Voice over Internet Protocol Introduction VOIP(voice over internet protocol) pronounced as voyp[vop1] is a technology that transform audio

analog signal entering them into digital data that can be transmitted over internet which allow us to communicate to different people through the internet.[1hstuff]People are using it for a long time without realizing it. Things msn (voice chat) , Skype, Yahoo(voice chat) all are the most used common example of VOIP. VoIP is revolutionary emerging technology that is completely reworking the phone system. Internet must be needed for VOIP at the beginning But today internet is not strictly important for VOIP But the protocols used by the internet is same needed for VOIP technology. Therefore VOIP can also be defined as the voice that travels from the same protocols used on the internet. [vop2]. Voice compression and decompression technologies are must for VOIP. "Voice will be first encoded from analogue to digital packets and then decoded back to analogue voice again at the receiver end". The codes like G.711, G7231, G.7231 and G.7291/B are most popular codecs that are used by today's VOIP. VOIP is also referred as IP telephony (IPT) because internet protocols make communication possible in VOIP. Technically, the implementation area of IPT is only to local area network (LAN) that is being used in single building by organization, but when IPT moves across LAN to WAN or any external network by same company at different location or to internet. Then it turns into VOIP. [vop3]

History of VOIP The use of VOIP is increasing rapidly. When we look at the network structure of VOIP we certainly determined VOIP is not possible without internet. But first VOIP call has been made at 1973 when there is no internet. Arpanet network is first used to send voice across digital network. But it can only send voice between two computers on the private network on the ARPANET network. VOIP slowly start emerging from ARPANET to the global internet from the first invention of VOIP software named "Internet Pone" was first introduced by a company called Vocaltec. Vocaltec can be run in most of the home PC and utilize and need the same devices like soundcards, headsets and speaker. At that time the internet phone software use H323 protocol instead of SIP protocol. Although Internet phone was introduced has been able to achieve some success. Due the slow internet the calls are not quite clear than today's VOIP services. Therefore use has to comprise on quality of voice in earlier VOIP services. Therefore the communication quality will be like walkie talkies. Another problem with earlier VOIP is that the computers who are make VOIP communication should have the same soundcard drivers install on their computer. Modems are taken the major source for transmission which utilize tradition phone lines and thus will provide service. The quality of VOIP is earlier days are worst than the normal Phone line. VOIP start growing with the development of broadband internet and cover around 1% of US phone call by 1998. Then various company start investing of VOIP market. Different other company starts developing software for VOIP. With the massive expansion of broadband internet user the customer of VOIP service in USA increase to 3%. IN 2005 the various improvements over voice traffic and data traffic has been addressed. At 2005 around 3 billion revenue has been acquired from different VOIP equipment sales.There are many service provider available today for personnel as well as business uses. The service will be depending upon the service provider some only provide PC-to-PC phone whereas some providers phone to pc and phone to phone call. The attracting and advance features and unlimited plans offered by VOIP providers like unlimited calls plans attracts more customer day by day.

http://www.articlesbase.com/networks-articles/a-history-of-voip-837826.html

Benefit of VOIP VOIP has lots of benefit that helps to overcome the deficiency that are in traditional phoning system. Here are some of the benefits of using VOIP telephone. 1. Cost efficiency VOIP helps to reduce the cost for individual or for Business Company. Comparison to normal phone VOIP can reduce 30 to 50 percent costs in long distance and international call. [buz1]. Therefore it works as great mechanism to reduce the cost in phone in business; the saving can be million while the individual can reduce their cost in phone. VOIP user can use it for free if the call is over computer to computer. 2. Advance Features VOIP has lots of features that are not available in traditional phone. User can share the data, picture at the same time they are having the call which is one of the best features for business users. The other features like conference call, voice mail, call forwarding makes VOIP popular among individual and companies. 3. Portability User can make VOIP call if they have VOIP account and broadband connection. They can simply log in into their use account whenever they are and just plug the headphone on their computer or laptop or just use internet phone and make the call. If user are using VOIP on wireless phone and they are on WIFI zone they can simply make call logging in their account. 4. Unlimited Call Most of the VOIP providers provide unlimited call services. The features is most applicable for the business user who have to user lots of phone or for the user who know exactly how much amount will be spend in the phone call.

