1. WIRED NETWORKS
2. ADHOC/MOBILE IP
3. WIRELESS SENSOR NETWORKS
WIRED NETWORKS
A wired network connects devices to the Internet or other network using cables. The most common wired networks use cables connected to Ethernet ports on the network router on one end and to a computer or other device on the cable's opposite end.
Ethernet and wireless networks each have advantages and disadvantages; depending on your needs, one may serve you better than the other. Wired networks provide users with plenty of security and the ability to move lots of data very quickly. Wired networks are typically faster than wireless networks, and they can be very affordable. However, the cost of Ethernet cable can add up - the more computers on your network and the farther apart they are, the more expensive your network will be. In addition, unless you're building a new house and installing Ethernet cable in the walls, you'll be able to see the cables running from place to place around your home, and wires can greatly limit your mobility.
For more click here : WIRED VS WIRELESS
The following list shows the models and protocols supported by ns2 in wired configuration;
ROUTING IN WIRED NETWORKS
Ethernet and wireless networks each have advantages and disadvantages; depending on your needs, one may serve you better than the other. Wired networks provide users with plenty of security and the ability to move lots of data very quickly. Wired networks are typically faster than wireless networks, and they can be very affordable. However, the cost of Ethernet cable can add up - the more computers on your network and the farther apart they are, the more expensive your network will be. In addition, unless you're building a new house and installing Ethernet cable in the walls, you'll be able to see the cables running from place to place around your home, and wires can greatly limit your mobility.
For more click here : WIRED VS WIRELESS
The following list shows the models and protocols supported by ns2 in wired configuration;
ROUTING IN WIRED NETWORKS
UNICAST ROUTING
In computer networking, Unicast transmission is the sending of messages to a single network destination identified by a unique address.Unicast routing is the process of forwarding unicasted traffic from a source to a destination on an internetwork. Unicasted traffic is destined for a unique address.
If an IP Unicast packet passes through a switch that does not know the location of the associated MAC Address, the packet will be broadcast to all ports on the switch. This failure of Unicast to 'cast to a single device' is called a Unicast flood.
Unicast messaging is used for all network processes in which a private or unique resource is requested.
Certain network applications which are mass-distributed are too costly to be conducted with unicast transmission since each network connection consumes computing resources on the sending host and requires its own separate network bandwidth for transmission. Such applications include streaming media of many forms. Internet radio stations using unicast connections may have high bandwidth costs.
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UNICAST ROUTING |
>> URP SECTION 2
MULTICAST ROUTING
In computer networking, multicast is the delivery of a message or information to a group of destination computers simultaneously in a single transmission from the source. Copies are automatically created in other network elements, such as routers, but only when the topology of the network requires it.
Multicast is most commonly implemented in IP multicast, which is often employed in Internet Protocol (IP) applications of streaming media and Internet television. In IP multicast the implementation of the multicast concept occurs at the IP routing level, where routers create optimal distribution paths for datagrams sent to a multicast destination address.
At the Data Link Layer, multicast describes one-to-many distribution such as Ethernet multicast addressing, Asynchronous Transfer Mode (ATM) point-to-multipoint virtual circuits (P2MP) or Infiniband multicast.
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| MULTICAST ROUTING |
IP MULTICAST
IP multicast is a technique for one-to-many and many-to-many real-time communication over an IP infrastructure in a network. It scales to a larger receiver population by requiring neither prior knowledge of a receiver's identity nor prior knowledge of the number of receivers. Multicast uses network infrastructure efficiently by requiring the source to send a packet only once, even if it needs to be delivered to a large number of receivers. The nodes in the network (typically network switches and routers) take care of replicating the packet to reach multiple receivers such that messages are sent over each link of the network only once. The most common low-level protocol to use multicast addressing is User Datagram Protocol (UDP). By its nature, UDP is not reliable—messages may be lost or delivered out of order. Reliable multicast protocols such as Pragmatic General Multicast (PGM) have been developed to add loss detection and retransmission on top of IP multicast.
Key concepts in IP multicast include an IP multicast group address, a multicast distribution tree and receiver driven tree creation.
