Despre Vo IP(ghid)

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VoIP este "capacitatea" telefoanelor sa apeleze si sa trimita faxuri prin retele de date bazate pe protocoale IP la un raport pret/castig superior si servicii de calitate superioara. Toata lumea vorbeste despre VoIP si fiecare vrea partea lui din prajitura.
Proiectantii de echipamente si constructorii au vazut in aceasta oportunitate un nou drum catre inovare si concurenta. S-au grabit sa dezvolte noi echipamente VoIP incercand in acelasi timp sa patrunda pe piata.
Furnizorii de servicii de internet disting posibilitatea unei competitii cu reteaua de telefonie publica (PSTN) pentru obtinerea clientilor.
Utilizatorii sunt interesati de integrarea vocii in conditiile unor costuri economice.
Cu toate ca acest concept -VoIP - este foarte atractiv, tehnologia nu a fost foarte bine dezvoltata astfel incat sa inlocuiasca serviciile si calitatea furnizata de PSTN. In primul rand, trebuie specificat faptul ca VoIP va creste costurile efective. Pentru a intra in competitie cu reteaua de telefonie existenta trebuie asigurat un pret foarte mic al serviciilor. Economia realizata este substantiala in cazul apelurilor pe distante foarte mari.
VoIP este o alternativa care poate concura furnizorii de servicii telefonice traditionale care vor imbunatati categoric serviciile pretutindeni In industrie.

Definitie
Telefonia prin internet se refera la serviciile de comnunicatii - voce, faxuri, aplicatii cu mesaje vocale - care sunt transportate prin internet mult mai repede decat prin reteaua de telefonie publica (PSTN). Primul pas implicat In producerea unui apel telefonic prin internet este conversia semnalului analogic de voce in format digital si compresia/transformarea in pachete ce pot fi transmise prin suita de protocoale TCP/IP; la primirea mesajului procesul se efectueaza in sens invers.
Generalitati
Acest ghid prezinta noul concept de comunicare - telefonia prin internet - care este inca in formare dar caruia i se preconizeaza o evolutie rapida.
Forta pietei determina aceasta evolutie si implicit beneficiile ce pot fi realizate de utilizatori. Sunt tratate, de asemenea, dificultatile ce trebuie trecute inainte ca telefonia prin internet sa fie adoptata pe scara larga.

1. Introducere
Posibilitatea comunicatiilor vocale prin intermediul retelei web, mult mai noua decat reteaua de telefonie publica, a devenit realitate In februarie 1995 cand Vocaltec Inc. a introdus propriul software - Internet Phone.
Proiectat sa ruleze pe un PC 486/33MHz echipat cu o placa de sunet, difuzoare, microfon si modem programul concentreaza semnalul vocal si il translateaza in pachete IP pentru transmiterea pe WEB. Aceasta telefonie prin internet de la calculator la calculator functioneaza numai daca ambele parti folosesc software-ul Internet Phone.
De atunci, telefonia prin internet a avansat rapid intr-o perioada relativ scurta de timp. In momentul de fata sunt multi producatori de software care ofera programe dedicate telefoniei prin internet, dar mult mai important, serverele portal actioneaza ca o intefata intre Internet si reteaua de telefonie publica.

Echipate cu placi de procesare a vocii, aceste servere permit utilizatorilor sa comunice prin telefoanele standard.O apelare traverseaza reteaua de telefonie publica spre cel mai apropiat server care transforma semnalul de voce (semnal analog) in semnal digital, il comprima in pachete IP si prin reteaua Internet iI transporta catre receptor.
Telefonia prin internet cu suportul sau de apelare calculator - telefon, telefon -calculator si telefon - telefon reprezinta un pas semnificativ catre integrarea vocii si retelelor de date.
Initial privita ca o noutate, telefonia prin internet este din ce in ce mai atractiva pentru utlizatori deoarece ofera o economie extraordinara a costurilor fata de reteaua de telefonie publica. Utilizatorii pot ocoli comisionarii de lunga distanta folosind telefonia prin internet in schimbul unei taxe lunare de acces la internet.

