network voip

8/6/2019 Network VOIP

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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 codec¶s

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 can¶t be traced and the

emergency service may not be available if you can¶t speak in phone for some reason.

Internet is Must

Without broadband or Internet connection no one can imagine VOIP. VOIP is useless if you

don¶t have internet connection. If there is problem with your internet connection or your ISP,

VOIP is not possible User can¶t 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 doesn¶t need electricity to operate. You personally don¶t have to arrange for

the phone but for VOIP power is another crucial factor. If there is blackout or power outage then

you can¶t 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 RJ-

11 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 codec¶s

Codec stands for coder and decoder. Codec converts the audio signal and compression audiosignal 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 codec¶s occurs bysampling 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'trealize 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 Codec¶s that is used in VOIP. The mostcommon used codec in VOIP is G.729A which has a sampling rate of 8000 times per second

Codec Bandwidth/kbps Comment

G.711 64 Deliver precise speechtransmission. Very lower

processor requirement, Need

at least 128 kbps for two wayG.722 48/56/64 Adapts to varying

compression and bandwidth isconserved with network

congestion

G.723 15.33/6/6.3 High compression with high

quality audio. Can use withdial-up. Lot of processor



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 over¶s 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

doesn¶t 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 withgatekeeper. After this procedure Q.931 protocol will be used to maintain the call by itself and thennegotiate 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 ³setup next´ methods ismostly 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.245establish 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 thehelp of oneroundtrip which offers a fast connect. The connect and setup message alter H.245signaling 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 ³OpenlogicalChannel´. 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 logicalchannels.

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.



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 beinganother 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 cansimply 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

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