An IP multicast group address is used by sources and the receivers to send and receive multicast messages. Sources use the group address as the IP destination address in their data packets. Receivers use this group address to inform the network that they are interested in receiving packets sent to that group. For example, if some content is associated with group 239.1.1.1, the source will send data packets destined to 239.1.1.1. Receivers for that content will inform the network that they are interested in receiving data packets sent to the group 239.1.1.1. The receiver joins 239.1.1.1. The protocol typically used by receivers to join a group is called the Internet Group Management Protocol (IGMP).
With routing protocols based on shared trees, once the receivers join a particular IP multicast group, a multicast distribution tree is constructed for that group. The protocol most widely used for this is Protocol Independent Multicast (PIM). It sets up multicast distribution trees such that data packets from senders to a multicast group reach all receivers which have joined the group. For example, all data packets sent to the group 239.1.1.1 are received by receivers who joined 239.1.1.1. There are variations of PIM implementations: Sparse Mode (SM), Dense Mode (DM), Source Specific Mode (SSM) and Bidirectional Mode (Bidir, or Sparse-Dense Mode, SDM). Of these, PIM-SM is the most widely deployed as of 2006;[citation needed] SSM and Bidir are simpler and scalable variations developed more recently and are gaining in popularity.
IP multicast operation does not require an active source to know about the receivers of the group. The multicast tree construction is receiver driven and is initiated by network nodes which are close to the receivers. IP multicast scales to a large receiver population. The IP multicast model has been described by Internet architect Dave Clark as, "You put packets in at one end, and the network conspires to deliver them to anyone who asks."
IP multicast creates state information per multicast distribution tree in the network. If a router is part of 1000 multicast trees, it has 1000 multicast routing and forwarding entries. On the other hand, a multicast router does not need to know how to reach all other multicast trees in the Internet. It only needs to know about multicast trees for which it has downstream receivers. This is key to scaling multicast-addressed services. It is very unlikely that core Internet routers would need to keep state for all multicast distribution trees,[citation needed] they only need to keep state for trees with downstream membership. In contrast, a unicast router needs to know how to reach all other unicast addresses in the Internet, even if it does this using just a default route. For this reason, aggregation is key to scaling unicast routing. Also, there are core routers that carry routes in the hundreds of thousands because they contain the Internet routing table.
ROUTING
Each host (and in fact each application on the host) that wants to be a receiving member of a multicast group (i.e. receive data corresponding to a particular multicast address) must use the Internet Group Management Protocol (IGMP) to join. Adjacent routers also use this protocol to communicate.
In unicast routing, each router examines the destination address of an incoming packet and looks up the destination in a table to determine which interface to use in order for that packet to get closer to its destination. The source address is irrelevant to the router. However, in multicast routing, the source address (which is a simple unicast address) is used to determine data stream direction. The source of the multicast traffic is considered upstream. The router determines which downstream interfaces are destinations for this multicast group (the destination address), and sends the packet out through the appropriate interfaces. The term reverse path forwarding is used to describe this concept of routing packets away from the source, rather than towards the destination.
A number of errors can happen if packets intended for unicast are accidentally sent to a multicast address; in particular, sending ICMP packets to a multicast address has been used in the context of DoS attacks as a way of achieving packet amplification.
On the local network, multicast delivery is controlled by IGMP (on IPv4 network) and MLD (on IPv6 network); inside a routing domain, PIM or MOSPF are used; between routing domains, one uses inter-domain multicast routing protocols, such as MBGP.
The following are some common delivery and routing protocols used for multicast distribution:
- Internet Group Management Protocol (IGMP)
- Protocol Independent Multicast (PIM)
- Distance Vector Multicast Routing Protocol (DVMRP)
- Multicast Open Shortest Path First (MOSPF)
- Multicast BGP (MBGP)
- Multicast Source Discovery Protocol (MSDP)
- Multicast Listener Discovery (MLD)
- GARP Multicast Registration Protocol (GMRP)
For more click here: >> MULTICAST ROUTING
>> INTRODUCTION TO IP MULTICAST
>> MULTICAST ROUTING PROTOCOLS
>> INTRODUCTION TO IP MULTICAST
>> MULTICAST ROUTING PROTOCOLS
HIERARCHICAL ROUTING
Hierarchical routing is a method of routing in networks that is based on hierarchical addressing. Hierarchical routing is the procedure of arranging routers in a hierarchical manner. A good example would be to consider a corporate intranet. Most corporate intranets consist of a high speed backbone network. Connected to this backbone are routers which are in turn connected to a particular work group. These work groups occupy a unique LAN. The reason this is a good arrangement is because even though there might be dozens of different work groups, the span (maximum hop count to get from one host to any other host on the network) is 2. Even if the work groups divided their LAN network into smaller partitions, the span could only increase to 4 in this particular example.