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Desi progreseaza rapid, telefonia prin Internet are anumite probleme in ceea ce priveste stabilitatea si calitatea sunetului datorate in primul rand limitarilor introduse de latimea de banda a transmisiei prin Internet si de tehnologiile curente de comprimare a datelor.
Majoritatea corporatiilor sunt interesate de cresterea calitatii serviciilor, si de reducerea facturilor telefonice; astfel, acestea limiteaza aplicatiile telefoniei prin Internet la propriile retele intranet. Retelele Intranet, avand latimea de banda anticipata superioara retelei publice Internet, pot suporta transmisii full-duplex si comunicatii vocale in timp real. Cele mai multe companii isi limiteaza traficul comunicatiilor vocale prin retea la aplicatii asincrone half-duplex (mesagerie vocala).
Telefonia prin internet aplicata in retele intranet asigura utilizatorilor economii importante in cazul apelurilor intre orase situate la mare distanta unul de altul; pot efectua apeluri punct-la-punct prin intermediul serverelor portal atasate retelelor locale (LAN), fara a fi necesare programe specializate sau conturi Internet.
De exemplu, utilizatorul A din New York doreste sa apeleze (punct-la-punct) un utilizator B situat la sediul companiei din Geneva. Ridica receptorul si apeleaza o extensie pentru a se conecta la serverul portal echipat cu o placa ce asigura transmisia comunicatiilor vocale si un program ce asigura compresia si conversia acestora. Server-ul configureaza centrala telefonica particulara (PBX) care digitizeaza convorbirea telefonica.
Apoi, apeleaza numarul biroului din Londra si server-ul transmite convorbirea (sub forma de pachete IP), prin intermediul protocoalelor IP ale retelei WAN, catre portalul din Geneva. Portalul din Geneva converteste semnalul digital in semnal analogic si il transmite utlizatorului B.
Aceasta versiune a telefoniei prin Internet asigura de asemenea transmiterea comunicatiilor vocale (digitizate) si traficul de date prin intranet avand ca suport aplicatii partajate.

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3. Bariere tehnice
Obiectivul final al telefoniei prin Internet este fiabilitatea si obtinerea unor servicii vocale de calitate superioara, in conditii similare sau chiar superioare serviciilor oferite de reteaua de telefonie publica. In acest moment, aceste caracteristici nu sunt obtinute in reteaua Internet in primul rand datorita limitarilor latimii de banda, care influenteaza pierderea pachetelor de informatii. In comunicatiile vocale, pierderea pachetelor de informatii apare sub forma unor pauze sau a unor perioade de liniste in timpul conversatiei. Efectul final obtinut este similar unui discurs cu intreruperi care este nesatisfacator pentru majoritatea utilizatorilor si inacceptabil in domeniul afacerilor.
Reteaua Internet este o colectie formata dintr-un numar impresionant de retele (peste 130.000) care isi mareste continuu popularitatea, tinand cont de faptul ca lunar peste 1 milion de utilizatori noi acceseaza site-uri din retea. Cresterea considerabila a utilizarii Internetului conduce la aglomerarea frecventa a traficului, deci la intarzierea transmiterii informatiilor. Astfel de intarzieri sunt determinate de pierderea sau schimbarea pachetelor de informatii.
In plus, datorita faptului ca Internetul este o retea fara conexiune sau cu comutare a pachetelor, pachetele individuale, pentru fiecare semnal vocal, parcurg linii de retea separate astfel incat la destinatie sa poata fi reasamblate corect.
Fiabilitatea retelei si calitatea sunetului depind de tehnicile folosite pentru codificarea si procesarea vocii in serverele portal.
Majoritatea dezvoltatorilor de software pentru telefonia prin Internet precum si furnizorii de servere folosesc o mare varietate de protocoale de transmitere a datelor, diferiti algoritmi de codificare a semnalelor - cu frecvente diferite de transmitere a datelor si diferite mecanisme de reconstituire a semnalelor vocale. Acest lucru afecteaza vizibil claritatea si fidelitatea transmiterii sunetelor pe Internet. Datorita absentei unor protocoale standard multe din produsele din domeniul telefoniei prin Internet nu pot fi utilizate si in alte sisteme sau chiar in reteaua de telefonie publica.