Disadvantage of VOIP Lack of emergency call VOIP service is not geographical based therefore it will create problem in local emergency call. Emergency call is originally designed for PSTN network. Your phone call cant be traced and the emergency service may not be available if you cant speak in phone for some reason. Internet is Must Without broadband or Internet connection no one can imagine VOIP. VOIP is useless if you dont have internet connection. If there is problem with your internet connection or your ISP, VOIP is not possible User cant truly reply on VOIP connection. If the user has to make some phone call and the internet is down then the call is not possible. Need of Electricity. Normal PSTN line doesnt need electricity to operate. You personally dont have to arrange for the phone but for VOIP power is another crucial factor. If there is blackout or power outage then you cant use VOIP phone. Alternatively you can arrange power backup using battery. Security: Security is big concert in VOIP. Since VOIP is unencrypted, anyone who can have access to the LAN of an office could easily tap telephone conversation by connecting to networking monitoring tools. The various security measures like barriers, locks, guards access control system can be use as defense to stop unauthorized access.[scv]

Types of VOIP There are commonly three different way in which VOIP services is used today.

ATA (Analog telephone adapter) ATA is simply a device that is used to connect standard phone to computer network to make VOIP calls. The main works of ATA is to convert the analog signal into digital signal to transmit

over internet. The connection of ATA is between existing telephone networks and IP networks (broadband connection).

IP phone: The design and looks of IP phone is just like a normal phone with handset, cradle and buttons. However instead of having the standard RJ11 connector IP phone use RJ-45 Ethernet connector. Therefore IP phone can be connected directly to switch, router. USB phone has a standard USB connector that can be plug into USB port of the computer which looks just like a normal phone.

Computer to Computer: It is the easiest way and most commonly used services to use VOIP. The call can be done with the help of few devices and software. Of course you need internet for this purpose. The software will be available for free or for very low cost depending upon the company you choose. You need a high speed internet, microphone, and soundcard on your computer to use computer to computer VOIP. There will be no cost involve in this process. Various numbers of companies offer it for free. User just needs to pay the monthly bill of his internet connection for this purpose.

VOIP codecs Codec stands for coder and decoder. Codec converts the audio signal and compression audio signal into digital form for transmission and again uncompressed audio signal for reply. Vocoder is also referred as Codec for VOIP for voice encoders The conversion in codecs occurs by sampling the audio signal 1000/second. For example the sampling of audio of G.711 is at 64000 times a second. Each and every tiny sample will be converted into digitized data and compressed the sample for transmission. At the end the total 64000 sample are resembled, there will be the pieces of audio signal missing which is very small to ear of human being Therefore they don't realize it and feel like the continuous flow of audio signal. The codec will determined the sampling rate used in VOIP. Below are some of the Codecs that is used in VOIP. The most common used codec in VOIP is G.729A which has a sampling rate of 8000 times per second

Codec G.711

Bandwidth/kbps 64

G.722

48/56/64

G.723

15.33/6/6.3

Comment Deliver precise speech transmission. Very lower processor requirement, Need at least 128 kbps for two way Adapts to varying compression and bandwidth is conserved with network congestion High compression with high quality audio. Can use with dial-up. Lot of processor

