Welcome to Scribd, the world's digital library. Read, publish, and share books and documents. See more
Standard view
Full view
of .
Look up keyword
Like this
0 of .
Results for:
No results containing your search query
P. 1
A Study of Voice over Internet Protocol

A Study of Voice over Internet Protocol

Ratings: (0)|Views: 196|Likes:
Published by ijcsis
Voice over Internet Protocol, is an application that enables data packet networks to transport real time voice traffic. VOIP uses the Internet as the transmission network. This paper describes VoIP and its requirements. The paper further discusses various VoIP protocol, security and its market.
Voice over Internet Protocol, is an application that enables data packet networks to transport real time voice traffic. VOIP uses the Internet as the transmission network. This paper describes VoIP and its requirements. The paper further discusses various VoIP protocol, security and its market.

More info:

Published by: ijcsis on Jun 12, 2010
Copyright:Attribution Non-commercial


Read on Scribd mobile: iPhone, iPad and Android.
download as PDF, TXT or read online from Scribd
See more
See less





(IJCSIS) International Journal of Computer Science and Information Security,Vol. 8, No. 2, 2010
A Study of Voice over Internet Protocol
Mohsen Gerami
The Faculty of Applied Science of Post and Communications
Danesh Blv, Jenah Ave, Azadi Sqr, Tehran, Iran.
Postal code: 1391637111
Voice over Internet Protocol, is an application thatenables data packet networks to transport real time voice traffic.VOIP uses the Internet as the transmission network. This paperdescribes VoIP and its requirements. The paper further discussesvarious VoIP protocol, security and its market.
 Keywords: VOIP; H.323; SIP; Security; Market;
VoIP, or Voice over IP, is an application that enables datapacket networks to transport real time voice traffic. It consistsof hardware and software that allows companies and persons toengage in telephone conversations over data networks. As aresult, more and more companies have become interested inimplementing VoIP [1].All VOIP services are not built alike. Some allow you tocall anyone with a phone, while others restrict your calls toonly other clients using the same VOIP service. You canchoose between three different ways to set up a VOIP systemon your computer.You can use an ATA (analog voice adaptor) which performsthe analog-to-digital conversion, and is plugged in to yourcomputer at one end and your telephone at the other.You can use an IP phone, a phone specifically made for usewith VOIP. While the IP phone looks exactly like a normalphone, it's got special Ethernet connectors that allow it to beplugged into your router. They're even working on WIFIphones for VOIP that you can take with you to the variousinternet hotspots popping up all over the world.Finally, you can make VOIP contact with your computer alone.Simply install the VOIP software, make sure you've got amicrophone, speakers, an internet connection (high-speed isbest, of course), and a sound card, and chat away.One thing many VOIP-users love about it is the cost, or moreaccurately, the savings. By using VOIP you save yourself oneunnecessary bill per month - your phone bill. VOIP charges,much cheaper usually than most people's phone bills, appear onyour regular broadband bill [2].The technology underpinning VoIP was initially developedin the late 1970s, but it took almost 20 years to evolve from acomputer novelty into a household service. It's now used byhundreds of thousands of people every day.VoIP works in a relatively simple way. Each time you makea phone call your voice is converted into a stream of data.Then, rather than being sent over the phone network, this datastream travels over your broadband internet connection.Each data packet is labelled with its destination address (theperson you're calling) and moves through the internet in thesame way as web pages and file downloads. When they get totheir destination, the packets are reassembled and convertedback into sound waves. When you have this process happeningsimultaneously in two directions, you've got a phone call.Most VoIP services also come with an allocated landlinephone number which allows other people to call you. In thesecases the call will be routed to the nearest handover point(called a POP or point of presence) and then travel over theinternet to your VoIP phone or computer [3].II.
VOIPObviously, the most important requirement is a broadbandinternet connection. Broadband connections are provided bycable companies (digital cable service), telephone companies(DSL, T1, etc.), and radio/microwave broadband internetconnections. Currently, satellite (ie., satellite uplink dish)internet connections are not compatible with VOIP equipmentbecause of the proprietary data compression algorithms used insatellite uplink and downlink. Further, the speed of light delayto and from a geosynchronous orbiting satellite would prove tobe very annoying people trying to talk.Broadband connection data uplink and downlink speeds of greater than 80 kilobits per second per telephone circuit (whilea call is in progress) are generally considered to be theminimum requirement for "decent" voice transmission quality.A "Telephone Adapter" (or "TA," and also known as an"Analog Telephone Adapter" or "ATA") is a piece of hardwarethat is used to digitize the voice and establish the IP session to
the internet phone company’s network switch. While it is possible to use a computer’s microphone and speakers and