4. Standarde
In urmatorii ani se vor aduce imbunatatiri substantiale in reteaua Internet in privinta transmiterii datelor a sunetelor sau al imaginilor video. Optimizarea retelei vizeaza in primul rand eliminarea blocarilor si a pierderii pachetelor de informatii. Problemele importante cum sunt fiabilitatea sau calitatea sunetului prin retea vor fi rezolvate gradual prin adoptarea unor standarde. Eforturile de stabilire a acestor standarde sunt concentrate in trei directii importante ale telefoniei prin Internet: codificarea/decodificarea audio, protocoalele de transport a datelor si serviciile de conducere.
In mai 1996, Uniunea Internationala a Telecomunicatiilor (UIT) a ratificat hotararea H.323 care defineste modul in care traficul datelor, al vocii sau al imaginilor video va fi transmis prin protocoale IP in retelele locale. Aceasta hotarare include si standardul T.120 . Recomandarea se bazeaza pe suita de protocoale RTP/RTCP (real-time protocol/ real-time control protocol) pentru administrarea semnalelor audio si video.
De asemenea, H.323 se refera la aplicatiile importante din domeniul telefoniei prin Internet, prin definirea unui trafic ce asigura o intarziere minima atat in transmiterea sunetului cat si a imaginilor, prin stabilirea unor prioritati in transportul datelor, in vederea asigurarii unor servicii de comunicare in timp real pe Internet.
Specificatiile din H.323 definesc transmiterea datelor, a vocii si a imaginilor video prin retele telefonice obisnuite, in timp ce H.320 stabileste protocoalele pentru transmiterea vocii, a datelor sau a imaginilor prin retele digitale cu servicii integrate. G.729 este una din recomandarile cuprinse in H.323 ratificata in noiembrie 1995 si care se refera la formatul codificarii/decodificarii audio.
La Forumul VoIP din martie 1997 a fost votat un nou set de specificatii G.723.1.
Consortiul industial condus de Intel si Microsoft au fost de acord sa sacrifice calitatea sunetului in schimbul cresterii latimii de banda - G.723.1 necesita 6,3 kb/s in timp ce G.729 necesita 7,9 kb/s. Adoptarea unui standard pentru codificarea si decodificarea semnalelor audio este un pas important care va permite cresterea fiabilitatii si a calitatii sunetului, mai ales in cazul retelelor intranet sau in cazul conexunilor IP punct-la-punct.
Protocolul RTP, pe care se bazeaza H.323, este un protocol nou folosit in aplicatiile de timp real, iar echipamentele vor include mecanisme de control pentru sincronizarea traficului de date in retea.
Este un protocol standard al carui scop este imbunatatirea traficului de date pe Internet in timp real, prin crearea unor linii de transmitere a datelor dedicate unor sesiuni specifice. Protocolul de rezervare a resurselor va fi implementat in rutere pentru stabilirea si intretinerea liniilor de transmitere a informatiilor si obtinerea unui nivel ridicat al calitatii serviciilor.

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5. Viitorul VoIP
Dezvoltarea produselor si serviciilor VoIP va fi influentata de diferiti factori.
Cele mai promitatoare domenii pentru VoIP sunt retelele intranet si retelele comerciale extranet. Infrastructurile acestor retele bazate pe IP permit operatorilor sa verifice daca exista utilizatori care pot sau nu pot folosi reteaua.
Un alt element important in evolutia telefoniei prin Internet este serverul VoIP. Aceste servere au evoluat de la platformele bazate pe calculatoare personale catre sisteme embedded robuste, capabile sa trateze sute de apeluri simultan. Prin urmare, marile companii vor folosi aceste tipuri de echipamente pentru a reduce cheltuielile asociate transmiterii vocii, faxurilor sau videoconferintelor prin IP. Noul drum deschis de telefonia prin Internet - transmiterea vocii, a datelor si a imaginilor video prin retele bazate pe IP - va atrage companiile spre aceasta directie, IP va actiona in primul rand ca un agent unificator.
Retelele comerciale extranet vor permite transmiterea faxurilor si a vocii prin protocoale IP catre publicul larg. Prin garantarea unor parametrii specifici si prin asigurarea interoperabilitatii serviciilor, aceste retele vor constitui un suport sigur si de incredere pentru diferite aplicatii din comertul electronic.
In viitor, Internetul va asigura servicii vocale si video fiabile datorita cresterii substantiale a vitezei de accesare a site-urilor, dezvoltarii continue a retelelor IP, ATM, SONET (retele optice sincrone) si ISDN (retele digitale cu servicii integrate), a modemurilor, a tehnologiei DSL.
Produsele si serviciile pentru transmiterea faxurilor prin reteaua Internet vor deveni viabile, din punct de vedere economic, mult mai rapid decat cele pentru transmiterea vocii si a imaginilor video, deoarece, in acest caz, blocarile tehnice in traficul datelor sunt minime. In urmatorii doi ani, marile companii vor trece la transmiterea faxurilor prin Internet, intr-o prima faza prin intemediul serverelor FAXoIP si apoi direct din aparate capabile sa transmita faxuri via IP. Standardele pentru transmiterea faxurilor prin IP vor apare la sfarsitul acestui an.
In viitor, videoconferintele prin IP vor deveni metoda generala folosita in comunicatii, iar camera video va deveni unul din perifericele standard ale unui calculator cu facilitati complexe multimedia.

