Alice in Telecomland

Witricity, standing for wireless electricity, is a term used to describe the ability to provide electricity to remote objects without wires. Witricity is nothing but wireless electricity used to transmit electrical energy from one object to another without using wires. due to witricity some device will not require batteries to operate.WiTricity is based on strong coupling between electromagnetic resonant objects .This differs from other methods like simple induction, microwaves, or air ionization.

Basic principle:

Consider two self resonating copper coils of same resonating  frequency  with  a  diameter  20  inches  each. One  copper wire  is  connected  to  the  power  source  (WiTricity  transmitter), while the other copper wire is connected to the device (WiTricity Receiver).  The  electric  power  from  the  power  source  causes  the  copper  coil connected  to  it  to  start  oscillating  at  a  particular  (MHz)  frequency. Subsequently,  the space around  the copper coil gets  filled with non-magnetic  radiations.  This  generated magnetic  field  further  transfers the power  to  the other copper  coil  connected  to  the  receiver. Since this coil is also of the same frequency, it starts oscillating at the same frequency as  the  first coil. This  is known as  ‘coupled resonance’ and is the principle behind WiTricity.

The system consists of transmitters and receivers having magnetic loop antennas tuned to same frequency.The receiving devices must be no more than about a quarter wavelength from the transmitter .During normal operation, an antenna emits electromagnetic fields into the surrounding nearfield region, then a portion of the field energy decays since it is re-absorbed by the antenna, while the remainder is radiated into the environment as EM waves.Non-radiative wireless power would have limited range, and the range would be shorter for smaller-size receivers .

It has been more than 100 years since this technology was initially discovered. witricity is not a new concept. it is safe and sound for human beings as well. We need it badly because of its plenty of merits.This could be useful to power consumer and industrial electronics like cell phones, laptops, etc

RFID stands for Radio Frequency Identification. It is a method of identifying unique items using radio waves. It can be applied to or incorporated into a product, animal, or person for the purpose of identification and tracking using radio waves .Typical RFID systems are made up of 2 major components: readers and tags. The reader, sometimes called the interrogator, sends and receives RF data to and from the tag via antennas. A reader may have multiple antennas that are responsible for sending and receiving the radio waves. The tag, or transponder, is made up of the microchip that stores the data, an antenna, and a carrier to which the chip and antenna are mounted.

There are many different versions of RFID that operate at different radio frequencies. The choice of frequency is dependent on the requirements of the application. Three primary frequency bands have been allocated for RFID use.

Low Frequency (125/134KHz) – Most commonly used for access control and asset tracking.
Mid-Frequency (13.56 MHz) – Used where medium data rate and read ranges are required.
Ultra High-Frequency (850 MHz to 950 MHz and 2.4 GHz to 2.5 GHz) – offer the longest read ranges and high reading speeds.

RFID technologies cannot reliably cover areas wider than 4 to 5 feet, unsuitable for wide openings that are the norm in manufacturing, distribution and store receiving dock environments. Since UHF (Ultra High Frequency) can cover portals up to 9 feet wide it is gaining industry support as the choice bandwidth for inventory tracking applications including pallets and cases.

RFID tags are further broken down into two categories:

Active RFID Tags are battery powered. They broadcast a signal to the reader and can transmit over the greatest distances 100+ feet and are used to track high value goods like vehicles and large containers of goods. Shipboard containers are a good example of an active RFID tag application.

Passive RFID Tags do not contain a battery. Instead, they draw their power from the reader. The reader transmits a low power radio signal through its antenna to the tag, which in turn receives it through its own antenna to power the integrated circuit (chip). The tag will briefly converse with the reader for verification and the exchange of data. As a result, passive tags can transmit information over shorter distances (10 feet or less) than active tags. They have a smaller memory capacity and are considerably lower in cost making them ideal for tracking lower cost items.