special software for telephony over the Internet, the obviouslimitation that the computer has to be turned on to make orreceive phone calls makes this unwieldy.A TA eliminates the need for a computer to be up andrunning and accepts a standard 4-wire RJ11 telephone cable to
272http://sites.google.com/site/ijcsis/ISSN 1947-5500
(IJCSIS) International Journal of Computer Science and Information Security,Vol. 8, No. 2, 2010
support either premise telephone wiring or a direct connectionof a standard analog telephone.In addition, the TA usually includes a built in "router" thatprovides firewall isolation for the computers connecting to theInternet, as well as a Local Area Network (LAN) switch orhub. This allows efficient Internet connection sharing. Theinternet phone company usually provides or rent the TA, orthey can be purchased at retail for a reasonable price [4].The third requirement is a VoIP Service Provider (VSP)also known as an Internet Telephony Service Provider (ITSP).The Provider will supply you with an account and some formof "Telephone Number" [5].The final requirement is one or two common variety analogtelephone handset. Almost all commonly available wirelesstelephones and most two line phone sets will work with VOIP[4].
 All together 
First, confirm that you have the correct template for yourpaper size. This template has been tailored for output on theUS-letter paper size. If you are using A4-sized paper, please
close this file and download the file for “MSW A4 format”.
 The TA is usually a cable or DSL router that connects to the
cable company’s or DSL provider’s supplied terminal (or "modem"). The customer’s computer is then connected to the
TA, as are the one or two standard (RJ11) telephone cables,which connect to either a wall outlet or a standard analogtelephone, depending on whether telephone extensions arepresent or not. More than one computer usually can beconnected to the TA to create a Local Area Network (LAN).
Figure 1. A sample installation of VOIP.
The above diagram illustrates a sample installation. Thetelephone on the left is directly connected to the TA, while theone on the right is connected to the premise distribution wiringwhich is connected to the TA via RJ 11 cable to a wall jack.The key to making VOIP work is the correct initializationof the TA to work with the internet phone company andconfiguring IP addresses on the router to use for the computeror premise LAN connected computers sharing the broadbandconnection [6].III.
To deliver voice, two types of VOIP protocol used: H.323and SIP. H.323 and SIP both support VoIP and multimediacommunications. H.323 is an older standard developed by theITU. A good chunk of it is based on ISDN which comes fromthe traditional telephony world. H.323 is a binary protocol andis fairly complex in nature. SIP was developed by the InternetEngineering Task Force (IETF) and is text based (similar toHTTP). Much of the infrastructure already in place to supportHTTP has been adapted to support SIP. IT managers withinbusinesses are generally more comfortable with SIP becausethey are used to handling HTTP traffic. SIP is an open standardand solutions based on SIP are highly interoperable. A lot of effort has gone into ensuring interoperability and manymanufacturers work together to regularly test to ensure this.Very few manufacturers are working on new H.323implementations. SIP has become the standard of choice and isbeing worked on by large companies such as Microsoft andCisco [7].
 H.323 Protocol Overview
H.323 is a ITU recommendation based on the H.320 familyof standards. The current version of the recommendation isversion 4 [8]. Initially, the protocol (version 1) was designed toprovide signalling for a multimedia conferencing system forLAN environments with no quality of service provisions.However, in is current state, it has evolved into an umbrella of specifications that define the complete architecture andoperation of a multimedia conferencing system over a widearea packet network. In contrast to its original scope, it hasbecome a scalable solution that can be interworked withmanaged large scale networks.A H.323 system provides the necessary signalling andcontrol operations for performing multimedia communicationsover an underlying packet based network which may notprovide a guaranteed quality of service. The actual network interface, the physical network and the transport protocols usedon the network are not included in the scope of H.323. A H.323system comprises of the following entities: Terminals,Gatekeepers, Gateways, Multipoint Controllers, MultipointProcessors and Multipoint Control Units.
provide the audio/video/data communicationscapability in point-to-point or multipoint conferences, as wellas handling the H.323 signalling issues on behalf of the user.
provide admission control and addresstranslation services
273http://sites.google.com/site/ijcsis/ISSN 1947-5500
(IJCSIS) International Journal of Computer Science and Information Security,Vol. 8, No. 2, 2010
are needed to provide interworking withterminals using other signalling protocols, such as PSTNterminals, ISDN terminals, SIP terminals, etc.
 Multipoint Controllers, Multipoint Processors and  Multipoint Control Units
provide support for multipointconferences.A central aspect of H.323 is the
 H.323 call
