HISTRY In the history of VoIP we gain information regarding both the development of phones as well as the development process

of the internet. Alexander gram bell introduced the first PSTN phones in the history of the world in the year 1876. Those phones were not made available locally but were leased to subscribers. The switching system was developed in 1889 and was only used by the soldiers and relays. The great mathematics professor Dr. Claude propounded a theory Mathematical theory regarding communication named A mathematical theory of communication which later on was used to develop touch tone dialing system by AT&T which was easier and friendlier and phone operated at home. US protection developed an internet system which was almost unbroken in any circumstances in the year 1957 and this web system was highly marketed by ARPANET in the year 1968. In early 1970s all the progress of the time-share of the computer was developed. After then only firms providing online services were popular and they started email services and bestowing proprietary. After the evolution of web service the concept gradually get developed in the concept of HTTP, URL and WWW in the year 18989. The credit of development of TCP/IP protocol goes to Dr. Vent Cerf which stood as a technique of travelling of the data packets as well as methods over IP networks. Here TCP/IP stands for Transmission manipulation protocol/Internet Protocol. At the early period after the development of the internet the transmission of audio in the packet form was only possible on manipulated environment only which occurred due to the insufficient tools for the compression of the audio below 64 kbps or 54kbps. No input or output devices for the sound transmission was developed. As the compression of audio was made possible in the year 1993 then the concept of the internet phone was invented. A small firm called Vocal Tec Inc. marketed commercially as the concept of VoIP by utilizing the sound preferences and the mobilization of the microphones to make a call via computers but the system only worked on the compute which has same multimedia mechanisms. In the beginning of 1998 the technology of PC-PC and phone to PC calls was widely in people concerns as VOIP. All the calls were made free and the people have to wait a long period listening advertisements after making calls. All the equipments required were provided by the firms namely 1. Nortel, Cisco, Lucent. The high rate of VOIP calls made was 3% and those were the calls made in US. After then it was increased by 3% and 25% in the year 2000 and 2003 simultaneously. The rapid commencement of VoIP gets marketed then. In order to minimize the telecommunication costs the demand of VoIP gets high in every person in the whole world. Then the people found the luxurious VoIP has the defect of poor quality of sound. After then the concept of ITSP was introduced to overcome the poor quality of sound during VoIP calls.

After then the technology of VoIP flourished over all around the globe and the many companies established in order to provide the VoIP services. And different firms started including sound cards in their system. And the reselling companies regarding VoIP were established and they started earning a good commission. The effort of the mankind in both computing and telecommunication as well as in internet for 40 years propounded the establishment of VoIP calls. The VoIP system was popular all over the globe due to its unlimited benefits and services. Objectives of Voice Over Internet Protocol: i) Increment of connectivity: Both voice and data are carried at a time in the particular network in VoIP which results the interconnection as well as intercommunication between all the computers and phones. When we connect our local phones to our PC then the movement ii) Effectiveness of cost: Long distance international calls are comparatively cheaper using VoIP. Also some calls are free of cost as per the scheme of service providers. iii) Maintenance totally simplified: Maintenance is totally simplified as both the computers and telephones networks are integrated in a single network. The arousal of the problem is fixed by a single technician. iv) Best sound quality: Voices as well as videos are clear and faster than other old and outdated phones. Most of the problems that are aroused by PSTN are completely fixed by VoIP. v) Better for distance surfing: Distance calls are better and cheaper than PSTN phones. All the modes are simplified and better quality than PSTN. vi) Additional features: VoIP has been able to provide loads of features and efforts that totally simplifies and attracts business organizations and different home users through different aspects.

Differences between PSTN and VOIP technology PSTN telephone networks are circuit switching technique of switching which uses dedicated end to end connection between two parties. All the required resources during the time period of the session are fully reserved. During the silent periods the circuits are generally idle which results the waste of resources. VoIP uses packet switching methods which are totally different than PSTN which uses as much paths as possible and its resources are not reserved. All the packets are easily switched without reserving any bandwidth in packet switching. VoIP configurations The main thing you should understand in VoIP is there is not only single way of using VoIP calls. 1. Analog telephone adapter Commonly used way Makes an ordinary landline to get connected to the internet in order to use VoIP. Helps in conversion of analog signal into digital signal. Easier to configure for the use of VoIP. In case of some ATA softwares are to be loaded at the host computer which is termed as straight forward set up. 2. Dedicated Routers: Ordinary phones can be converted to VoIP. No software setup required if once configured.