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AVoIP este rezultatul unei provocari. Avand in reteaua de telefonie publica si Internet doua sisteme de comunicatie foarte puternice, ideea de a uni aceste doua sisteme a fost primul pas in dezvoltarea conceptului VoIP - transmiterea vocii si a faxurilor prin protocoale IP. Aplicatiile de transmitere a faxurilor prin protocoale Internet permit integrarea aparatelor FAX standard in retelele de date. Atat in transmiterea semnalelor vocale cat si in transmiterea faxurilor, procedura este aceeasi: semnalele analogice - voce, imagine - sunt convertite in semnale digitale si comprimate in pachete ce pot fi transmise in retele prin suita de protocoale TCP/IP.

Consideratii generale
Organizatiile din intreaga lume cauta sa-si reduca costurile datorate comunicatiilor. Prin urmare posibilitatea integrarii vocii, a faxurilor si a datelor intr-un singur sistem a devenit una din prioritatile administratorilor de retele. Organizatiile cauta solutii care le vor asigura obtinerea unor avantaje prin utilizarea retelelor cu banda larga pentru transmiterea vocii a faxurilor si a datelor, precum si utilizarea Internetului si a retelelor locale Intranet ca alternative in vederea realizarii unor cheltuieli medii.
Transmiterea vocii, a faxurilor si a datelor prin intermediul retelelor cu banda larga, precum si utilizarea Internetului si a retelelor locale Intranet ca alternative, vor asigura economii importante pentru majoritatea organizatiilor.
Acest curs prezinta principiile de realizare a sistemelor de transmitere in timp real a vocii si a faxurilor prin protocoale IP. Vor fi prezentate elementele principale ale programelor si echipamentelor folosite pentru transmiterea vocii si a faxurilor sub forma pachetelor de retea. Sistemele embedded folosite asigura obtinerea unor costuri reduse atat pentru producatori, din punct de vedere al preturilor de productie, cat si pentru utilizatori.
Cursul se refera la o categorie generala de retele, ce contin echipamente software si hardware care permit regim asincron de trasfer (ATM) si protocale Internet /Intranet (IP) pentru transmiterea vocii si a faxurilor.
Uniunea Internationala a Telecomuniatiilor a definit protocoalele standard pentru transmiterea faxurilor sub forma pachetelor de retea, principiile descrise putand fi aplicate si in retelele ATM.

Glosar
ATM - regim de transfer asincron
FAXoIP- transmiterea faxurilor prin protocolale Internet
IP - protocol de transmitere a datelor in reteaua Internet
ISDN - Retea digitala cu servicii integrate
ITU - Uniunea Internationala a Telecomuniatiilor
KBPS - kilo bytes pe secunda
LAN - retea locala de calculatoare
MHz - megaherti
PBX - centrala telefonica privata
PC - calculator personal
PSTN - retea telefonica comutata publica
QoS - calitatea serviciilor
RTCP - protocol de control in timp real
RTP - protocol in timp real
SONET - retele optice sincrone
VoIP - transmiterea vocii prin protocolale Internet
VPN - retea particulara virtuala
WAN - retele de arie larga

