use Binary phase shift keying for the representation of '1' and '0' rather than using different volatage levels if we have to transfer in RF waves.
   
   Binary Phase Shift Keying
   
3. Data Transmission and Reception: The transfer of each bit of data is the responsibility of this layer. This layer assures the transmissoin of each bit with a high probability. The transmission of the bits is not completely reliable as their is no error correction in this layer.
4. Topology and Network Design: The network design is the integral part of the physical layer. Which part of the network is the router going to be placed, where the switches will be used, where we will put the hubs, how many machines is each switch going to handle, what server is going to be placed where, and many such concerns are to be taken care of by the physical layer. The variosu kinds of netopologies that we decide to use may be ring, bus, star or a hybrid of these topologies depending on our requirements.
   
   
   

Data Link Layer

This layer provides reliable transmission of a packet by using the services of the physical layer which transmits bits over the medium in an unreliable fashion. This layer is concerned with :

1. Framing : Breaking input data into frames (typically a few hundred bytes) and caring about the frame boundaries and the size of each frame.
2. Acknowledgment : Sent by the receiving end to inform the source that the frame was received without any error.
3. Sequence Numbering : To acknowledge which frame was received.
4. Error Detection : The frames may be damaged, lost or duplicated leading to errors.The error control is on link to link basis.
5. Retransmission : The packet is retransmitted if the source fails to receive acknowledgment.
6. Flow Control : Necessary for a fast transmitter to keep pace with a slow receiver.

Data Link Layer

Network Layer

Its basic functions are routing and congestion control.
Routing: This deals with determining how packets will be routed (transferred) from source to destination. It can be of three types :

• Static : Routes are based on static tables that are "wired into" the network and are rarely changed.
• Dynamic : All packets of one application can follow different routes depending upon the topology of the network, the shortest path and the current network load.
• Semi-Dynamic : A route is chosen at the start of each conversation and then all the packets of the application follow the same route.

Routing
The services provided by the network can be of two types :

• Connection less service: Each packet of an application is treated as an independent entity. On each packet of the application the destination address is provided and the packet is routed.
• Connection oriented service: Here, first a connection is established and then all packets of the application follow the same route. To understand the above concept, we can also draw an analogy from the real life. Connection oriented service is modeled after the telephone system. All voice packets go on the same path after the connection is established till the connection is hung up. It acts like a tube ; the sender pushes the objects in at one end and the receiver takes them out in the same order at the other end. Connection less service is modeled after the postal system. Each letter carries the destination address and is routed independent of all the others. Here, it is possible that the letter sent first is delayed so that the second letter reaches the destination before the first letter.

Congestion Control: A router can be connected to 4-5 networks. If all the networks send packet at the same time with maximum rate possible then the router may not be able to handle all the packets and may drop some/all packets. In this context the dropping of the packets should be minimized and the source whose packet was dropped should be informed. The control of such congestion is also a function of the network layer. Other issues related with this layer are transmitting time, delays, jittering.

Internetworking: Internetworks are multiple networks that are connected in such a way that they act as one large network, connecting multiple office or department networks. Internetworks are connected by networking hardware such as routers, switches, and bridges.Internetworking is a solution born of three networking problems: isolated LANs, duplication of resources, and the lack of a centralized network management system. With connected LANs, companies no longer have to duplicate programs or resources on each network. This in turn gives way to managing the network from one central location instead of trying to manage each separate LAN. We should be able to transmit any packet from one network to any other network even if they follow different protocols or use different addressing modes.

Inter-Networking
Network Layer does not guarantee that the packet will reach its intended destination. There are no reliability guarantees.

Transport Layer

Its functions are :

• Multiplexing / Demultiplexing : Normally the transport layer will create distinct network connection for each transport connection required by the session layer. The transport layer may either create multiple network connections (to improve throughput) or it may multiplex several transport connections onto the same network connection (because creating and maintaining networks may be expensive). In the latter case, demultiplexing will be required at the receiving end. A point to note here is that communication is always carried out between two processes and not between two machines. This is also known as process-to-process communication.
• Fragmentation and Re-assembly : The data accepted by the transport layer from the session layer is split up into smaller units (fragmentation) if needed and then passed to the network layer. Correspondingly, the data provided by the network layer to the transport layer on the receiving side is re-assembled.
  

  Fragmentation	Reassembly

  
  
• Types of service : The transport layer also decides the type of service that should be provided to the session layer. The service may be perfectly reliable, or may be reliable within certain tolerances or may not be reliable at all. The message may or may not be received in the order in which it was sent. The decision regarding the type of service to be provided is taken at the time when the connection is established.
• Error Control : If reliable service is provided then error detection and error recovery operations are also performed. It provides error control mechanism on end to end basis.
• Flow Control : A fast host cannot keep pace with a slow one. Hence, this is a mechanism to regulate the flow of information.
• Connection Establishment / Release : The transport layer also establishes and releases the connection across the network. This requires some sort of naming mechanism so that a process on one machine can indicate with whom it wants to communicate.

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References of Images

• http://www.putergeek.com/.../ pci_combo_card_sm.jpg
• http://blue.utb.edu/libertad/clipart/pi_wireless_pc_card_b.jpg
• http://www.commscope.com/ images/hybrids.jpg hybrid cable
• http://www.cba.nau.edu/facstaff/maris-j/SavedStuff/Images/net_topo.gif
• http://www.ces.net/doc/2003/research/xl-unicast-routing.gif
• http://www.infinitygroup.com/images/internetworking.gif
• http://www.microway.com/.../ data_link_layer.gif
• http://searchnetworking.techtarget.com/WhatIs/images/coaxla.gif
• http://www.df.lth.se/~pkj/thesis_report/img13.gif
• http://www.df.lth.se/~pkj/ thesis_report/img12.gif
• http://www.ifla.org/VI/5/reports/rep3/rep3-2.gif
• http://msp.gsfc.nasa.gov/tdrss/bpsk.gif