There are two basic types of chips available on RFID tags, Read-Only and Read-Write. Read only chips are programmed with unique information stored on them during the manufacturing process. The information on read-only chips can never be changed. With Read-Write chips, the user can add information to the tag or write over existing information when the tag is within range of the reader. Read-Write chips are more expensive that Read Only chips. Another method used is something called a “WORM” chip (Write Once Read Many). It can be written once and then becomes “Read Only” afterwards.

Radio Frequency Identification (RFID) technology has been attracting considerable attention with the expectation of improved supply chain visibility for both suppliers and retailers.RFID is expected to provide huge advantages to manufacturers by offering the tools to better plan production and respond more quickly to market demand. It will facilitate automation of inventory counts and speed shipping and receiving at the distribution level.

LTE (Long Term Evolution) is the step toward the 4G, designed to increase the capacity and speed of mobile telephone networks. LTE is a set of enhancements to the Universal Mobile Telecommunications System. The LTE specification provides downlink peak rates of at least 100 Mbps, an uplink of at least 50 Mbit/s and RAN round-trip times of less than 10 milli sec. LTE supports scalable carrier bandwidths, from 20 MHz down to 1.4 MHz and supports both Frequency Division Duplexing and Time Division Duplexing.

In addition to enabling fixed to mobile migrations of Internet applications such as Voice over IP (VoIP), video streaming, music downloading, mobile TV and many others, LTE networks will also provide the capacity to support an explosion in demand for connectivity from a new generation of consumer devices tailored to those new mobile applications. 3GPP LTE radio technology is optimized to enhance networks by enabling significant new high capacity mobile broadband applications and services, whilst providing cost efficient ubiquitous mobile coverage.

LTE characteristics include:

• Peak LTE throughputs (high spectral efficiency)

• Increased Spectrum efficiency

• Ultra low Latency

- Less than 10 msec for round-trip delay (RTD) from UE to server

- Reduced call setup times (50-100ms)

- wired user experience

• Capacity per cell

- 200 users for 5 MHz, 400 users in larger spectrum allocations

• Flexible spectrum use maximizes flexibility

- 1.4, 3/3.2, 5, 10, 15, 20 MHz

- All frequencies of IMT-2000: 450 MHz to 2.6 GHz

Another key driver behind LTE is the reduction of the cost per byte, which is expected to decrease by a factor of six compared with HSPA today. LTE has multiple inputs and multiple output & collaborative MIMO. Alcatel Lucent, Nokia & Motorola are working to make further enhancements to make LTE successful in industry and to do further enhancements for 4G.

Voice over Internet Protocol is a transmission technology for delivery of voice communications over the Internet or other packet-switched networks frequently encountered terms with VoIP are IP telephony and Internet telephony, as well as voice over broadband, broadband telephony, and broadband phone. VoIP systems usually interface with the traditional public switched telephone network (PSTN) to allow for transparent phone communications worldwide. VoIP can be a benefit for reducing communication and infrastructure costs by routing phone calls over existing data networks and avoiding duplicate network systems. Voice-over-IP systems carry telephony speech as digital audio, typically reduced in data rate using speech data compression techniques, packetized in small units of typically tens of milliseconds of speech, and encapsulated in a packet stream over IP.

VoIP components perform the same features as the PSTN network, which are

• Signaling
• Database services
• Call connect and disconnect (bearer control)
• CODEC operations

·         SIGNALING

The signaling in a VoIP network activates and coordinates the various components to complete a call. Although the underling nature of the signaling is the same, there are some technical and architectural differences in a VoIP network.

Signaling in a VoIP network is accomplished by the exchange of IP datagram messages between the components. The format of these messages is covered by any number of standard protocols. Regardless of which protocol and product suites that is used, these message streams are critical to the function of a voice-enabled network and might need special treatment to guarantee their delivery.

·         DATABASE SERVICES

Database services are a way to locate an endpoint and translate the addressing that two networks use. For example, the PSTN uses phone numbers to identify endpoints, while a VoIP network could use an IP address (address abstraction could be accomplished with DNS) and port numbers to identify an endpoint. A call control database contains these mappings and translations. Another important feature is the generation of transaction reports for billing purposes. You can employ additional logic to provide network security, such as to deny a specific endpoint from making overseas calls on the PSTN side. This functionality, coupled with call state control, coordinates the activities of the elements in a VoIP network.