3. Dedicated VoIP phones: Besides ordinary phones there are different specialized phones which come with handset, cradle, and various verities of buttons. These phones come up with RJ-45 connectors which allow to directly get connected to the router. All the required components (software and hardware) are easily available to retain IP calls. Are also capable of operating on wireless networks.

4. Software controlled VoIP calls: Easiest way to manipulate VoIP technology. Arranged for long distance calls. Easy communication between users by the help of hardware devices provided (microphone, sound card, speakers and internet connection i.e. cable/DSL connection).

Features of VoIP phone 1. Online call log: All the incoming as well as outgoing calls logs can be maintained using VoIP technology. Easy calculation of expenditures of all calls epically international calls.

2. Conference calling: Conference calling is also found in the old phones although using VoIP is efficient. Almost up to ten persons can get connected at one conferencing call using VoIP and the cost is comparatively cheaper. 3. Locate Me: Capability of ringing multiple lines simultaneously or sequentially at a time. Ex: both home phone as well as office phone can be ring at same time. Important calls are as far not missed.

4. Do not disturb: Forward of the calls to the specific number at specific time is possible through VoIP. The busy time or the rest time is not disturbed by your calls.

5. Online voicemail: Either mediums phones or the computers can be used for voicemail features. You can access your voicemail from any parts of the using internet.

6. Free in network calling: Cost is absolutely minimized in comparison to other service providers. Long distance calls are also cheaper.

7. Call transfer: VoIP provides the facility of easy transfer of international calls.

8. Area code transfer: Area code selection is necessary while integrating a number for VoIP. Area code is compulsory for grouping the charges further more.

VoIP area code transfer feature is necessary for authenticating business on another part of the city or nation.

9. Ring lists: Multiple lines can be associated with one virtual number. This feature allows ringing simultaneously for one or more numbers sequentially. Most used for the company with multiple numbers.

10. Computer dial: Capacity to store numerous phone numbers. Use of computer in order to dial numbers. Flexible as the dial through the computer rings until the receiver does not pick up the call. 11. Auto attendant: For business purposes with VoIP duration of calls as well as total no of calls must be recorded. Auto attendant feature enables you to organize calls virtually.

12. Call center features: All the calls are managed through call center and the necessary approval of service is also done through call center features. 13. Message alerts: If we miss the calls or we are out of reach because of internet this service is totally useful to notify all the events via message. 14. Integration with outlook: All the data can be assimilated from the PC using VoIP. As per the datas of Microsoft outlook address book calls can be made accordingly. 15. FAX number: The number we have in VoIP can be also integrated to fax line which is called virtual faxing process.

Devices used in VoIP technology The architecture of VoIP is found matching with the architecture of traditional telephones. But the features differ in some ways. In VoIP all the packets of data are sent over IP networks. The main activity of the underlying network is to provide clear connection to all the devices in the network. Traditional telephone devices were restricted to communication between each other where as in VoIP all the standards and protocols must be capable of handling communication and routing features. Some of the commonly used devices in VoIP technology are listed below: 1. End system: Such are machines that are electronic typically used for receiving and dialing calls. Helps in initiating and receiving signals for the use of media transmission. It also keeps the record and track of the calls.

2. Gateways: The activity of calls to be placed to and from is handled by this device.

3. Signaling servers: All the routing of signal in application level is handled by signaling servers. Mostly used to perform location services of user.

Types of VoIP VoIP services depend upon the following factors: The alternative to traditional phones is VoIP services. The process of receiving and transmitting traditional calls. Payment by the user for the services. Equipment must be added by the user. All the traffic routes to or from PSTN.

There are mainly three categories of VoIP which are basically argued below: 1. VoIP via computer: A wide range of voice communication services is provided by VoIP through computer. Such services include person to personal voice calls, push-to-talk, chatting while playing games etc. such communication requires sound card, internet and necessary software components between both parties. Modern service providers such as yahoo, Skype are to be firstly installed and then the communication is set up to other people on the network which uses peer to peer mode of communication. It also has some drawbacks which I have listed below: PSTN service is not accessible via this service. The facility of logging to is only done by subscribers. The location as well as the color id is not available in this service. Differs in sound quality in comparison to convectional phones.

2. VoIP partially accessible to and from PSTN: All the long distance incumbent carriers call along with calling card service where people are allowed to call from cellphones is included in such types of VoIP services. The termination and origination of calls are done via PSTN and the remaining portions are carried via the use of internet. These types of calls are generally made by enterprises in their enterprise networks. This service of VoIP is friendlier than VoIP via computer. Things to consider during this VoIP kinds are: 1. Subscription of package 2. Internet capability(faster) 3. Modem 4. Terminal adapter required for the conversion of analog signal to digital.

