A network often simply referred to as computer network, is a collection of computers and devices connected by communications channels. Computers on a network are sometimes called nodes. In information technology network is series of points or nodes interconnected by communication paths. These communication channels facilitate communications among users and allow users to share resources with other users.
Networks may be classified according to a wide variety of characteristics.
The geometric arrangement of a computer system. The most common topology or general configurations of networks include the:
Token ring topology
Networks can also be characterized in terms of spatial distance: A group of two or more computer systems linked together. There are many types of computer networks, including:
Local-area networks (LANs): In this type of network the computers linked
within the organization.
Wide-area networks (WANs) : The computers are farther apart and are connected by different communication channels.
Metropolitan-area networks MANs): A data network designed for a town or city.
Campus-area networks (CANs): The computers are within a limited geographic area.
Home-area networks (HANs): A network contained within a user’s home that connects a person’s digital devices.
Before going to Internet protocols first we have to know about the protocol.
The protocol defines a common set of rules and signals that computers on the network use to communicate.
The Internet Protocol (IP) is a protocol used for communicating data across a packet-switched internetwork using the Internet Protocol Suite, also referred to as TCP/IP.
The Internet Protocol is responsible for addressing hosts and routing datagram (packets) from a source host to the destination host across one or more IP networks. The sending of packets from source to destination is performed based on two types of networks:
These networks include:
A wired network is a physical system that allows multiple computers communicate with each other. Wired networking is usually accomplished by using cables and other hardware to connect one computer to another. Once a wired home network has been installed, users within the network can share computer files and documents as well as printers and scanners. In This network the packets are transferred by using TCP/IP.
Connecting one computer to another computer is called network. We are connecting these computers to transfer packets from source to destination. But we are using wireless network, instead of cables we are using radio waves to transfer these packets by using UDP (User Datagram packet).
Internet protocol suite which standardized for the entire networking environment describes specific rule guidelines for communication strategy. This protocol architecture deals with communication between source and destination which means how the data should be formatted, how the packets should identify the address of the receiver and how the packets are transmitted from source to destination.
In this TCP/IP architecture the data should be transferred in the form of packets from source to destination. The transmission of packets starts from source and these packets transmits through different types of layers and finally it reaches the destination. In this architecture there are 4different layers, each layer performs its own functionality to reach destination.
In this architecture each layer sends the information to the lower layer to process the functionality of that layer, after completion of processing the information passed to lower layer and so on. This process repeats till the packet should reach the destination.
The 4 different layers of TCP/IP protocol architecture and each layer is described below:
This layer provides the some functionality to perform different services of the lower layers. And this layer provides different protocols for the purpose of exchanging data between different layers. The protocols which are defined by application layer are known as application protocols and these are used to exchange the information:
There are different protocols used in application layer are described as follows:
The Hypertext Transfer Protocol (HTTP): This protocol used in application layer which is used to perform the transmission of files among the network.
The File Transfer Protocol (FTP): It is also one protocol used in application layer to perform the transmission of interactive files.
The Simple Mail Transfer Protocol (SMTP) by using this protocol the transmission of mail messages and file attachments are done.
The functionality of application layer similar to the functionalities of OSI Application, Presentation Layer and Session Layer.
This is the second layer of TCP/IP protocol architecture. This Transport layer deals with Transmission Control Protocol (TCP).This protocol is used to know the status of packet transmission whether it reached the exact destination or not. Before reaching the destination it performs the error checking which means it is in the right order or not.
This is the one of the important layer of TCP/IP protocol architecture. In this layer it deals with the Internet protocol (IP).The communication of these layers should be done by using this protocol.
Network Access Layer
The network access layer is the combination of Data link layer and physical layer. The Data link layer deals with the transmission of data packets across the architecture. This Data link layer provides different functionalities like formatting the data in packets, error detection and the flow of data from source to destination. And coming to physical layer deals with the physical medium such as voltage levels, connector types and handshake procedures.
TCP/IP Protocol Layers
Chapter – 2
In chapter 1, I have mainly discussed about the VoIP and brief structure of dissertation. This chapter helps in understanding the VoIP system, key components, and protocols. It is good to understand the concepts and the technology when working on a new system.
The generations have changed, so the way of communicating to your family, friends has changed. Not only quality is a concern but low cost is also an important factor for making calls. These are the few things which brought a lot of research work for Voice over internet protocol (VoIP). VoIP is a way sending voice packet over the internet. VoIP not only provides cheap calls but also provides better quality than before.