·         CALL CONNECT AND DISCONNECT (BEARER CONTROL)

The connection of a call is made by two endpoints opening communications sessions between each other. In the PSTN, the public or private switch connects logical DS-0 channels through the network to complete the calls. In a VoIP implementation, this connection is a multimedia stream (audio, video, or both) transported in real time. This connection is the bearer channel and represents the voice or video content being delivered. When communication is complete, the IP sessions are released and optionally network resources are freed.

·         CODEC OPERATIONS

Voice communication is analog, while data networking is digital. The process of converting analog waveforms to digital information is done with a coder-decoder (CODEC, which is also known as a voice coder-decoder [VOCODER]). There are many ways an analog voice signal can be transformed, all of which are governed by various standards. Most of the conversions are base on pulse coded modulation (PCM) or variations.

In addition to performing the analog to digital conversion, CODECs compress the data stream, and provide echo cancellation. The bandwidth savings for the voice services can come in several forms and work at different levels. For example, analog compression can be part of the encoding scheme (algorithm) and does not need further digital compression from the higher working layers of the media gateway application. Another way to save bandwidth is the use of silence suppression, which is the process of not sending voice packets between the gaps in human conversations. Using compression or silence suppression can result in sizable bandwidth savings. However, there are some applications that could be adversely affected by compression. Compression schemes can interfere with the functioning of modems by confusing the constellation encoding used. The result could be modems that never synchronize or modems that exhibit very poor throughput. Some gateways might implement some intelligence that can detect modem usage and disable compression.

How VoIP Works

VoIP services convert voice into a digital signal that travels over the Internet. If calling a regular phone number, the signal is converted to a regular telephone signal before it reaches the destination. VoIP can allow making a call directly from a computer, a special VoIP phone, or a traditional phone connected to a special adapter. In addition, wireless “hot spots” in locations such as airports, parks, and cafes allow you to connect to the Internet and may enable to use VoIP service wirelessly.

VoIP APPLICATIONS

VoIP could be applied to almost any voice communications requirement, ranging from a simple inter-office intercom to complex multi-point teleconferencing/shared screen environments, passing through applications like Net2Phone and voice mails. Some examples of VoIP applications are:

• Remote access from a branch office: A small office could gain access to corporate voice, data, and facsimile services using the company’s Intranet. This may be useful for home-based agents working in a call centre. An example can be a bank that wants to reduce costs and combine traffic to provide voice and data access to the main office. This is accomplished by using a packet network to provide standard data transmission while at the same time enhancing it to carry voice traffic along with the data.
• Internet-aware telephones: Ordinary telephones can be enhanced to serve as an Internet access device as well as providing normal telephony, using VoIP technology. The Internet is so rich in resources and the telephone is relatively poor, though that much popular and useful. Using VoIP, directory services, for example, could be accessed over the Internet by submitting a name and receiving a voice reply. The telephone can be used to query database for any information, including membership details to communication companies.
• PSTN gateway: Interconnection of the Internet to the PSTN can be accomplished using a gateway either integrated into a PBX (the iPBX) or provided, for example, would have access to the public network by calling a gateway at a point close to the destination (thereby minimizing long distance charges).
• Internet call centre access: Access to call centre facilities via the Internet is emerging as a valuable adjunct to electronic commerce applications. Internet call centre access would enable a customer who has questions about a product being offered over the Internet to access customer service agents online. VoIP can be further be used to interconnect different call centers, thereby coordinating the work between them.
• Voice calls from a mobile PC via the Internet: Calls to the office can be achieved using a multimedia PC that is connected via the Internet. One example would be using the Internet to call from a hotel instead of using expensive hotel telephones. This could be ideal for submitting or retrieving voice messages. All you need is a PC (laptop), equipped with the necessary hardware and software to allow for VoIP application.