3. VoIP fully accessible to and from PSTN: Both the dialing as well as receiving ends require the PSTN along with the use of internet in order to get video and voice communication.

Working procedure of VoIP: 1. When we early select the receiver of the phone, it sends precise gesture to ATA.

2. When ATA receives the gesture given by phone, it sends dial tone that helps to understand that there is internet connection. 3. Then we dial the phone number of the party alongside whom we desire to talk. The tone is next modified into data by ATA and it is stored temporarily. 4. Data encompassing the dialed phone number is dispatched to the call processor of VoIP abiility provider in form of request. It is vital to understand whether the dialed number is in appropriate format or not. 5. Call processor of VoIP bestowing firm helps to understand to whom to chart the phone number. Mapping includes the translation of phone number into IP address. Above procedures defines the act of ATA and call processor. Nowadays its onset from the digitization of data. 6. After the analog signals are modified to digital by analog to digital converter there is a procedure of speech compression. 7. Traditional phone arrangement uses compression method shouted pulse program modulation. (8K samples/sec). Here examples of 12 bit are compressed into 8 bits by nonlinear gaze up table emerging the transfer rate of 8kbit/s. The compression method utilized by IP telephony or VoIP knowledge gives the ratio 16:1 that can be elucidated as 12Kbit/s can be compressed into 8Kbits/s. This kind of compression is beyond PCM and ADPCM. Adequate bandwidth is obtainable in LAN, consequently we dont demand compression. 8. After changing voice packets into data packets Real period Protocol (RTP) is utilized for period stamping. RTP additionally utilized for content identification of UDP voice packets. 9. Then the act of indicating arrangement comes into part. Indicating arrangement present pursuing operations; a. Finding IP address (destination)

b. Communication formation alongside consenting party afterward discovering destination IP c. After the arbitration, IP does compression of voice, length of buffer, packet period stamping and initiation of communication. Convoluted situation could be confronted if

indicating arrangement demand to converse alongside gateway that is present amid PSTN and internet. 10. After the formation of connection, as conversing there is back and forth transmission of packets. 11. Finally at the receivers conclude the packets are disassembled for the extraction of data and additionally for the conversion of data signals to voice signals. Next those signals are dispatched to sound cards/sound devices. 12. When we locale off the phone, connection is closed. ATA sends gesture to soft switch terminating the connection session. Three kinds of calls are there in VoIP. These three kinds of calls are a. b. c. PC- to PC PC to phone Phone to phone

In Route switching, dedicate line is used. Say the data are sent in the rate of 64 Kbits/sec. (each direction) for finished of 128 Kbits/sec. As 1 byte=8 bits, this translates transmission rate of 16Kb/second or 960Kb/second the route is open i.e. extra or less 10 Mb for talk of 10 minutes. The main reason of above example is to display that the far data is wasted. This is due to dedicated route used. But this doesnt transpire alongside packet switching. Packet Switching When one party is conversing and one more is listening at the alike time. In this scenario merely half of the connection is in use at precise time. Based into this, we can dispatch half of the file for efficiency matter. Momentous period is indulged as dead air i.e. nobody is talking. Even removing these soundless intervals, file can be tinier as contrasted to preceding one. There is no demand of possessing dedicated line or steady data rate as in route switching. Here we can dispatch and accord data packets whenever we need. Here packets of data move across IP possessing thousands of probable ways.

Figure showing complete VoIP system

Codecs in VoIP: Codec has its expanded form called codec decoder. The expanded form of codec is Coder Decoder. Codec converts voice (audio signal) to digital compressed form for the intention of transmission and once more change into uncompressed audio format at receivers end. This is frank frank portion of VoIP. Codec present the job of conversion by sampling countless thousands period in every single second. For example G.711 codec performs sampling audio at sixty for thousands examples in one second. Every single tiny example are next modified to digital form and sent across media. Those 64,000 examples are once more re-assembled at receivers end. Even though little portions amid 64,000 examples are missing, there is negligible difference. It is the reason that even though a little examples are missing, it touches like constant sound. Many kinds of codec can be utilized in VoIP that has disparate sampling rate such as 8,000 periods in one second 32,000 periods in one second 64,000 periods in one second

A codec shouted G.729 A is utilized most commonly. This kind of codec has the sampling rate of 8,000 periods each second. Codec aid in sorting, sampling, compressing and in the end packetizing the data alongside aid of elevated algorithms. Most utilized codec algorithm utilized in VoIP is CS-ACELP. CS-ACELP stands for Conjugate-Structure Algebraic-Code-Excited Linear Prediction. CS-ACELP grips updating and coordinating finished bandwidth. Two portions of CS-ACELP shouted Annex-A and Annex B grips disparate functionalities. Annex-A Creates law of transmission.