. It is defined asthe point-to-point multimedia communication between twoH.323 endpoints. If the H.323 endpoint communicates with anendpoint which uses a different signalling protocol, then theH.323 call is defined as the call segment between the H.323entity and the gateway that provides interworking with theforeign network.The H.323 protocol is a tightly coupled family of subprotocols which must all interoperate in order to completesuccessfully a multimedia call session. The sub protocols aredescribed in ITU recommendations. The main ones are:
Sub protocol for messages exchanged betweenH.323 endpoints for setting up and tearing down a call as wellas for messages between an H.323 endpoint and its controllingH.323 entity, such as a gatekeeper.
Sub protocol for messages exchanged betweenendpoints in order to control the call session, exchangeresource capabilities and establish media channels.
Sub protocol for security and encryption for H.323terminals.
Sub protocols for supplementary services, such asCall Transfer, Call Park, Call Waiting etc [9].
SIP Protocol Overview
SIP, which stands for Session Initiation Protocol, is anIETF application layer control protocol, defined in RFC 2543[10], for the establishment, modification and termination of multimedia sessions with one or more participants. SIP makesminimal assumptions about the underlying transport andnetwork layer protocol, which can provide either a packet orbyte stream service with either reliable or unreliable service.A SIP system is based on a client/server model and iscomprised of the following logical entities:
• A
User Agent (UA)
is an application that acts on behalf of the user, both as a client (User Agent Client) and as a server(User Agent Server). As a client it initiates SIP requests and asa server it accepts calls and responds to SIP requests made byother entities. The user agent is usually part of a multimediaterminal whose media capabilities it controls without havingany media capabilities of its own.
 Registrar Server 
is a SIP server that accepts onlyregistration requests issued by user agents. A registrar servernever forwards requests.
• A
 Location Server 
is a server which provides informationto a proxy/redirect server about the possible current locationsof a user. Usually, this entity is part of the proxy/redirectservers.
 Redirect Server 
is a SIP server that provides addressmapping services. It responds to a SIP request destined to an
address with a list of new addresses. A redirect server doesn’taccept calls, doesn’t forward requests nor does it initiate any of 
its own.
• A
Proxy Server 
is a SIP server that acts both as a server touser agents by forwarding SIP requests and as a client to otherSIP servers by submitting the forwarded requests to them onbehalf of user agents or proxy servers.With the exception of the user agent, which is usually partof a multimedia terminal, the rest of the logical entities(registrar, redirect and proxy servers)a may be combined in asingle application. Therefore, a single entity can act either as aproxy or as a redirect server, according to the SIP request, andat the same time accept registration requests. A SIP call isdefined as the multimedia conference consisting of allparticipants invited by a common source.Although not partitioned formally, the SIP system can beviewed as divided into domains each serviced by oneredirect/proxy server and one registrar. A user agent hasusually a home domain, which is specified by its address, but itcan roam and use services in other domains as well, in which
case it is considered to be ’visiting’. Otherwise it is considered
 be ”at home” [9]
 Related Work- Comparison of two Protocols
The authors of Nortel Networks [11] conclude byrecommending SIP as their preference for a control protocol.They point out that even though H.323, unlike SIP, hascurrently more enterprise oriented and campus scale productsdeployed, SIP provides long term benefits which are related toand affect time to market, extensibility, multi-party serviceflexibility, ease of interoperability and complexity of development.The Dalgic and Fang [12] concluded that In terms of functionality and services that can be supported, H.323v3 andSIP are very similar. However, supplementary services inH.323 are more rigorously defined and therefore fewerinteroperability issues are expected to arise. Furthermore,H.323 has better compatibility among its different versions andbetter interoperability with the PSTN. The two protocls arecomparable in their QoS support (similar call setup delays, nosupport for resource reservation or class of service (QoS)setting), but H.323v3 will allow signaling of the requested
QoS. On the other hand, according to the paper, SIP’s primary
advantages are its flexibility to add new features and its relativeease of implementation and debugging. Finally, the authorsnote that H.323 and SIP are improving themselves by learningfrom each other, and the differences between them arediminishing with each new version.The Schulzrinne and Rosenberg [13] wrote that SIPprovides a similar set of services to H.323, but provides farlower complexity, rich extensibility, and better scalability.They point out that future work is due to more fully evaluatethe protocols, and examine quantitative performance metrics tocharacterize these differences. They also imply that a study
274http://sites.google.com/site/ijcsis/ISSN 1947-5500

Activity (5)

You've already reviewed this. Edit your review.
1 thousand reads
1 hundred reads
jdr114 liked this
Mawahib Bashir liked this
mg89 liked this

You're Reading a Free Preview

/*********** DO NOT ALTER ANYTHING BELOW THIS LINE ! ************/ var s_code=s.t();if(s_code)document.write(s_code)//-->