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Aplicatii VoIP

O mare varietate de aplicatii pot fi realizate folosind facilitatile oferite de retelele care permit transmiterea vocii prin protocoale IP.
In configuratia unei retele dintr-o organizatie (de exemplu o banca), al carui obiectiv este reducerea costurilor prin combinarea traficului de date si voce prin retea intre sediul central si diferite sectoare (sucursale). Acest lucru este posibil utilizand o retea cu comutare de pachete (packet network) care asigura o transmitere de date standard simultan cu transmiterea semnalelor vocale.
Aceasta configuratie de retea, datorita largimii reduse a benzii de transmisie disponibila pentru aceasta aplicatie, va aduce beneficii reale daca semnalele vocale sunt comprimate. VoIP asigura functii de interactiune (interworking function-IWF) care permit implementarea echipamentelor hardware si software ce asigura transmiterea combinata a vocii si a datelor prin retele cu comutare de pachete. Interfetele IWF sunt interfete analogice, in acest caz, si se conecteaza direct la telefon. IWF trebuie sa simuleze functionarea unei centrale telefonice particulare atat pentru terminalele telefonice ale sucursalelor cat si pentru cele existente in sediul central al organizatiei. IWF realizeaza acest lucru prin implementarea programelor de semnalizare care efectueaza aceste functiuni.
Pt o aplicatie de trafic interurban de date si semnale vocale,aici, una din organizatii transmite semnalele vocale intre doua asezari prin intermediul unei retele.cu comutare de pachete Aceasta aplicatie necesita interfata IWF pentru a asigura transmisia prin canale digitale de mare capacitate cum ar fi T1/E1 de 1,544 sau 2,048 MB/s. IWF simuleaza functionarea unei centrale telefonice private obtinandu-se astfel o economie importanta a costurilor aferente comunicatiilor.
-VoIP permite interconexiunea cu retelele de telefonie mobila...Pentru transmiterea semnalelor vocale de la telefoanele mobile prin intermediul retelelor digitale de telefonie mobila acestea sunt deja comprimate. Retelele cu comutare de pachete pot transmite astfel semnalele vocale receptionate de la telefoanele mobile in conditii optime realizand o economie importanta a largimii de banda. IWF asigura functiile de convertire a codurilor necesare pentru codificarea semnalelor vocale transmise prin intermediul telefoanele mobile in formatul necesar retelelor de telefonie publice.

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How VoIP Works


Introduction in how this works

If you've never heard of VoIP, get ready to change the way you think about long-distance phone calls. VoIP, or Voice over Internet Protocol, is a method for taking analog audio signals,like the kind you hear when you talk on the phone, and turning them into digital data that can be transmitted over the Internet.
How is this useful? VoIP can turn a standard Internet connection into a way to place free phone calls. The practical upshot of this is that by using some of the free VoIP software that is available to make Internet phone calls, you're bypassing the phone company (and its charges) entirely.Above all else, VoIP is basically a clever "reinvention of the wheel."
We'll explore the principles behind VoIP, its applications and the potential of this emerging technology, which will more than likely one day replace the
traditional phone system entirely.

The interesting thing about VoIP is that there is not just one way to place a call. There are three different "flavors" of VoIP service in common use today:

• ATA -- The simplest and most common way is through the use of a device called an ATA (analog telephone adaptor). The ATA allows you to connect a standard phone to your computer or your Internet connection for use with VoIP. The ATA is an analog-to-digital converter. It takes the analog signal from your traditional phone and converts it into digital data for transmission over the Internet. Providers like Vonage and AT&T CallVantage are bundling ATAs free with their service. You simply crack the ATA out of the box, plug the cable from your phone that would normally go in the wall socket into the ATA, and you're ready to make VoIP calls. Some ATAs may ship with additional software that is loaded onto the host computer to configure it; but in any case, it's a very straightforward setup.
• IP Phones -- These specialized phones look just like normal phones with a handset, cradle and buttons. But instead of having the standard RJ-11 phone connectors, IP phones have an RJ-45 ETHERNET connector. IP phones connect directly to your router and have all the hardware and software necessary right onboard to handle the IP call.WI-FI phones allow subscribing callers to make VoIP calls from any WE-FI hot spot.
• Computer-to-computer -- This is certainly the easiest way to use VoIP. You don't even have to pay for long-distance calls. There are several companies offering free or very low-cost software that you can use for this type of VoIP. All you need is the software, a microphone,spekers,a soundcard and a internet connection, preferably a fast one like you would get through a cable or DSL modem.