It basically follows If no one party in talk conversations, no necessity of

dispatching the data The efficiency consequence pursuing above law has aid packet switching to stand as

extra superior to route switching technique. Annex-B It grips the obligation of the aspect of calls made across VoIP technology.

Codec helps in sorting and changing everything but it doesnt understand whereas the data ought to be sent. This procedure is grasped by Soft Switch. E.164 is the term that is given to average for NANP that stands for North American Numbering Plan. This is the kind of numbering arrangement exceptionally utilized whereas to dispatch the need data. For example the number (212) 555-2341 has three disparate portions i.e. (212) is State program, 555 is metropolis program and last 2341 is Road address. There is something disparate alongside the number of VoIP. It doesnt pursue NANP as ordinary phone web follows. In VoIP there is IP address. It looks like 192.158.10.7. IP address corresponds each computer, router, gateway or switch. IP addresses are not always same. There are allocated by DHCP server so IP becomes disparate in every single connection. The trial of VoIP is to change the NANP numbers into IP address and next to find the destination IP address, a locale whereas we demand to dispatch each data. This kind of mapping procedure is usually grasped by something shouted Call Processor that is a hardware that runs a mapping plan shouted soft switch. The purpose of the soft switch is to link the conclude points (package of contraption and user). Switches have to know: Location of the conclude point. IP address of the corresponding conclude points. Phone number associated alongside precise conclude points.

VoIP protocols: Protocols are required in VoIP for the full functioning implementation of all the services of VoIP (call services, streaming, signaling of both audio and video) and also QoS (Quality of Service). H.323 protocol, SIP protocol, MGCP protocol are the protocols included in VoIP services. The common protocol stack is used in VoIP which is presented below:

The commonly used feature is contained by protocol stack (H.323 or SIP). From the above figure we came to know that IP is passed over the network and physical and transport layer first. After that UDP and TCP is compressed into IP packets. Again all the VoIP packets are encapsulated in the form TCP/UDP depending upon the operation of the signal. Then RTP and RTCP protocols are encapsulated as protocol into datagram called UTP. The quality maintenance of the voice within whole VoIP is maintained by RTCP as it is liable of gathering information about delays, packet loss and jitter.

H.323 family of protocol A standard which is globally recognized for providing disclaimers for speech communication over Local Area Network propounded by International

Telecommunication Union. Officially it was invented for the multimedia conferencing on the process of advancement it also covered VoIP. In 1996 and 1998 its first and second version was released gradually. It is capable of allocating both multi point as well as point to point communication. It is mainly used in the process of communicating packet data over packet switched networks that helps configuring calls, terminating it, authenticating it, registering and vivid functions. The protocols are gradually carried over UDP or TCP protocols. Terminals, gatekeepers, MCU, and gateways are the major components involved in H.323 protocol.

Components of H.323

H.323 Protocol Stack UDP is used by registration, audio, videos whereas TCP is used by data control applications. Below listed figure is a better example of H.323 protocol stack.

H.323 protocol Stack Signaling and Control in H.323 H.323 is capable of providing three protocols i.e. H:255.0/Q.931 call signaling, H425 Media/Conference Control and H.255.0 RAS. H.255/Q.931 with H323 used for controlling signals. H.255RAS is used in order to establish connection between receiver and the host. As soon as the connection gets established H.245 starts controlling media streaming.

H.225.0 RAS It is essential in order to maintain smooth communications among end points and gate keeper which is sent over UDP which makes it capable of performing time outs and retries count.

H.225.0 Call signaling It controls the message carried via the channels of signaling. By the use of transport address connects as a direct communication between calling and receiving end in the absence of gatekeeper. It also sends signals with the help of transmission control protocol. Transmission of calls are of two types call routed signaling via gatekeeper and direct signaling of call to its endpoint.

H.245 media/Conference Control As soon as the establishment of the connection is finished H.245 media/Conference Control is utilized. All the media channels are to be necessarily negotiated which are carried though RTCP/RTP. Following are the functions of H.245 media control: a. Assigns multipoint controller for central control. b. As soon as the call gets established capability of exchange begins at any time. c. All the media channels are termed as logical channels in H.245. d. Capability of conference control. Call Setup in H.323 includes Detection of gatekeeper Registration of receiving ends Record of end point call setup Capability to exchange Call termination

Below figure shows H.323 protocols with transport mechanism.