Voice over Internet Protocol (VoIP) is a technology that allows making voice calls using a broadband Internet connection instead of a regular (or analog) phone line. Some VoIP services may only allow calling other people using the same service, but others may allow calling anyone who has a telephone number - including local, long distance, mobile, and international numbers. Also, while some VoIP services only work over computer or a special VoIP phone, other services allow you to use a traditional phone connected to a VoIP adapter.

MPLS stands for Multiprotocol Label Switching. MPLS gives network operators a great deal of flexibility to divert and route traffic around link failures, congestion, and bottlenecks. Multi Protocol Label Switching (MPLS) is a data-carrying mechanism that belongs to the family of packet-switched networks. MPLS operates at an OSI Model layer that is generally considered to lie between traditional definitions of Layer 2 (Data Link Layer) and Layer 3 (Network Layer), and thus is often referred to as a “Layer 2.5″ protocol. It was designed to provide a unified data-carrying service for both circuit-based clients and packet-switching clients which provide a datagram service model. It can be used to carry many different kinds of traffic, including IP packets, as well as native ATM, SONET, and Ethernet frames.

A number of different technologies were previously deployed with essentially identical goals, such as frame relay and ATM. MPLS technologies have evolved with the strengths and weaknesses of ATM in mind. Many network engineers agree that ATM should be replaced with a protocol that requires less overhead, while providing connection-oriented services for variable-length frames. MPLS is currently replacing some of these technologies in the marketplace. It is highly possible that MPLS will completely replace these technologies in the future, thus aligning these technologies with current and future technology needs.

MPLS was originally proposed by a group of engineers from Ipsilon Networks, but their “IP Switching” technology, which was defined only to work over ATM, did not achieve market dominance. Cisco Systems, Inc. introduced a related proposal, not restricted to ATM transmission, called “Tag Switching”. It was a Cisco proprietary proposal, and was renamed “Label Switching”. It was handed over to the IETF for open standardization. The IETF work involved proposals from other vendors, and development of a consensus protocol that combined features from several vendors’ work. MPLS works by prefixing packets with an MPLS header, containing one or more ‘labels’. This is called a label stack. Each label stack entry contains four fields:

• 20-bit label value.
• 3-bit field for QoS (Quality of Service) priority
• 1-bit bottom of stack flag. If this is set, it signifies that current label is the last in the
stack
• 8-bit TTL (time to live) field.

These MPLS-labeled packets are switched after a Label Lookup/Switch instead of a lookup into the IP table. When MPLS was conceived, Label Lookup and Label Switching were faster than a RIB lookup because they could take place directly within the switched fabric and not the CPU.
The entry and exit points of an MPLS network are called Label Edge Routers (LER), which, respectively, push an MPLS label onto the incoming packet and pop it off the outgoing packet. Routers that perform routing based only on the label are called Label Switch Routers (LSR). Labels are distributed between LERs and LSRs using the “Label Distribution Protocol” (LDP). Label Switch Routers in an MPLS network regularly exchange label and reach ability information with each other using standardized procedures in order to build a complete picture of the network they can then use to forward packets. Label Switch Paths (LSPs) are established by the network operator for a variety of purposes, such as to create network-based IP Virtual Private Networks or to route traffic along specified paths through the network. In many respects, LSPs are no different than PVCs in ATM or Frame Relay networks, except that they are not dependent on a particular Layer 2 technology.

In the specific context of an MPLS-based Virtual Private Network (VPN), LSRs that function as ingress and/or egress routers to the VPN are often called PE (Provider Edge) routers. Devices that function only as transit routers are similarly called P (Provider) routers. The job of a P router is significantly easier than that of a PE router, so they can be less complex and may be more dependable because of this. When an unlabeled packet enters the ingress router and needs to be passed on to an MPLS tunnel, the router first determines the forwarding equivalence class (FEC) the packet should be in, and then inserts one or more labels in the packet’s newly-created MPLS header. The packet is then passed on to the next hop router for this tunnel. When a labeled packet is received by an MPLS router, the topmost label is examined. Based on the contents of the label a swap, push (impose) or pop (dispose) operation can be performed on the packet’s label stack. Routers can have prebuilt lookup tables that tell them which kind of operation to do based on the topmost label of the incoming packet so they can process the packet very quickly.