Using VoIP

Chances are good you're already making VoIP calls any time you place a long-distance call. Phone companies use VoIP to streamline their networks. By routing thousands of phone calls through a circuit switch and into an IP gateway, they can seriously reduce the bandwidth they're using for the long haul. Once the call is received by a gateway on the other side of the call, it's decompressed, reassembled and routed to a local circuit switch.
Although it will take some time, you can be sure that eventually all of the current circuit-switched networks will be replaced with packet-swicthing technology(more on packet switching and circuit switching later). IP telephony just makes sense, in terms of both economics and infrastructure requirements. More and more businesses are installing VoIP systems, and the technology will continue to grow in popularity as it makes its way into our homes.
With VoIP, you can make a call from anywhere you have broadband connectivity. Since the IP phones or ATAs broadcast their info over the Internet, they can be administered by the provider anywhere there's a connection. So business travelers can take their phones or ATAs with them on trips and always have access to their home phone. Another alternative is the softphone. A softphone is client software that loads the VoIP service onto your desktop or laptop. As long as you have a headset/microphone, you can place calls from your laptop anywhere in the broadband-connected world.


Most VoIP companies provide the features that normal phone companies charge extra for when they are added to your service plan. VoIP includes:

• Call waiting
• Call transfer
• Repeat dial
• Return call
• Three-way calling
• Caller ID

There are also advanced call-filtering options available from some carriers. These features use caller ID information to allow you make a choice about how calls from a particular number are handled. You can:

• Forward the call to a particular number
• Send the call directly to voice mail
• Give the caller a busy signal
• Play a "not-in-service" message
• Send the caller to a funny rejection hotline

With many VoIP services, you can also check voice mail via the Web or attach messages to an e-mail that is sent to your computer or handheld. Not all VoIP services offer all of the features above.


VoIP: Circuit Switching


Existing phone systems are driven by a very reliable but somewhat inefficient method for connecting calls called circuit switching.
Circuit switching is a very basic concept that has been used by
telephone networks.
Here's how a typical telephone call works:

1. You pick up the receiver and listen for a dial tone. This lets you know that you have a connection to the local office of your telephone carrier.
2. You dial the number of the party you wish to talk to.
3. The call is routed through the switch at your local carrier to the party you are calling.
4. A connection is made between your telephone and the other party's line using several interconnected switches along the way.
5. The phone at the other end rings, and someone answers the call.
6. The connection opens the circuit.
7. You talk for a period of time and then hang up the receiver.
8. When you hang up, the circuit is closed, freeing your line and all the lines in between.

Telephone conversations over today's traditional phone network are somewhat more efficient and they cost a lot less. Your voice is digitized, and your voice along with thousands of others can be combined onto a single fiber optic cable for much of the journey (there's still a dedicated piece of copper wire going into your house, though). These calls are transmitted at a fixed rate of 64 kilobits per second (Kbps) in each direction, for a total transmission rate of 128 Kbps. Since there are 8 kilobits (Kb) in a kilobyte (KB), this translates to a transmission of 16 KB each second the circuit is open, and 960 KB every minute it's open. In a 10-minute conversation, the total transmission is 9,600 KB, which is roughly equal to 10 megabytes.


VoIP: Packet Switching




A packet-switched phone network is the alternative to circuit switching. It works like this: While you're talking, the other party is listening, which means that only half of the connection is in use at any given time. Based on that, we can surmise that we could cut the file in half, down to about 4.7 MB, for efficiency. Plus, a significant amount of the time in most conversations is dead air -- for seconds at a time, neither party is talking. If we could remove these silent intervals, the file would be even smaller.
Data networks do not use circuit switching. Your Internet connection would be a lot slower if it maintained a constant connection to the web page you were viewing at any given time. Instead, data networks simply send and retrieve data as you need it. And, instead of routing the data over a dedicated line, the data packets flow through a chaotic network along thousands of possible paths. This is called packet switching. While circuit switching keeps the connection open and constant, packet switching opens a brief connection -- just long enough to send a small chunk of data, called a packet, from one system to another. It works like this:

• The sending computer chops data into small packets, with an address on each one telling the network devices where to send them.
• Inside of each packet is a payload. The payload is a piece of the e-mail, a music file or whatever type of file is being transmitted inside the packet.
• The sending computer sends the packet to a nearby router and forgets about it. The nearby router send the packet to another router that is closer to the recipient computer. That router sends the packet along to another, even closer router, and so on.
• When the receiving computer finally gets the packets (which may have all taken completely different paths to get there), it uses instructions contained within the packets to reassemble the data into its original state.

Packet switching is very efficient. It lets the network route the packets along the least congested and cheapest lines. It also frees up the two computers communicating with each other so that they can accept information from other computers, as well.