Basic H.323 Call

Session Initiation Protocol: For the formation of VoIP connections it is given by IETF. It mainly processes initiation, modification, and termination of the preferred sessions. It is merely similar to client server architecture. From all the transactions appeal and reply are done plainly. In order to maintain reliable connections SIP generally uses INVITA and ACK memos. As SIP has its own protocol it is the autonomous form of TCP. For the initiation of grasping codec negotiation it is fully dependent on session description protocol. In order to be succeeded on compatible media SIP maintains description of session permitted to members. By demanding and proxying request SIP upholds the user mobility. Services provided by SIP are: 1. For the successful communication SIP determines local end points of the user. 2. It maintains the parameters formation and ringing process between caller and receiver. 3. Identification of willingness of the caller in order to make a call. 4. Transfer as well as termination of calls is handled by SIP. Components USED in SIP: 1. Agents of user 2. Network server

This figure points SIP architecture Messages in SIP The various verities of message types are used by SIP. The medium of communication in between SIP server by and client is done SIP message.

Diagram showing SIP messages are provided below:

SIP operation overview: The process of identifying the caller as well as receiver in done by SIP addresses. If a caller has to make a SIP call he finds an applicable server and only then the appeal can be made. The call id in the memo helps to identify the call smoothly and uniquely. The procedures regarding SIP operation overview are presented below: 1. Addressing of SIP: Sip hosts are identified by SIP URL. It stays in the form example@host.

2. SIP servers location: The correspondence to URI and the Internet Protocol Address helps to maintain SIP server location (unified appeal identifier). 3. Transaction regarding SIP 4. Invitation of SIP 5. Location of user. 6. Change of continuous session

Figure showing working process of SIP Sample SIP operation

Other Supporting protocols These are the some other protocols that work along with SIP: For the reservation of resources Resource Reservation Protocol (RSVP) is used. For the transportation of real time data RTP/RTCP is used. For the streaming and controlling RTSP is used. For multimedia session and advertisement SAP is used. For the description of session SDP is used.

H.323 is also found to be working with RTCP and RTP. Media gateway and signaling are the two parts of voice gateways. MGCP is used to communicate between those voices gateways. MGCP is capable of operating with both H.323 and SIP.

Figure showing MGCP architecture

Issues in next-generation VoIP technology for equivalence with PSTN There are countless subjects that are to be address vitally to furnish quality VoIP service. Points relevant to subjects in VoIP knowledge are debated below: 1. Types of conclude terminal utilized in VoIP

The upheld abiility set ought to be utilized to use VoIP technology. Countless works are being seized to furnish PSTN equivalent services. Probable choices of conclude terminals that can be utilized for prosperous implementation of VoIP knowledge are tabulated below: 2. IP phones POTS black phones Key arrangements and PBX Soft Clients in PC (i.e. web established application) Choice of protocols for signaling

Many indicating protocols have been made that prop the VoIP solution. The decision concerning correct protocol is vital for the proper implementation of VoIP. The protocol selected for the knowledge are reliant alongside abiility set and equipments utilized in technology. For example, to use SIP phones, we ought to select the correct protocol i.e. SIP for the web to admission it. A little protocols include 3. Device manipulating protocols such as MGCP, H.248, NCS etc. Service accessing indicating protocols e.g. H.323/SIP N/W abiility indicating protocol such as SIP, CMSS, BICC Security

Security is the main key word to contemplate concerning the proper implementation of VoIP. There are countless protection subjects connected to VoIP technology. Most vital are Theft of Service Denial or service Privacy invasion

Security subjects are extra debated on the protection serving of this project. 4. Quality of Service

Although countless messengers are concentrated on making voice quality in VoIP to be possessing extra quality, the main necessity of the expansive placement of VoIP is to furnish toll quality abiility that is equivalent to PSTN that continue today. The quality of voice in VoIP needs pursuing three criteria mainly. They are: Delay Packet Loss Jitter

We all understand that VoIP knowledge use IP (Internet Protocol) for the transmission of the data. We cannot disregard the data defeat or each setbacks considering the use of IP. IP doesnot furnish the maximum period promise concerning the services, so there is the demand of requesting the proper QoS solution. There are countless technologies to furnish QoS support. They are RSVP, MPLS, even ATM. The main goal of this is to safeguard that the quality of voice is not compromised by IP traffic. 5. Reliability and Availability

As it is recognized that PSTN has five-nine reliability that is equivalent to less than 5 minutes downtime/year and it is prosperous to grasp million of calls at the alike time. VoIP web additionally needs to have alike kind of reliable outline that PSTN possess. In VoIP, employing burden allocating and redundant webs and corresponding equipements ought to be requested to furnish reliable services. Obligation tolerant gateway, indicating servers can be used. Procedures needed possessing obligation tolerant abiility includes a little frank points. They are: 6. Hardware upholding redundancy Redundant web and connections encompassed in it Capability shouted hot-swap No wreck in even a solitary point Up gradation of services and equipments lacking each defeat of the service. Lawful interception