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2.2 What is VoIP?
VoIP is one of greatest invention in human kind; it basically allows transmission of human voice in the form of digital packet over a network. It is in demand due to lower call rates, implementation on a existing network, better quality than before. Many people, organisations have been consistently working on this technology to get a better performance of the service. This technology could be also implemented in a local area network. The process involves compressing and digitization of voice and transforming into Internet Protocol packets and sending the format over network. This is the process which is known as Voice over Internet Protocol (VoIP). 
2.3 VoIP History
Computer to Computer calling begin in Israel in 1995 by vocaltech Software by name internet phone. This software was designed and used to make Voice calls . The whole system consists of basic things such as speakers, modem, and sound cards. The voice signals were digitized, turned into voice packets, so that they could be travelled over the network and in the end, packets would be shifted back into voice signals. The overall quality was not at all good and not even near traditional phone calls. But still it is considered as a major milestone as it was birth of new technology. The designed software had to be installed on both systems (caller and receiver) to process the voice call. Therefore it was the birth of first IP phone.
Later after few years in 1998, number of private companies, organisation started working and using this technology more effectively. It is valuable assets as it is due to provide cheap long distance calls, avoid high charges with traditional phone system. These companies were able to design and organize VoIP doors on the internet, which would allow cheaper calls through Computer to Computer and also from regular phone to phone. This enhancement brought more awareness of the technology, market profits among private organisation and investors. However on the other side general people were unknown with the technology due non availability of high speed internet connection in most of the areas. This was one of the main reasons that it couldn’t get popular quickly. Enterprises, small organisation already started using the benefits of VoIP; communication with voice across the networks was deployed and used. As time passed more standards and protocols were introduced for better quality as VoIP in beginning stages was not a good quality .
In initially stages when it was introduced, it was not considered as a potential technology. In some parts of North America it was offered free, which allowed free phone calls using regular phone. Though it was not considered potential technology, it was slowly getting popular and accepted in various areas such as enterprises, general public. Finally it was a well known model by 1998, where VoIP was used about 1% of voice traffic and in just two years of time by the year 2000, it was used over 3 % of traffic. This growth was mainly due improvement in the system. . VoIP is rapidly growing year by year. It is likely to reach around 300 million users of VoIP around the world.
A protocol can be defined as a combination or set of rules. These are necessary for data transmission by two or more computers over a network. These protocols determine data compression, acknowledgements, and error checking methods. 
2.4.1 Internet Protocol
Internet protocol is the primary network protocol used across the network. By using this protocol the packets are sent across the network.It is a connectionless protocol which means each packet is travelling in the network is independent and is not related to any other unit. The connection oriented protocol transmission control protocol (TCP) handles all the packets.
The H.323 is one of the two important protocols used in VoIP. H.323 protocol is a standard designed by international telecommunications union (ITU), which sets rules for multimedia applications across the network. It is used for sending audio, data packets, and video over IP network, which basically means providing real time multimedia applications. It behaves as real time protocol and also non real time protocol. .
Most of the ITU-T terminal recommendations provide only terminals. However H.323 provides various components along with terminals on a network . They are gateway, gatekeepers, and multiple control units. These components provide point to point/multipoint multimedia communications, when connected together. Below is the figure of H.323 architecture
Figure: 1 H.323 architecture Ref 
18.104.22.168 H.323 terminal
The H.323 terminal is also known as client; usually it is a computer or internet phone with H.323 client. The H.323 terminal gives real time bidirectional video, voice, data interactions. H.225.0 is used to specify call signalling, packetization, synchronizations. On the other side H.245 is used to specify messages for opening and closing logical connections beside other commands .
22.214.171.124 Gate keeper
Gatekeeper supports address translation and other access control for H.323 standards. H.323 is the standard that points the system; this protocol provides different types of services such as voice, video and packet transmission. A gatekeeper acts as routing manager, which is responsible for all end nodes in a particular area. 
Gateway is a part of the VoIP system, which is used to modify audio and fax call in real time. Mainly it works between internet and plain old telephone system as a converter. This IP telephony gateway converts analogy signal into internet data packets and sends data as a regular data packet over a network. Once the packet is received on the receiver end, it is re converted into analogy voice signal. It allows regular telephone calls to be achieved through the internet and provides communication between Private Branch Exchange (PBX) and internet.