Considering alternative solutions with every router connected to every other router, or if every router was connected to 2 routers, shows the convenience of hierarchical routing. It decreases the complexity of network topology, increases routing efficiency, and causes much less congestion because of fewer routing advertisements. With hierarchical routing, only core routers connected to the backbone are aware of all routes. Routers that lie within a LAN only know about routes in the LAN. Unrecognized destinations are passed to the default route.
Most Transmission Control Protocol/Internet Protocol (TCP/IP) routing is based on a two-level hierarchical routing in which an IP address is divided into a network portion and a host portion. Gateways use only the network portion until an IP datagram reaches a gateway that can deliver it directly. Additional levels of hierarchical routing are introduced by the addition of subnetworks.
In hierarchical routing, routers are classified in groups known as regions. Each router has only the information about the routers in its own region and has no information about routers in other regions. So routers just save one record in their table for every other region. In this example, we have classified our network into five regions (see below).
If A wants to send packets to any router in region 2 (D, E, F or G), it sends them to B, and so on. As you can see, in this type of routing, the tables can be summarized, so network efficiency improves. The above example shows two-level hierarchical routing. We can also use three- or four-level hierarchical routing.
In three-level hierarchical routing, the network is classified into a number of clusters. Each cluster is made up of a number of regions, and each region contains a number or routers. Hierarchical routing is widely used in Internet routing and makes use of several routing protocols.
In here, DV algorithms are used to find best routes between nodes. In the situation depicted below, every node of the network has to save a routing table with 17 records. Here is a typical graph and routing table for A:
TRAFFIC SOURCES
FILE TRANSFER PROTOCOL [FTP]
File Transfer Protocol (FTP) is a standard Internet protocol for transmitting files between computers on the Internet. Like the Hypertext Transfer Protocol (HTTP), which transfers displayable Web pages and related files, and the Simple Mail Transfer Protocol (SMTP), which transfers e-mail, FTP is an application protocol that uses the Internet's TCP/IP protocols. FTP is commonly used to transfer Web page files from their creator to the computer that acts as their server for everyone on the Internet. It's also commonly used to download programs and other files to your computer from other servers.
As a user, you can use FTP with a simple command line interface (for example, from the Windows MS-DOS Prompt window) or with a commercial program that offers a graphical user interface. Your Web browser can also make FTP requests to download programs you select from a Web page. Using FTP, you can also update (delete, rename, move, and copy) files at a server. You need to logon to an FTP server. However, publicly available files are easily accessed using anonymous FTP.
Basic FTP support is usually provided as part of a suite of programs that come with TCP/IP. However, any FTP client program with a graphical user interface usually must be downloaded from the company that makes it.
Telnet is a user command and an underlying TCP/IP protocol for accessing remote computers. Through Telnet, an administrator or another user can access someone else's computer remotely. On the Web, HTTP and FTP protocols allow you to request specific files from remote computers, but not to actually be logged on as a user of that computer. With Telnet, you log on as a regular user with whatever privileges you may have been granted to the specific application and data on that computer.
A Telnet command request looks like this (the computer name is made-up):
telnet the.libraryat.whatis.edu
The result of this request would be an invitation to log on with a userid and a prompt for a password. If accepted, you would be logged on like any user who used this computer every day.
Telnet is most likely to be used by program developers and anyone who has a need to use specific applications or data located at a particular host computer.
For more click here: TELNET
CONSTANT BIT RATE [CBR]
Constant bitrate (CBR) is a term used in telecommunications, relating to the quality of service. Compare with variable bitrate. When referring to codecs, constant bit rate encoding means that the rate at which a codec's output data should be consumed is constant. CBR is useful for streaming multimedia content on limited capacity channels since it is the maximum bit rate that matters, not the average, so CBR would be used to take advantage of all of the capacity. CBR would not be the optimal choice for storage as it would not allocate enough data for complex sections (resulting in degraded quality) while wasting data on simple sections.