Lawful interception additionally knowned as wiretapping is relevant in phone services. With PSTN, there is frank procedure to tap the phone line by regulation implementation

association alongside the court order. The introduction of VoIP has made the weighty subject concerning this matter. The necessities for tapping in PSTN as well as VoIP webs include: 7. No chance of tapping lacking court order Tapping doesnt apply to suspects but merely for phone numbers. Suspect have to not notice tapping. Tap have to not transpire in gateway but inside the network. More than one association cannot tap suspects. Emergency and operator services

PSTN provides flexible emergency and operator services. For example we can seize as 911. Placement of VoIP web ought to additionally contain those supports. Legacy data prop and locale data abiility ought to be seized into thought as requesting VoIP. 8. Number Strategies and Call Routing

PSTN has skill to path call globally. This is probable due to possessing well described numbering design both globally and nationally. Routing tables are made to furnish end-toend connectivity. A subsequent creation VoIP knowledge is needed to contain pursuing points: 9. National and global addressing strategies Addressing schemes of SIP Connection to E.164 and PSTN numbers Routing of call amid addresses or numbers. DTMF and Phone events

There is an subject in transmission of DTMF and countless supplementary tones encompassing assorted phone events. They can be completed alongside G.711 that are maximum rate codecs but, lower bit codec doesnt prop this kind of services. There are countless resolutions obtainable for transmission of these kinds of services. Most utilized kind of resolutions is as given below: Use of RTP packets

 10.

Using indicating protocols Firewall and NAT transversal

For the supplies such as subscribers gateway and IP phones, there is firewall at frontier of premises. Supplement to this there could be attendance of Web address translation to elucidate inner to external IP address. It is extra vital that both flow of gesture and RTP traffic can debate both firewall and Web address translation. Authorized flow ought to be implemented. 11. Billing and reconciliation

PSTN has precise and comprehensive billing arrangement for billing and reconciliation amid providers. It uses generally each minute billing method and a little uses flat billing. VoIP abiility providers have to the mechanisms comparable to that to produce revenue. It is probable that continuing mechanism of billing will continue both for subscriber and inter messenger conciliation. 12. Network Interconnection

PSTN is not a solitary web but it is a collection of networks. At every single point web interconnection is required. Scenario of collection of web alongside interconnection will be needed in subsequent creation VoIP networks. 13. OSS support

PSTN has Procedure prop arrangement possessing pursuing functions: Fault isolation Performance monitoring Alarms Loop testing Enforcement and meaning of policy

A subsequent creation VoIP web will seized these points into thought employing the infrastructures upholding these functionalities. 14. Utilization of Bandwidth

Digitized voice signals are compressed in VoIP networks. They are less than 100bytes possessing header of 40 bytes. Digitized voices are transported employing RTP. Header are

momentous and reduces bandwidth and number of voice channels. Employing lower bit codec by VoIP web is the good supremacy of it. 15. Modem, Fax and TYY support

As we understand that PSTN supports fax, modem and TYY calls. VoIP additionally ought to contain the comparable reliable abiility but those services demand a little supplementary laws beyond IP traffic. Voice traffic has less packet defeat but those Modem, Fax and TYY is extra sensitive towards packet defeat alongside less towards finished delay. Modem, Fax and TYY is upheld by VoIP webs if maximum rate codec is used. 16. Auto Configuration

A outstanding difference amid established phones and VoIP web is that VoIP web needs extra configuration required. Auto configuration is the vital task as the web grows up. Auto configuration can be completed employing DHCP.

VoIP vulnerabilities:

As VoIP is performed via internet it is not totally secured. There might be different vulnerabilities. Threats concerned to VoIP are mainly branched into three parts namely: Received threats from IP Associated threats together with VoIP. Specific threats to VoIP. Fluxuation of execution. Insufficient data verification Flaws in string/arrays manipulation Insufficient resources Flaws in file/resource manipulation Improper password management Privileges and permissions Errors during authentication and certificates Scarcity of contingency system Haphazardness and crypto

Services regarding quality in VoIP: The service regarding quality is the essential service in VoIP. Although the main aim of VoIP is to provide quality audio service it is unable to maintain its quality of sound in weaker network connection in comparison to PSTN phones. The implementation of different protection measures in order to maintain secure environment might degrades the quality of VoIP process. The complications regarding VoIP are delay in call setup, latency, encryption and jitter. The vital role in VoIP is paid by the quality of service. Following are the list of the different quality concerns connected within VoIP: 1. Latency