In a swap operation the label is swapped with a new label, and the packet is forwarded along the path associated with the new label.

In a push operation a new label is pushed on top of the existing label, effectively “encapsulating” the packet in another layer of MPLS. This allows hierarchical routing of MPLS packets. Notably, this is used by MPLS VPNs.

In a pop operation the label is removed from the packet, which may reveal an inner label below. This process is called “de-capsulation”. If the popped label was the last on the label stack, the packet “leaves” the MPLS tunnel. This is usually done by the egress router. During these operations, the contents of the packet below the MPLS Label stack are not examined. Indeed transit routers typically need only to examine the topmost label on the stack. The forwarding of the packet is done based on the contents of the labels, which allows “protocol-independent packet forwarding” that does not need to look at a protocol-dependent routing table and avoids the expensive IP longest prefix match at each hop.

At the egress router, when the last label has been popped, only the payload remains. This can be an IP packet, or any of a number of other kinds of payload packet. The egress router must therefore have routing information for the packet’s payload, since it must forward it without the help of label lookup tables. An MPLS transit router has no such requirement.

MPLS versus IP

MPLS cannot be compared to IP as a separate entity because it works in conjunction with IP and IP’s IGP routing protocols. MPLS gives IP networks simple traffic engineering, the ability to transport Layer 3 (IP) VPNs with overlapping address spaces, and support for Layer 2 pseudo wires. Routers with programmable CPUs and without LSP can either be

(a) Explicitly configured hop by hop,
(b) Dynamically routed by the Constrained Shortest Path First CSPF algorithm,
(c) Configured as a loose route that avoids a particular IP or that is partly explicit and partly dynamic.

In a pure IP network, the shortest path to a destination is chosen even when it becomes more congested. Meanwhile, in an IP network with MPLS Traffic Engineering CSPF routing, constraints bandwidth of the traversed links can also be considered, such that the shortest path with available bandwidth will be chosen.

MPLS versus Frame Relay

Frame relay aimed to make more efficient use of existing physical resources, which allow for the under provisioning of data services by telecommunications companies to their customers, as clients were unlikely to be utilizing a data service 100 percent of the time. Telcos often sell frame relay to businesses looking for a cheaper alternative to dedicated lines. Many customers are likely to migrate from frame relay to MPLS over IP or Ethernet within the next two years, which in many cases will reduce costs and improve manageability and performance of their wide area networks.

MPLS versus ATM

Both MPLS and ATM provide a connection-oriented service for transporting data across computer networks. In both technologies, connections are signaled between endpoints, connection state is maintained at each node in the path, and encapsulation techniques are used to carry data across the connection.

The most significant difference is in the transport and encapsulation methods. MPLS is able to work with variable length packets while ATM transports fixed-length (53 byte) cells. Packets must be segmented, transported and re-assembled over an ATM network. MPLS simply adds a label to the head of each packet and transmits it on the network.
An MPLS connection (LSP) is unidirectional. ATM point-to-point connections (Virtual Circuits), on the other hand, are bi-directional, allowing data to flow in both directions over the same path. Both ATM and MPLS support tunneling of connections inside connections.

The biggest single advantage that MPLS has over ATM is that it was designed from the start to be complementary to IP. Modern routers are able to support both MPLS and IP natively across a common interface allowing network operators great flexibility in network design and operation. ATM’s incompatibilities with IP require complex adaptation, making it comparatively less suitable for today’s predominantly IP networks.

MPLS provides networks with a more efficient way to manage applications and move information between locations. With the convergence of voice, video and data applications, business networks face increasing traffic demands. MPLS enables class of service (CoS) tagging and prioritization of network traffic, so administrators may specify which applications should move across the network ahead of others. This function makes an MPLS network especially important to firms that need to ensure the performance of low-latency applications such as VoIP and their other business-critical functions.