power. G.726 16/24/32/40 An improved version of G.721 and G.723(different from G.723,1) G.729 8 Excellent bandwidth utilization, Error Tolerant. License required. GSM 13 High compression ratio. Free and available in many hardware and software platforms. Same encoding is used in GSM cell phones ILBC 15 Robust to packet loss. Free SPEEX 2.15/44 Minimizes bandwidth usage by using variable bit rate. Advanced algorithm is used in codec for compress, sample, sort and packetize audio data. "The mostly used algorithm in VOIP IS CS ACELP algorithm (CS-ACELP = conjugate-structure algebraic-code-excited linear prediction)". It streamlines and arranges the bandwidth that is available. The transmission rule is created by Annex B which mainly purpose is not to send data if no one is using it or no one is talking. The ability of Annex B makes packet switching better than circuit switching. Therefore Annex B is CS ACELP is responsible for VOIP call. E.164 named given to NANP (North American Numbering plan) helps to route the people's call based on the number they have dialed. Switches use different code number to route the phone call to different region. But VOIP works on the IP address like 192.168.99.125 to correspond with different devices used for VOIP. Therefore in VOIP NANP will not be used to determine any phone number. IP address may be static or dynamic. DHCP server will assign a dynamic IP address, the main problem or challenge in VOIP is to change NANP number into IP address and determine the IP address of that particular number. This process in handled by Central call processor a hardware device that includes a program called soft switch that acts as a database mapping program.

Protocols used in VOIP There are different device which can be called as hardware and software that are used for VOIP communication. Therefore protocols make all these communication possible between different hardware and software. Protocol is just a set of rules that is used to determinate the transmission of data. There are various protocol that is used for VOIP communication. Protocol creates rules from where device can connect to each others and to network for VOIP. Before audio transferring from one device to another protocol should be there to establish remote device and find the means to transfer the media between two different devices. The mostly used protocol in VOIP is H.323; it is initially designed for video conferencing. SIP (session initiation protocol) is

another protocol most commonly used in VOIP. Both h.323 and SIP has been originated in 1995 to eradicate communication problem between two computer in order to transfer audio and video. However H.323 has been able to gain huge commercial success compare to SIP. Even though the basic uses of these two protocols is to establish a communication between users. There is slightly different in design between these protocols. H.323 is a binary protocol, whereas SIP is ASCII based protocol. With the invention of this protocol overs the years there is a debate between SIP and H.323. Though both protocols can do the job but H.323 is more than SIP in various following reason     H.323 has better interoperability with PSTN, Video support is better in H.323 Excellent interoperability with legacy video system. DTMF transport is reliable

On the other hand SIP was not designed to target that issue. The popularization of SIP WAS from the promotion statement that SIP is very easy to debug and implement. However there will be certain portion of difficulty that would be in communication system. There are various non standards SIP like SIP-T and SIP-1). Both H.323 and SIP is well known as "intelligent endpoint protocols Others protocol like H.248 and MGCP are also commonly used in VOIP. There are other different protocols used in VOIP. Currently ITU is focusing on H.325 new protocol that has better quality than h.323 and SIP protocol. Beside these there are other protocol which happens to be very successful in terms of quality and business. Skype is another technology that uses non standard protocol and has been able to achieve high success in VOIP in terms of business and services. http://www.packetizer.com/ipmc/papers/understanding_voip/voip_protocols.html H.323 Protocol

H.323 is combination of different terminals known as endpoints, multipoint control unit, back end service, gateway and a gatekeeper. The bandwidth control and address resolution will be maintained by gatekeeper thus is the regarded as the major components of h.323. The main duty of gateway is it acts as a bridge between H.323 network and other different network which doesnt include H.323. Multi control unit is taken as a optional element that assists other communication and multipoint conferencing between two different end point. On the other hand gatekeeper plays important role in the VOIP. It maintains the Back End Service (BES) on endpoints also includes services permission and configuration. In H.323 standard, various types of H.323 calls have been mentioned.
    Gatekeeper routed call with gatekeeper routed H.245 signaling Gatekeeper routed call with direct H.245 signaling Direct routed call with gatekeeper Direct routed call without gatekeeper