126.96.36.199 Multi Point Control Units
Multipoint Control Unit (MCU) acts as a multipoint conference to establish a connection among three or more devices and gateways. It is usually integrated with a client , gateway. There are two elements associated to MCU, one is multipoint controller and second is multipoint processor. In this multipoint conferencing, two types of stages involved, centralized and decentralized. Centralized multipoint mode provides direct communication of H.323 devices to MCU in P2P method. Multipoint controller is used in a point to point call way and moreover it is also used to check if it has to multi cast or uncast the information such as audio and video. In the other mode decentralized multipoint, multicast method is used to transfer all data to other devices. The multi point control unit checks the readiness of all devices and sends different data streams .
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It is a protocol used for modifying, terminating, establishing sessions over a network. It is also called as signalling protocol and mainly used for internet telephony VoIP calls. It is designed by the internet engineering task force (IETF) for real time transmission. There are different components in SIP.SIP is less weight than H.323 and moreover when it comes to flexibility, it is more flexible . It can also used for video, audio, multimedia sessions. It is more popular and being adopted more due to more flexibility than H.323. The structure of SIP is same as of Hyper Text Transport Protocol (HTTP); it follows the client server model 
188.8.131.52 Main Components involved in SIP
User Agent works as logical entity at SIP end station software. It acts as client when initialising session requests and acts as server when replying to a session request. It has the capabilities of initiating and answering calls. It also has the ability to save and manage call state that is why it is known as intelligent .
They are used as computers and requests are forwarded by them on the behalf of other computers. They are one kind of intermediate server, forwarding requests from the user to SIP server .
Figure: 2 an example of proxy server Ref 
It is a second type of intermediate server. The main responsibilities of this server is to give name resolution and user location. It simply gives the information needed to start the originating client. Once it has been done, it is no more involved in the process 
Registrar Server is a server that is used to register the location information from the user agent and saves the registration information. This sort of concept creates a directory for all logged and also specifies their location.
QOS is Quality of Service refers to the capability of a network to provide better service to selected network traffic over various technologies, including Frame Relay, Asynchronous Transfer Mode (ATM), Ethernet and 802.1 networks, SONET, and IP-routed networks that may use any or all of these underlying technologies. The primary goal of Qos is to provide priority including dedicated bandwidth of service is most important part for any multimedia application. It is more considered in a private network such as connection of multiple offices than on worldwide internet. Initially, internet was known for sending email and file. However it is more than just sending files, now it has real time applications such as video, gaming, internet telephony. As the VoIP technology has evolved so much, there is more need for better Qos.
Below table provides a list of Qos parameters :
For a successful VoIP deployment, Qos is an important tool. Bandwidth is one of important factor in VoIP, if enough bandwidth is provided; the voice quality will be more than just satisfied. This is the reason services in organisation doesn’t need planned QOS as they have enough bandwidth. A successful VoIP deployment requires a careful examination and network assessment, as well as a system to measure and monitor the system in order to benefit from the cost savings, flexibility and functionality VoIP offers.
There are five areas to consider when planning a VoIP deployment:
Make sure the network can handle VoIP.
Keep the deployment simple.
Create network service maps and update service-level agreements.
Consider QoE. Quality of experience (QoE)
. Review, reassess and repeat.
However as the network grows such as internet, more complications, issues arise as there are multiple services to be handled, several area’s are to be covered, security is concerned. The aim is quite simple, IP telephony must meet the requirements of a user, which means voice quality should be similar to that of PSTN in such a way to retain same standards and satisfy the user expectations. When the voice signals are transmitted over the network, the data is divided into packets and transmitted across the network through network elements such as routers, switches. In this process to achieve better results, Qos is to be followed which would give importance to real time application over non real time applications.. The main factors network analysis are mainly delay, jitter (variation in delay), and packet loss.
One of the very important factors in VoIP Qos is packet loss. This generally occurs when the packet is dropped during peak loads and congestion. This happens as VoIP packets are very sensitive of voice transmission. The main reason or largest cause of a packet loss is a packed being dropped due to overload in the network. This is the main factor which can significantly affect VoIP quality 
Delay is the key performance of QoS , which is needed to be reduced . End to end delay is the duration taken by a analog signal to transmit from the sender to receiver. Delay does not produce noise , however it affects the quality of conversation. When the delay reaches about 250ms, the telephonic conversation observes its impact. Between 300 ms and 500 ms the talk gets slightly difficult. Finally if it reaches above 500 ms makes an impossible communication. In VoIP the delay could be anywhere between 50 to 100 ms , however in POTS , the delay is generally under 10 ms.