Signaling System Number 7 (SS7) is a set of telephony signaling protocols which are used to set up most of the world’s public switched telephone network telephone calls. The main purpose is to set up and tear down telephone calls. Other uses include number translation, prepaid billing mechanisms, short message service (SMS), and a variety of other mass market services

There are two essential components to all telephone calls. The first, and most obvious, is the actual content—our voices, faxes, modem data, etc. The second is the information that instructs telephone exchanges to establish connections and route content to an appropriate destination. SS7 is designed to operate in two modes:

• Associated Mode
• Quasi-Associated Mode

When operating in the Associated Mode, SS7 signaling progresses from switch to switch through the PSTN following the same path as the associated facilities that carry the telephone call. This mode is more economical for small networks.

When operating in the Quasi-Associated Mode, SS7 signaling progresses from the originating switch to the terminating switch following a path through a separate SS7 signaling network composed of STPs. This mode is more economical for large networks with lightly loaded signaling links.

SS7 clearly splits the signaling planes and voice circuits. An SS7 network has to be made up of SS7-capable equipment from end to end in order to provide its full functionality. The network is made up of several link types (A, B, C, D, E, and F) and three signaling nodes - Service switching point (SSPs),
- Signal transfer point (STPs),
- Service Control Point (SCPs).

Each node is identified on the network by a number, a point code. Extended services are provided by a database interface at the SCP level using the SS7 network. The links between nodes are full-duplex 56, 64, 1,536, or 1,984 kbit/s graded communications channels.

Route Management

This function provides a means for rerouting traffic around failed or congested nodes. Route management works together with link management. Route management informs other nodes of the status of the affected node. It uses Message Signal Units (MSUs) generated by adjacent nodes and is not usually generated by the affected nodes.

Traffic Management

This function provides flow control if a node has become congested. It allows the network to control the flow of certain messages based on protocol. Traffic management deals with a specific user part within an affected node.

Message Routing

Message discrimination will pass messages to message routing if it determines the message is not local. Message routing reads the called and calling party addresses to determine the physical address in the form of a point code. Every SS7 node must have its own unique point code. Message routing determines the point code from an address contained in the routing table.

Message Transfer Part

Protocols are used within the layers of the SS7 protocol to accomplish functions called for at each level. Levels 1, 2 and 3 are combined into one part, the Message Transfer Part (MTP). MTP provides the rest of the levels with node-to-node transmission, including basic error detection and correction schemes and message sequencing. It provides routing, message discrimination and distribution functions within a node.

SS7 OVER IP
• Integrate IP-based nodes into the SS7 network.
• No special hardware requirements for the IP-based nodes.
• Interworking at different protocol layers.
• A common transport protocol is used.
• Similar performance requirements as the classical SS7 network
• Minimize end-to-end delay.
• Short failover time in case of network failures.

SS7 can be transported over IP. Ss7 is fast, accurate and secure protocols system for the network .it has wide range of applications

To carry a demand between two nodes on a SONET ring, traffic is routed simultaneously clockwise and counter-clockwise, one as the primary path and the other as the backup path. The master ring problem (MRP) is to find such a ring, whenever it exists. The goal is to find a master ring whenever it exists. As a network evolves with growing traffic, it expands from an initially small number of SONET rings. As a result, the network may have unnecessarily complex topology. To replace a spaghetti-like network, one simple topology is a master ring. Another problem is to perform load balancing using a linear programming based formulation in SONET dual ring.

There are different topologies of SONET that enables a number of different networks to solve networking requirements, including survivability, cost, and bandwidth efficiencies. The SONET configurations include:

• Point-to-point configuration
• Hub configuration
• Linear Add/Drop configuration
• Ring configuration

SONET technology has made the ring a popular network topology. To carry a demand between two nodes on a SONET ring, traffic is routed simultaneously clockwise and counter-clockwise, one as the primary path and the other as the backup path. Often an optical network consists of a collection of interconnected SONET rings. A master ring contains every node in the network exactly once and respects the node ordering of every individual SONET ring. The master ring problem (MRP) is to find such a ring, whenever it exists.