H.323 sessions start by either UDP or TCP. It depends upon the call module used. If the starting point is RAS with the h.225 signal connection then UPD connection will be used. The signal thus have RAS (registration Address protocol ) that acquire the address at the end by contacting with gatekeeper. After this procedure Q.931 protocol will be used to maintain the call by itself and then negotiate the information of address for H.245 signal which is performed by TCP; Q.931 which perform the escapsultate the h.225 call signaling message to H.225 . The setup next methods is mostly used till the H.323 progression. In the process the protocol walk though the configuration of the another protocol that is being used. This process is necessary because there is no standard port in

H.245. H.225 is simply used to make a connection while for the media transfer process H.245 establish a channel. This process is performed though TCP. The most benefit of this process is H.255 message can be combined with h.245 message in urgent situation. H.323 call can be setup with the help of oneroundtrip which offers a fast connect. The connect and setup message alter H.245 signaling element. The several stuff should be make with the help of H.245 which include audio codec and logical channel for the transfer of different media information. RTP and RTCP ports are brokers by OpenlogicalChannel. RTP and RTCP logical are only one directional therefore four connection should be established. The call begins with the establishment of H.245 and other logical channels. H.323 includes the various other protocol associated with other different form of communication H.450.1, H450.2 and H.450.3 H.235 and H.246. At the end each authentication is possible with the help of prior shared secret or symmetric keys. Therefore the different extra protocol creates some technical difficulties in setup process of H.323.

Call Monitoring in VOIP Monitoring is one of the great aspects of improving the quality of the services because none of the service provider can guarantee the perfect service with 100% uptime When it comes to phone, VOIP, webhosting, or any other network related services, Monitoring is the most valuable things in any of the network related services. Like every monitoring is different from others in several reason. VOIP call monitoring is simply referred as quality monitoring (QM). Quality Monitoring basically rate overall quality of VOIP call. These monitoring is done by using various hardware and software which helps to do all testing and rate the quality of calls. VoIP calls travels over IP network. IP networks can be track much more easily than PSTN lines therefore it is much easier to monitor VOIP calls than normal PSTN calls. However there is some security thread as it travels on the internet. Internet is more vulnerable than any other network. The hardware and software that are used in VOIP monitoring uses mathematical algorithm to determine the quality of the call. Each call will be given a specific score which ranges from 1 to 5. However 4.4 is referred as the most highest score and 3.5 is referred as good.. The most common score is referred as mean opinion score (MOS) The most commonly being used call quality guide used by various hardware are software are as follows: Latency The time delay in the conversation in VOIP between two calls is known as latency. Latency can be taken by one way or round-trip. Round trip is the problem where the person experiences the talk over effect due to poor calls, they stop the conversation between each other because they think that other person has stopped speak. The poor routrip latency will be considered as 3000 millisecond.

Jitter: The stream of packets may arrive to destination later or with improper order in VOIP call which is known as jitter. Jitter problem can be address with jitter buffer which ultimately gather the packets in small group and gathers all the packets in right order. Packet Loss: Sometimes in the VOIP conversation there might the issue called random loss and brusty loss. Random lost is just the lost of packet periodically through whole conversation whereas random loss is the drop of whole sentence in a conversation. Therefore the measurement of packet loss to received packets in percentage is called packet loss There are two different approach usually used in VOIP call monitoring. Active and passive call monitoring. Active call monitoring: Active VOIP monitoring is practice before the initialization of VOIP networks. It is generally done for the testing purpose and various equipment and network specialist perform this task. Therefore Active testing will not be into practice after the VOIP has been deployed. Passive Call monitoring Passive call monitoring monitor a calls in a real time after the company deploys network Of VOIP. The several problem will be detected by passive call monitoring like traffic problem, buffer overloads etc Security in VOIP Security is one of the biggest concern IN VOIP network. The internet is required for VOIP calls which itself is vulnerable therefore VOIP is suspectable to attack. The various type of attack can be performed in different circumstances, the hacker who knows the security issue or vulnerabilities of VOIP can perform DOS(denial of service) attack, they may further can perform any illegal activity by recording different phone conversation, destroying customer data, or misuse of mailbox of customer. When it comes to VOIP the security must be implementing in two most valuable assets of the customer data and voice. Sometimes the endpoint of SIP may freeze and crash when trying to tranfer the high rate of packet. There may be the failure on SIP proxy server with VOIP specific signaling attack of 1 mega bit per second. Even there is the low consumption of bandwidth the high packet rate may sometimes result denial of service attack. While routing the VOIP traffic though NAT and firewall there are often a several challenges. Along with the firewall private session border controller is often used to make VOIP