The problem of not allocating enough data for complex sections could be solved by choosing a high bitrate (e.g., 256 kbit/s or 320 kbit/s) to ensure that there will be enough bits for the entire encoding process, though the size of the file at the end would be proportionally larger.
Most coding schemes such as Huffman coding or run-length encoding produce variable-length codes, making perfect CBR difficult to achieve. This is partly solved by varying the quantization (quality), and fully solved by the use of padding. (However, CBR is implied in a simple scheme like reducing all 16-bit audio samples to 8 bits.)
In the case of streaming video as a CBR, the source could be under the CBR data rate target. So in order to complete the stream, it's necessary to add stuffing packets in the stream to reach the data rate wanted. These packets are totally neutral and don't affect the stream.
For more click here: CBR
QUEUING DISCIPLINES
DROP TAIL
Considering alternative solutions with every router connected to every other router, or if every router was connected to 2 routers, shows the convenience of hierarchical routing. It decreases the complexity of network topology, increases routing efficiency, and causes much less congestion because of fewer routing advertisements. With hierarchical routing, only core routers connected to the backbone are aware of all routes. Routers that lie within a LAN only know about routes in the LAN. Unrecognized destinations are passed to the default route.
Most Transmission Control Protocol/Internet Protocol (TCP/IP) routing is based on a two-level hierarchical routing in which an IP address is divided into a network portion and a host portion. Gateways use only the network portion until an IP datagram reaches a gateway that can deliver it directly. Additional levels of hierarchical routing are introduced by the addition of subnetworks.
In hierarchical routing, routers are classified in groups known as regions. Each router has only the information about the routers in its own region and has no information about routers in other regions. So routers just save one record in their table for every other region. In this example, we have classified our network into five regions (see below).
In three-level hierarchical routing, the network is classified into a number of clusters. Each cluster is made up of a number of regions, and each region contains a number or routers. Hierarchical routing is widely used in Internet routing and makes use of several routing protocols.
In here, DV algorithms are used to find best routes between nodes. In the situation depicted below, every node of the network has to save a routing table with 17 records. Here is a typical graph and routing table for A:
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| Network Graph and Routing Table of A |
For more click here: HIERARCHIAL ROUTING
TRANSPORTATION
TRANSMISSION CONTROL PROTOCOL [TCP]
The Transmission Control Protocol (TCP) is one of the core protocols of the Internet protocol suite (IP), and is so common that the entire suite is often called TCP/IP. TCP provides reliable, ordered and error-checked delivery of a stream of octets between programs running on computers connected to a local area network, intranet or the public Internet. It resides at the transport layer.
Web browsers use TCP when they connect to servers on the World Wide Web, and it is used to deliver email and transfer files from one location to another. HTTP, HTTPS, SMTP, POP3, IMAP, SSH, FTP, Telnet and a variety of other protocols are typically encapsulated in TCP.
While IP takes care of handling the actual delivery of the data, TCP takes care of keeping track of the individual units of data (called packets) that a message is divided into for efficient routing through the Internet.
For example, when an HTML file is sent to you from a Web server, the Transmission Control Protocol (TCP) program layer in that server divides the file into one or more packets, numbers the packets, and then forwards them individually to the IP program layer. Although each packet has the same destination IP address, it may get routed differently through the network. At the other end (the client program in your computer), TCP reassembles the individual packets and waits until they have arrived to forward them to you as a single file.
TCP is known as a connection-oriented protocol, which means that a connection is established and maintained until such time as the message or messages to be exchanged by the application programs at each end have been exchanged. TCP is responsible for ensuring that a message is divided into the packets that IP manages and for reassembling the packets back into the complete message at the other end. In the Open Systems Interconnection (OSI) communication model, TCP is in layer 4, the Transport Layer.
For more click here: TCP OVERVIEW
USER DATAGRAM PROTOCOL
The User Datagram Protocol (UDP) is a transport layer protocol defined for use with the IP network layer protocol. It is defined by RFC 768 written by John Postel. It provides a best-effort datagram service to an End System (IP host).