Management & protection techniques in the optical networks are becoming more and more important with increasing demand for the availability of high-speed networks. A master ring respects the node ordering of every existing SONET ring; it has the advantage of preserving the routing label of every demand intra to an existing SONET ring. By reconfiguring the network, overall utilization can be improved. Ring topology has advantages over mesh but rings are considered for poor routing so the reconfiguring rings may increase the efficiency of the network. By combining highly accurate network synchronization systems with advanced optical network technology, high-speed transport systems like SDH or SONET.

Reduction of cross talk

In WDM there may be a problem of cross talk because the signals may mix up while travelling from the window. The work is being done to increase the number of multiplexed channels, by decreasing the channel spacing, and increasing the bit (data) rate of a single channel reduce the crosstalk in WDM optical communication systems. Both these factors, decrease in the channel spacing and increase in the data rate, increases the crosstalk of the systems.

WDM consists of Ring topology. Once the WDM ring is up, light paths need to be established in accordance with the traffic pattern to be supported. This helps to solve the problem of routing and wavelength assignment. Essentially, the light path routes need to be determined. WDM is employed by carriers to boost the data rates of their networks dramatically. The frequency increases as the wavelength is shortened. The WDM promises of unlimited bandwidth and fast protection capabilities. The deployment of WDM devices thus must be economically well justified, as well as well planned.

METHOD & SOLUTION

To reduce cross talk the authors are using return-to-zero (RZ) format in place of NRZ format. The RZ format is more advantageous, because the RZ-modulated signal can withstand better the impact of fiber nonlinearity and polarization-mode dispersion. The most widely used being is the polarization interleaving method. In dispersion interleaving method, the signals in the odd channels are delayed by a half-bit period relative to the signals in the even channels. Thus, the interference from the adjacent channels near sampling point is greatly reduced.

In Polarization & Dispersion Interleaved WDM systems the total channels (N) are separated into two odd and even channels in a single stage and then separated odd and even channels are multiplexed separately. In polarization interleaving model the separation of channels into odd and even channels improves capacity and spectral efficiency of WDM systems. So modifications in PI and DI systems are proposed. Separation of total number of channels into odd and even channels is done in several stages instead of a single stage. In first stage, N channels are divided into two odd and even channels of number N/2.  In second stage, each N/2 channel is again divided into two odd and even channels of number N/4. This process is continued till the divided odd and even channels have only one number. The proposed design is for eight channels (N = 8). The odd and even channels are routed through optical polarizer’s having vertical and horizontal polarizations, respectively, before interleaving with each other.

 The fiber channel is used to serve both networks and channels.

FIBER CHANNEL DESIGNS:

1.      Point -to- point: It is used for the transmission of large blocks of data and also as a mass data storage device.

 2.      Switched fabric:It is used to cluster a number of devices through a switching fabric.

3.      Attributed loop: It is also known as FC-AL. it is used to connect three or more devices without fabric. It allows the connection of several devices in a ring forming a “virtual single device.”

CLASS OF SERVICES:Fiber channel has three classes to satisfy the large number of data communication.

Class 1: it is based on circuit switched or hard switched connection. It is a dedicated link that connects two devices and is capable of providing a continuous and graduate delivery of data with acknowledgement of receipt.

Class 2: this connection is based on frame switched service. It is not a dedicated connection. It is not a dedicated connection. In frame switching mechanism reads the switching mechanism reads the frame header code, and then makes a decision for the pay load destination. It also provides guaranteed delivery with acknowledgement of receipt.

Class 3: it is data gram transfer. This service has no connection and that allows data to be transmitted very quickly to a number of devices. Class 3 does not provide acknowledgement of receipt.

EFFICIENCY:Fiber channel is based on IBM 8B/10 B scheme and was specifically designed for data transmission over optical fiber.