2. Jitter

3. Loss of packet 4. Normal and competent bandwidth 5. Speed requirement 6. Power management and backup facilities

Solutions to VoIP security issues 1. Encryption at end points In the router, bottlenecking problem may arise due to the cause of encryption. For this kind of problem the only solution is that to install the encryption solely at the endpoint. Due to the high

power of end point have been able to handle the encryption mechanism. It is not flexible to install on every hop, such as IP phones. 2. Secure real time protocol for the transmission of audio/video data RTP is used. Attacker can easily misuse the use of RTP if there is no security. SRTP (secure real time protocol) is used as a protocol which help to keep the confidentiality among the different sectors like authenticating messages, IP traffic protection and protection of replays . 3. Key management for SRTP-MIKEY Different parameter is derived in the session keys which are used by the SRTP like authentication, encryption and protection of integrity. Such scheme that are related to the MIKEY deals with the management which helps in the addressing of multimedia session. Mainly deals for the purpose of multimedia session with security management and to update. 4. Better scheduling schemes: Stand in line for the long period of time huge amount of packet can cause packet delay and data loss. Quality of service is supported by the crypto engine but this cannot be as a sensible scenario because of compression and issues related to crypto engine. There are some routers that quality of service keeps in mind for the packet schedules. After the encryption, quality of service can be prioritizing. Whole portion is dependent on hardware and software so the functionality cannot be guaranteed. 5. Packet compression: Due to header size and content packet can be compressed, internal header can be compressed up to 4 bytes (approx.). For the achievement of the good quality compression both the end point must follow the same architecture compression that is used for the other. 6. Resolving NAT/IP security Incompatibilities: UDP encapsulation, Realm Specific IP, IPv6 tunnel broker and IP next layer(IPNL) are used for the problem to solve NAT/IP compatibilities.

Following are the benefits of VoIP: 1. Savings of Cost:

VoIP is the price competent method of voice transmission. After we switch from PSTN knowledge to VoIP arrangement, global calls are cheap. Instead of employing convectional business contact line, traffic in VoIP arrangement flows on Internet or confidential lines. VoIP is prosperous in cutting price for lines, maintenance, manpower and equipment. All of the traffic is consolidated into solitary physical web consequently there is no demand of distinct PBX. Even though there is extra price needed to mount VoIP arrangement, there can be extra savings. Different manpower is not demanded as maintaining VoIP systems. There is no demand of disparate teams for data web and voice network. VoIP reduces subscription number for global and long distance voice calls. 2. Rich mass media Service:

Traditional phone arrangement merely provides voice and fax communications. Mass media services is lacking in established phone system. Across VoIP knowledge, people can make video calls, transfer pictures etc. VoIP arrangements are able users to use mass media services due to the integration of protocols and supplementary applications. VoIP arrangement not merely provides options for mass media but additionally craft marketplace for contact industry. For e.g. mobile phones employing VoIP 3. Phone portability:

PSTN uses dedicated line, so we cannot move phone set from one locale to one more if the use of alike number is required. Long procedures are to be pursued to use number if anybody move house from one locale to another. IP phones can be utilized anywhere employing alike number. It can be utilized whereas there is internet connectivity. 4. Service mobility:

The mobility context additionally includes ability mobility. Wherever the phone is seized, we can use alike features and services. 5. Integration alongside supplementary applications:

VoIP protocols such as H.323, SIP runs on request layer and those are consolidated alongside supplementary requests such as instant envoy, browser etc. Collaboration alongside request provides extra priceless services to users. Examples of this are dispatching voicemail across email. 6. User manipulation interface:

Most of the providers of VoIP abiility furnish a web GUI association arrangement to change, add and delete features, think call logs, rates etc. By use of that GUI, users can change voicemail number, call forwarding number, data of attendance, music on grasp option etc. 7. No geographical boundary:

There is no geographical check in VoIP system. People living in Nepal can use the number of US and Nepal to US calls are all indulged as internal calls hence, cheaper than established global long distance calls. The span program or geographic program is not attached to particular location. 8. Rich features:

Features are extra in VoIP arrangement than in established phones. The features of VoIP phone have by now been debated above. Public features that the VoIP phone poses are ring back tones, call forwarding, find-me-follow-me, ring at countless numbers at alike time, click-to-call on webpage etc. 9. Cheap hardware and software:

Only sound boxes and microphones are needed as supplementary hardware than our computer and internet. There is no demand of phone set. Multimedia needed are free in internet. Supplies needed for VoIP is moderately cheaper 10. Simplifying maintenance

Only one cable can be utilized in VoIP making the maintenance procedure easier. Maintaining VoIP web is extra or less comparable to computer network. SO, person possessing the skill of computer web can uphold VoIP. There have to not need distinct accomplished manpower to be retained for maintenance of VoIP. 11. Secure call:

Secure call can be made employing safeguard protocol such as safeguard real period transport protocol (SRTP).