call from and to protected network. Example Skype use their own protocol to transfer call to Skype peers on the network which let it to transfer through firewall and symmetric NATs. There are other methods to traverse NAT which is known as STUN or ICE. Some of the security thread in VOIP Spam over internet telephony (SPIT) The uses of VOIP have been increased in tremendous amount in recent days. There are wide varieties of tools that are already available in the internet. Beside email spamming the spamming via VOIP is very fast, easy and cheap. VOIP voicemails require a large storage than traditional email spam. One voicemail spamming may require some megabytes of storage depends upon the size of data. Spoofing Spoofing is another vulnerable security thread in VOIP. An attracker may masquerade as being another VOIP caller. FOR example Mr. Smith may inject bogus caller id into any normal VOIP calls. Then the receiver thinks that the call might be from the known source for example the bank. Therefore the receiver may give any of this credit information thinking that he is talking to of the person of bank. In such scenario Mr. Smith may use this valuable account information like account number, security number, credit card information. Confidentiality concerns: Sometimes unencrypted data or voice may travels over the internet in VOIP. Therefore technically someone have a capacity to gather the data and again try to reconstruct a conversation. This activity is very rare to some portion. But there are various tools and software that helps to gather the pieces of information and reconstruct it. This may be increase whenever the VOIP call will be widely used. How to protect against security cause in VOIP There are various security precaution and principles that is being used for safe VOIP usage. Neglecting these principally may make you victim fro another attackers. Here are some of the good practice that you should follow to make any VOIP call secured.     Always user firewall on of your computer Use Antivirus program and update them regularly Create and User strong password. Choose a renowned VOIP service provider

VOIP call termination: When the call is made from a precise location using any device is known as call orgination whereas when the call reach in some VOIP device or computer then call will be terminated. Call termination can be occurs in two different way in VOIP. 1. Carrier class co-location facility based termination 2. Small home based termination

Home Based termination After all the previously mentioned stuffs needed for VOIP call like internet connection analog or digital phone line, VoIP call can be terminated with the use of gateway like quantum or Weltech. That is the only stuff that is need for VOIP call termination in home based. For per voice call in VOIP 16 kbps bandwidth is required. Therefore is the user has 256 connection . The user can simply terminate 16 different voice calls. However if the internet connection of the user goes below the 256 kbps/sec then the call termination should be decrease to give the quality voice.

Co location based termination Co location is most used word when it comes with networking or web server. Co location is nothing but just a services or facilities provides by various data center that allows you to put your equipment into highly advance secure premises. The equipments will be organized in rackspace with a multiple digital line and redundant internet bandwidth. The price of co location depends upon the different company. The gateway that will be put in data center will usually be digital gateway. The digital gateway takes 4T1 or 4 E1 lines. The different other larger capacity gateway also available

Benefit of VOIP http://www.buzzle.com/articles/benefits-of-voip.html security concern for voip, Special Publication 800-58, D. Richard Kuhn, Thomas J. Walsh, Steffen Fries csrc.nist.gov/publications/nistpubs/800-58/SP800-58-final.pdf

Books [vop1] VOIP for Dummies, Timothy Kelly. Page 34-35 [vop2]VOIP for Dummies, timothy Kelly page 34-36 [vop3]VOIP for Dummies, timothy Kelly page 36-45