FIBER CHANNEL STANDARDS: The set of standards for fiber channel are as follow:

1. The fiber channel’s physical standards. It is subdivided into five sub levels FC-0 to FC-4. FC-0 performs within physical layer.FC-1 is a transmission protocol. Fc-2 is signaling protocol. FC-3 is a common service. FC-4 is a bus interface.

2. The fiber channel arbitrated loop standard (FC-AL)

THE FIBER CHANNEL TRANSCIEVER:

At the transmitter input, the applied 10 bit parallel TTL data is encoded into 8B/10 B format and then multiplexed into high speed serial data stream. The PLL circuit in the transmitter section locks onto the 106.25 MHz clock frequency supplied by the user, then multiples it by ten to generate the required clock signal for high speed serial output signal.At the receiver end, the PLL circuit locks onto the incoming 1.0625 GBd serial data, from which it recovers both the data and clock signals. It also recovers two 53.125 MHz bytes clock signals with a phase difference of 180 degree which is used to align the parallel data at the positive going transition.

DETAILED OPERATION OF FIBER CHANNEL:

The transceiver is capable of transmitting and receiving 10 bit parallel data over high speed line in accordance with the fiber channel standards for FC-0 layer specifications. The transceiver incorporates the following:

• TTL parallel inputs/outputs (I/Os)
• High speed PPL circuits (transmitter receiver)
• High speed serial clock and data recovery circuitry
• Parallel-to-serial converters
• Comma character recognition circuitry
• Byte alignment circuitry
• Serial to paralle conversion circuitry

The functions are accomplished by building block within transceiver blocks. These are:

Transmitter:

• Input latch
• PLL/clock generator
• Frame multiplexer
• Output select

Receiver:

• Input select
• PLL/clock recovery
• Input sampler
• Frame de multiplexer and byte synchronization
• Output drivers
• Signal detect

Ray tracing is a method for calculating the path of waves or particles through a system with regions of varying propagation velocity, absorption characteristics, and reflecting surfaces. Under these circumstances, wave fronts may bend, change direction, or reflect off surfaces, complicating analysis. Ray tracing solves the problem by repeatedly advancing idealized narrow beams called rays .

Technique:

Ray tracing works by assuming that the particle or wave can be modeled as a large number of very narrow beams and that exists at some distance over which such a ray is locally straight. The ray tracer will advance the ray over this distance, and then use a local derivative of the medium to calculate the ray’s new direction. From this location, a new ray is sent out and the process is repeated until a complete path is generated. If the simulation includes solid objects, the ray may be tested for intersection with them at each step, making adjustments to the ray’s direction if a collision is found.

Algorithm
The basic ray tracing algorithm is called a “recursive” algorithm. Recursion is a means of obtaining a result in which a given process repeats itself an arbitrary number of times. Infinite recursion is recursion that never ends, and this is almost never useful. The algorithm begins by shooting a ray from the eye and through the screen, determining all the objects that intersect the ray, and finding the nearest of those intersections. It then repeats itself by shooting more rays from the point of intersection to see what objects are reflected at that point, what objects may be seen through the object at that point, which light sources are directly visible from that point, and so on.

1) Reflection
If the surface that the ray intersected was reflective, like a mirror, the ray tracer must determine the color at that point by finding, not only the color of the surface, but also the color of the reflection of any objects at that point.

2) Transparency
Transparency is modeled similarly to reflection, but instead of bouncing the new ray off of the surface, the ray is bent into and through the surface to model refraction. Refraction is an optical phenomenon caused when light bends as it travels through a given substance.

3) Shadows
Shadows are the third standard feature in ray tracing. Imagine the surface of an object. If seeing the light source, then there’s a clear path between you and the light, and at least some photons will certainly travel along this path. If any opaque objects are in way, then no light is coming directly from the light into your eye, and you are in shadow with respect to that light.

Methods:
Ray tracing with mirrors
Ray tracing with thin lenses

Applications:
a) Radio signals
b) Ocean acoustics
c) Optical design