The service provided by UDP is an unreliable service that provides no guarantees for delivery and no protection from duplication (e.g. if this arises due to software errors within an Intermediate System (IS)). The simplicity of UDP reduces the overhead from using the protocol and the services may be adequate in many cases.
UDP provides a minimal, unreliable, best-effort, message-passing transport to applications and upper-layer protocols. Compared to other transport protocols, UDP and its UDP-Lite variant are unique in that they do not establish end-to-end connections between communicating end systems. UDP communication consequently does not incur connection establishment and teardown overheads and there is minimal associated end system state. Because of these characteristics, UDP can offer a very efficient communication transport to some applications, but has no inherent congestion control or reliability. A second unique characteristic of UDP is that it provides no inherent On many platforms, applications can send UDP datagrams at the line rate of the link interface, which is often much greater than the available path capacity, and doing so would contribute to congestion along the path, applications therefore need to be designed responsibly [RFC 4505].
One increasingly popular use of UDP is as a tunneling protocol, where a tunnel endpoint encapsulates the packets of another protocol inside UDP datagrams and transmits them to another tunnel endpoint, which decapsulates the UDP datagrams and forwards the original packets contained in the payload. Tunnels establish virtual links that appear to directly connect locations that are distant in the physical Internet topology, and can be used to create virtual (private) networks. Using UDP as a tunneling protocol is attractive when the payload protocol is not supported by middleboxes that may exist along the path, because many middleboxes support UDP transmissions.
UDP does not provide any communications security. Applications that need to protect their communications against eavesdropping, tampering, or message forgery therefore need to separately provide security services using additional protocol mechanisms.
For more click here: >> PART 1
>> PART 2
TRAFFIC SOURCES
FILE TRANSFER PROTOCOL [FTP]
File Transfer Protocol (FTP) is a standard Internet protocol for transmitting files between computers on the Internet. Like the Hypertext Transfer Protocol (HTTP), which transfers displayable Web pages and related files, and the Simple Mail Transfer Protocol (SMTP), which transfers e-mail, FTP is an application protocol that uses the Internet's TCP/IP protocols. FTP is commonly used to transfer Web page files from their creator to the computer that acts as their server for everyone on the Internet. It's also commonly used to download programs and other files to your computer from other servers.
As a user, you can use FTP with a simple command line interface (for example, from the Windows MS-DOS Prompt window) or with a commercial program that offers a graphical user interface. Your Web browser can also make FTP requests to download programs you select from a Web page. Using FTP, you can also update (delete, rename, move, and copy) files at a server. You need to logon to an FTP server. However, publicly available files are easily accessed using anonymous FTP.
Basic FTP support is usually provided as part of a suite of programs that come with TCP/IP. However, any FTP client program with a graphical user interface usually must be downloaded from the company that makes it.
For more click here: FTP
TELNET
Telnet is a user command and an underlying TCP/IP protocol for accessing remote computers. Through Telnet, an administrator or another user can access someone else's computer remotely. On the Web, HTTP and FTP protocols allow you to request specific files from remote computers, but not to actually be logged on as a user of that computer. With Telnet, you log on as a regular user with whatever privileges you may have been granted to the specific application and data on that computer.
A Telnet command request looks like this (the computer name is made-up):
telnet the.libraryat.whatis.edu
The result of this request would be an invitation to log on with a userid and a prompt for a password. If accepted, you would be logged on like any user who used this computer every day.
Telnet is most likely to be used by program developers and anyone who has a need to use specific applications or data located at a particular host computer.
For more click here: TELNET
CONSTANT BIT RATE [CBR]
Constant bitrate (CBR) is a term used in telecommunications, relating to the quality of service. Compare with variable bitrate. When referring to codecs, constant bit rate encoding means that the rate at which a codec's output data should be consumed is constant. CBR is useful for streaming multimedia content on limited capacity channels since it is the maximum bit rate that matters, not the average, so CBR would be used to take advantage of all of the capacity. CBR would not be the optimal choice for storage as it would not allocate enough data for complex sections (resulting in degraded quality) while wasting data on simple sections.