Areas applicable: Todays market is widely covered by VoIP due to its easier ease of use. The VoIP technology offers long distance calls cheap or almost free of cost. The VoIP technology brings benefits to both individual and institutions. VoIP are widely used in different fields as per the requirement of the communication. 1. Exchange of carriers: It acts as a medium for long distance as well as international carriers. The telephony system from the every corners of the world is run through VoIP and act as a medium of communication for all individuals. As the long distance calls are provide at low cost it act as a medium of international communication very efficiently.

2. LEC, Cellular, and cable enterprise: To maintain low cost communication in LEC and other enterprise VoIP is used. Cable companies are now providing a triple play package coaxial cable, cable televisions, and internet package at additional cost through the use of VoIP.

3. Among business and enterprise: VoIP has been successful in attaining strict and secure exchange divisions and central workplaces phones in tiny business. It has been due to the price saving and extra supplementary features provided by VoIP. All the business associations include VoIP

services in their office as call centers employing personnels. VoIP technologies are widely used by outsourcing company. 4. Individuals: People can individually install VoIP in order to make voice communication for their private purpose as it is easier as well as friendlier to work with. The most widely used telephone circle is use of the internet. Nowadays video conference has been popular in different organizations as the medium of chatting. VoIP applications: These days are the days of VoIP in both individuals and business firms. With the help of computer we can have voice communication among the world in the presence of internet connectivity. The use of software in computer is necessary to manipulate VoIP. Nowadays there are loads of VoIP applications downloaded over internet in our day to day use. Following shows some of the lists of VoIP applications mostly used.

Disadvantages of VoIP 1. Problem if power failure occurs: VoIP requires power to operate whereas PSTN gets power from the service provider. The equipments are required for the generation of power to operate all the tools used in VoIP. The failure of the power may cause the system failure and end of VoIP services. The use of battery might control the sudden system failure but the installation of battery increases the installation cost. 2. Limited Emergency Calls: The emergency services are not included in VoIP thus; there is no use of VoIP in emergency cases. The tracing of the location is easier in PSTN comparing it to VoIP. Whereas the service of tracing is totally absent in VoIP. This issue is being addressed by E911. 3. Sound quality and reliability: The entrance to threats and mitigation is commonly high in VoIP in comparison to PSTN as VoIP uses internet protocol. The order of the data sent over IP gets scrambled. The voice packets used in VoIP are sending from and to different ends. The voice quality gets poor if there occurs packet delay or loss of the packet. PSTN is considered as a dedicated line and is more secure than VoIP. The standard phone has maximum delayed up to 10ms whereas VoIP delays up to 400ms. 4. Use of firewalls: People might feel easier if the calls are free and those calls are secure. The sophistication of the business increases due to the use of VoIP as the business firms uses firewalls and the firewalls creates problem during the routing of VoIP. 5. Relies on Bandwidth: The fluctuation of the bandwidth results increment of latency, data loss, jitters etc. as we know that the bandwidth is not always high VoIP seems to be unused in the places with low bandwidth. The quality of sound gets better in high bandwidth and if there lacks the good connection the quality gets degraded. 6. Lack of consumer protection measures: VoIP lacks the legal requirement for the providers in order to overcome the faults in it.

Conclusion VoIP is considered as the emerging technology which is implemented by many people in their day to day lives or in business. The matter of its high demand in the market is it is different in comparison to other traditional phones which always and for the transmission of the voice in packets form IP is used. VoIP is considered to be more vulnerable as per the security purposes than PSTN. VoIP is mostly used in business organization among them which most of them are the companies of outsourcing as their location varies. The increase in the trend of the use of VoIP gradually increases the trend of competition and quality of services in the market. VoIP is popular because of its friendliness and easy configuration. In comparison to PSTN telephone networks VoIP is cheaper in terms of long distance calls especially international calls. The expanded and wide features of VoIP has proved to be successful thus, individual and the business organizations prefers it in comparison to PSTN. The point of drawbacks in VoIP is it lacks emergency calls in which PSTN gets stronger. Thus, the next upgrade of VoIP must possess emergency features.