Delay Variation sometimes also known as Jitter is the variation in packet delay. It is mainly the difference value between the delay of two consequent voice packets. In the internet terminology they can explained in a simple manner , if the packet n has delayed highly then chances are high for the packet n+1 also. To reduce the delay between two queuing packets, a jitter buffer is used. This buffer stores arriving packets temporarily in order to reduce the difference value between two packets. For some reason, if the packets are delayed then it simply discards the packet. The acceptable value considered for jitter is between 0ms and 50ms and anything above this is not acceptable  done
Techniques for jitter absorption
1. same playout time is set for all packets for whole session or for the duration of one
2. The second technique involves adaptive adjustment of the playout time during silence period related to present network. This technique is also called as talkspurts adjustment.
3. This technique is also known as within talkspurts adjustment. In this method constantly adapting the playout time for each packet, which needs the scaling of voice packets 
2.6 Initial VoIP Study and Design
There are various real time applications running on internet, such as calling over PC, Video and voice. When deploying these services across networks, there are many difficulties and issues are faced. Some of them are directly related to QoS ; is the network capable of handling the new application and whether there will be any negative impact on existing services in the network. For real time application such as VoIP, which uses UDP as transport layer protocol, QoS is not guaranteed. Researchers are always working on these issues depending upon the real time application. In order to achieve good results, a good policy , QoS should be implemented. There are many things to be considered to give a high standard service. To begin packetization , delay , network design are few things to establish QoS. However other things are signalling protocol , security, bandwidth, power failures. Certainly there are numerous methods available to achieve QoS. But it is really important to understand and estimate the requirement of this QoS by a particular system. Sometimes it is noticed by the engineers that spending on the network parameters is less than on QoS. As a result, it is important thing to consider before deploying, whether to design a QoS or not.
2.7 Components of VoIP
Below figure gives an end to end VoIP system Components from sending point to receiving point. As seen in the figure the different components involved are encoder, packetizer, playback buffer, depacketizer , decoder. Voice is in the form of analog , which moves along the time , which usually is limited within 4kHz. Bandwidth . The analog or audio signal is first converted into samples , which is a digital signal , which is then transmitted across the network . Next to encoder it is packetizer , which encapsulates the digital samples into packets along with real time protocol (RTP),User Datagram Protocol(UDP),IP and Ethernet headers. Finally the digital packet moves into the network and passes through playback buffer. Playback buffer mainly used to absorb any kind of changes, delay. In final stages the packet enters depacketizer and then decoder which convert the digital signal into analog signal.
Fig 3: source Assessing the Quality of Voice Communications Over Internet Backbones [17 ].
A codec is software that converts analog signal into digital bits and transmits them as data across the network. These Codecs use high level methods for coding .
Table 2 :
2.9 Performance related QoS parameters.
One of the most important metrics for QoS in VoIP is bandwidth. It plays an important role as the voice channels rely on the present bandwidth. Apart from this constraint, there are other important factors which also decide the quality of calls; they are loss of packets, delay and other variations.
End to End delay
It can be explained as the duration taken by the information to pass from the speaker or caller to the receiver. VoIP requires real time traffic transmission across the network and this transmission end-to-end delay is most important factor to decide QoS. ITU-T suggests that end-to-end delay should be less than 150ms for a high quality conversation 
The bandwidth required to make a single call in one way is 64kbps. If the call is using G.711, the codec samples 20ms of voice for each packet. We need 50 packets to send per second. In addition to this packet, other headers of protocols are also added. They are real time protocol, user datagram protocol, internet protocol and Ethernet. In the end 226 bytes must be transmitted 50 times per sec. This value is only for one way. The total required bandwidth for each call is 100pps for both directions .
One of the very important factor before planning or designing is the calculation of internal and external calls. The call usage statistics plays an important role in VoIP deployment. The statistics would give a clear picture of nature of call, time, duration, peak time. In order to achieve best QoS , always a worst scenario should be considered. Like, busiest call transmitting time should be considered of the month. This would make sure that all other calls are provided with well planned QoS.
2.10 Advantages and Disadvantages
There are many advantages associated with VoIP, especially when compared to PSTN it gives various benefits. The good thing about this technology is that it is not only beneficial to small and large enterprises but also for residential users.