The problem of not allocating enough data for complex sections could be solved by choosing a high bitrate (e.g., 256 kbit/s or 320 kbit/s) to ensure that there will be enough bits for the entire encoding process, though the size of the file at the end would be proportionally larger.
Most coding schemes such as Huffman coding or run-length encoding produce variable-length codes, making perfect CBR difficult to achieve. This is partly solved by varying the quantization (quality), and fully solved by the use of padding. (However, CBR is implied in a simple scheme like reducing all 16-bit audio samples to 8 bits.)
In the case of streaming video as a CBR, the source could be under the CBR data rate target. So in order to complete the stream, it's necessary to add stuffing packets in the stream to reach the data rate wanted. These packets are totally neutral and don't affect the stream.
For more click here: CBR
QUEUING DISCIPLINES
DROP TAIL
It is a simple queue mechanism that is used by the routers that when packets should to be drop. In this mechanism each packet is treated identically and when queue filled to its maximum capacity the newly incoming packets are dropped until queue have sufficient space to accept incoming traffic.
When a queue is filled the router start to discard all extra packets thus dropping the tail of mechanism. The loss of packets (datagram’s) causes the sender to enter slow start which decreases the throughput and thus increases its congestion window.
FAIR QUEUING
It is a queuing mechanism that is used to allow multiple packets flow to comparatively share the link capacity. Routers have multiple queues for each output line for every user. When a line as available as idle routers scans the queues through round robin and takes first packet to next queue. FQ also ensure about the maximum throughput of the network. For more efficiency weighted queue mechanism is also used.
DEFICIT ROUND ROBIN
It is a modified weighted round robin scheduling mechanism. It can handle packets of different size without having knowledge of their mean size. Deficit Round Robin keeps track of credits for each flow. It derives ideas from Fair Queuing and Stochastic FQ. It uses hashing to determine the queue to which a flow has to be assigned and collisions automatically reduce the bandwidth guaranteed to the flow. Each queue is assigned a quantum and can send a packet of size that can fit in the available quantum. If not, the idle quantum gets added to this meticulous queue’s deficit and the packet can be sent in the next round. The quantum size is a very vital parameter in the DRR scheme, determining the upper bound on the latency as well as the throughput.
This queue mechanism used a well-designed idea to get better performance and can also be implemented in a cost effectiveness manner. It provides a generic framework to implement fair queuing efficiently.
Although DRR serves fine for throughput fairness, but when it comes to Latency bounds it performs rather poorly. Also it does not operate well for real time traffic. The queuing delays introduced by DRR can have exciting results on the congestion window sizes.
RANDOM EARLY DETECTION
Random Early Detection (RED) is a congestion avoidance queuing mechanism (as opposed to a congestion administration mechanism) that is potentially useful, particularly in high-speed transit networks. Sally Floyd and Van Jacobson projected it in various papers in the early 1990s.It is active queue management mechanism. It operates on the average queue size and drop packets on the basis of statistics information. If the buffer is empty all incoming packets are acknowledged. As the queue size increase the probability for discarding a packet also increase. When buffer is full probability becomes equal to 1 and all incoming packets are dropped.
RED is capable to evade global synchronization of TCP flows, preserve high throughput as well as a low delay and attains fairness over multiple TCP connections, etc. It is the most common mechanism to stop congestive collapses.
When the queue in the router starts to fill then a small percentage of packets are discarded. This is deliberate to start TCP sources to decrease their window sizes and hence suffocate back the data rate. This can cause low rates of packet loss in Voice over IP streams. There have been reported incidences in which a series of routers apply RED at the same time, resulting in bursts of packet loss.
STOCHASTIC FAIR QUEUING
This queuing mechanism is based on fair queuing algorithm and proposed by John Nagle in 1987. Because it is impractical to have one queue for each conversation SFQ uses a hashing algorithm which divides the traffic over a limited number of queues. It is not so efficient than other queues mechanisms but it also requires less calculation while being almost perfectly fair. It is called "Stochastic" due to the reason that it does not actually assign a queue for every session; it has an algorithm which divides traffic over a restricted number of queues using a hashing algorithm. SFQ assigns a pretty large number of FIFO queues.
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