Advantages of Using VoIP


VoIP technology uses the Internet's packet-switching capabilities to provide phone service. VoIP has several advantages over circuit switching. For example, packet switching allows several telephone calls to occupy the amount of space occupied by only one in a circuit-switched network. Using PSTN, that 10-minute phone call we talked about earlier consumed 10 full minutes of transmission time at a cost of 128 Kbps. With VoIP, that same call may have occupied only 3.5 minutes of transmission time at a cost of 64 Kbps, leaving another 64 Kbps free for that 3.5 minutes, plus an additional 128 Kbps for the remaining 6.5 minutes. Based on this simple estimate, another three or four calls could easily fit into the space used by a single call under the conventional system. And this example doesn't even factor in the use of data compression which further reduces the size of each call. Let's say that you and your friend both have service through a VoIP provider. You both have your analog phones hooked up to the service-provided ATAs. Let's take another look at that typical telephone call, but this time using VoIP over a packet-switched network:

1. You pick up the receiver, which sends a signal to the ATA.
2. The ATA receives the signal and sends a dial tone. This lets you know that you have a connection to the Internet.
3. You dial the phone number of the party you wish to talk to. The tones are converted by the ATA into digital data and temporarily stored.
4. The phone number data is sent in the form of a request to your VoIP company's call processor. The call processor checks it to ensure that it's in a valid format.
5. The call processor determines to whom to map the phone number. In mapping, the phone number is translated to an IP address.The soft switch connects the two devices on either end of the call. On the other end, a signal is sent to your friend's ATA, telling it to ask the connected phone to ring.
6. Once your friend picks up the phone, a session is established between your computer and your friend's computer. This means that each system knows to expect packets of data from the other system. In the middle, the normal internet infrastructure handles the call as if it were e-mail or a web page. Each system must use the same protocol to communicate. The systems implement two channels, one for each direction, as part of the session.
7. You talk for a period of time. During the conversation, your system and your friend's system transmit packets back and forth when there is data to be sent. The ATAs at each end translate these packets as they are received and convert them to the analog audio signal that you hear. Your ATA also keeps the circuit open between itself and your analog phone while it forwards packets to and from the IP host at the other end.
8. You finish talking and hang up the receiver.
9. When you hang up, the circuit is closed between your phone and the ATA.
10. The ATA sends a signal to the soft switch connecting the call, terminating the session.

Probably one of the most compelling advantages of packet switching is that data networks already understand the technology. By migrating to this technology, telephone networks immediately gain the ability to communicate the way computers do.
It will still be at least a decade before communications companies can make the full switch over to VoIP.


Disadvantages of Using VoIP



The current Public Switched Telephone Network is a robust and fairly bulletproof system for delivering phone calls. Phones just work, and we've all come to depend on that. On the other hand, computers, e-mail and other related devices are still kind of flaky. Let's face it -- few people really panic when their e-mail goes down for 30 minutes. It's expected from time to time. On the other hand, a half hour of no dial tone can easily send people into a panic. So what the PSTN may lack in efficiency it more than makes up for in reliability. But the network that makes up the Internet is far more complex and therefore functions within a far greater margin of error. What this all adds up to is one of the major flaws in VoIP: reliability.
First of all, VoIP is dependant on wall power. Your current phone runs on phantom power that is provided over the line from the central office. Even if your power goes out, your phone still works. With VoIP, no power means no phone. A stable power source must be created for VoIP.
Another consideration is that many other systems in your home may be integrated into the phone line.Digital video recorders,digital subscription
TV services and home security systems all use a standard phone line to do their thing. There's currently no way to integrate these products with VoIP. The related industries are going to have to get together to make this work.Emergency 911 calls also become a challenge with VoIP. As stated before, VoIP uses IP-addressed phone numbers, not NANP phone numbers. There's no way to associate a geographic location with an IP address. So if the caller can't tell the 911 operator where he is located, then there's no way to know which call center to route the emergency call to and which EMS should respond. To fix this, perhaps geographical information could somehow be integrated into the packets.

• Because VoIP uses an Internet connection, it's susceptible to all the hiccups normally associated with home broadband services. All of these factors affect call quality:
  
  • Latency
  • Jitter
  • Packet loss
  
  Phone conversations can become distorted, garbled or lost because of transmission errors. Some kind of stability in Internet data transfer needs to be guaranteed before VoIP could truly replace traditional phones.
  
• VoIP is susceptible to worms,viruses and hacking, although this is very rare and VoIP developers are working on VoIP encryption to counter this.

Another issue associated with VoIP is having a phone system dependant on individual.