Most of the VoIP companies provide low calling cost for distant places, which are almost same as local call. Only condition is that the user must have broadband connection to use the VoIP service. Now a day’s most of the places such as homes, offices are equipped with internet connection. This is the main advantage with VoIP technology, as it does not need a completely new system to make cheaper calls. In some situations the calling cost will be absolutely free, even though the call will be an international call. Let’s take the scenario where the user is using his or her computer connected with internet. In this process the user just needs software installed on the computer and also at the receiver end, software must be installed. The entire conversation between these two computers is free of cost.
Apart from regular international calls VoIP service also provides unified messaging, real time applications such as audio and video conferencing, e-learning and other entertainment applications. VoIP has brought number of new things in the internet world.
VoIP, or Voice over Internet Protocol, is rapidly becoming a top choice for people wishing to avoid costly telephone service. For this reason it became more popular. The following are three top reasons for using VoIP. The reasons are
It’s Not Going Anywhere
VoIP calling can significantly remove all international charges for companies making distant calls overseas. These savings can be also included when calls are made within the organisations. But when it comes to international calling more savings are achieved. Especially for companies with data networks have an edge over the technology. With recent improvement in VoIP quality, call can be compared to PSTN and used as regular method of calling.
However, there are still some disadvantages of VOIP – especially when it comes to using the technology for functions beyond the one caller to one caller scenario. one of the disadvantage is If multiple users need to make VOIP calls, it can be difficult for the company to know exactly how much bandwidth to provide – especially if internet access, video conferencing or other data transmission services are using the same path.
A high speed internet service is necessary to run VoIP service; it is applicable for both enterprises and residential. If the internet speed is low, then quality of VoIP may degrade.
One of the major drawbacks of VoIP is disconnection of service in power outage. Whereas PSTN works even when there is power outage. Comparatively PSTN is more reliable in such situation. However this situation can be controlled in VoIP by using generators to provide power.
In order to provide more security the usage of VOIP network deployment is increased. Because of low price calls most of the enterprises uses network, it causes network congestion. To avoid network congestion we preferred VOIP network deployment system. After many researches, VOIP is issued in both wire and wireless. The demand of VOIP service has increased because of its security, performance, quality of service (QOS) and network infrastructure enhancement. This chapter mainly focused on research papers on VOIP deployment and present reviews of performance and quality of service (QOS) measurements.
3.1 VoIP Network Securities
In general there are always problem with network attacks like denial of service, worms, network attacks which can be made by man and etc. similarly VOIP system can also faces the attacking problems and degrade the performance of voice services. Here we need to prefer intruder detection system which can handle the network oriented attacks efficiently. Even we will get attacks like spams on the mailing system. The main aim of denial of service is to interrupt the services from resources by making them unavailable to the user.
Multi-Layered VoIP Security
Because of threats attacks the quality of voice will be degrade, it causes the service useless. Denial of service and distributed denial of service attaches degrades the quality of voice.
VoIP service introducing the greatest achievement by providing security in communication by control (signaling) and content (voice) parallel channels which together form the VoIP service. The attacks damage the whole service useless.
3.2 VoIP network Performance
In communication world , a lot of research work is going on to evaluate VoIP performance by analyzing different inputs such as traffic, packet delivery time, receiving time, jitter and network congestion. In the reference , the author designed a network model using computer simulation. This research explains the network performance in respect to various available links and network.
The VoIP performance review shows the loss patterns in terms of packet loss, delay time and network congestion, it measures the delay and loss characteristics. Kostas  deliberated the VoIP performance on different networks and they discuss the delay and loss measurement characteristics, they discuss founded on round trip delay measurements. The quality results can be seen with the delay and packet loss.
3.3 VoIP Quality of Services
VoIP services are gradually adapting in to new level of services and used by different sectors. Currently there are some challenges for this technology. Anyone providing VoIP services should be able to achieve certain standards and quality of the service should satisfy the user. The availability of QoS is an important factor for the end users. Therefore understanding the issues related to QoS is one of the important factor , QoS parameters should be properly analysed and if any loop hole is found , it should be accordingly rectified. Cole et al.  propose a quality measurement method by studying the review on E-Model  by measuring the quintiles of delay and loss. Markopoulou et al.  employs subject quality measures to review internet backbone ability, this work is based on packet tracer which are able to present overall quality of VoIP. In this
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