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Voice over Internet Protocol (VoIP) is a technology that allows you to make voice calls using a broadband Internet connection instead of a regular (or analog) phone line. It is a protocol optimized for the transmission of voice through the Internet or other packet switched networks. VoIP is also known as IP Telephony, Internet telephony, Broadband telephony, Broadband Phone and Voice over Broadband. When VoIP technology was first developed, many people were skeptical, mainly because it sounded too good to be true. However, as time has passed and this technology has proven itself time after time, these same people are now realizing that the future of VoIP technology is indeed solid. What actually makes VoIP so special over a conventional telephone network? VoIP can facilitate tasks and provide services that may be more difficult to implement or expensive using the more traditional Public Switched Telephone Network (PSTN). VoIP to VoIP phone calls are sometimes free, while VoIP to public switched telephone networks, PSTN, may have a cost that is borne by the VoIP user. How VoIP / Internet Voice Works? VoIP services convert your voice into a digital signal that travels over the Internet. VoIP can allow you to make a call directly from a computer, a special VoIP phone, or a traditional phone connected to a special adapter. Voice over IP protocols carry telephony signals as digital audio, typically reduced in data rate using speech data compression techniques, encapsulated in a data packet stream over IP. phone, and turning them into digital data that can be transmitted over the Internet.

If you've never heard of VoIP, get ready to change the way you think about longdistance 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 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.

fee, there is usually no charge for computer-to-computer calls, no matter the distance.

Types of VoIP Services:

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.

Analog Telephone Adapter

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 normal telephone networks. When a call is made between two parties, the connection is maintained for the duration of the call. Because you're connecting two points in both directions, the connection is called a circuit. This is the foundation of the Public Switched Telephone Network (PSTN). Here's how a typical telephone call works: 1. You pick up the receiver and listen for a dial tone (a connection is set up). 2

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. Computer-to-computer -- This is certainly the easiest way to use VoIP. All you need is the software, a microphone, speakers, a sound card and an Internet connection; preferably a fast one like you would get through a cable or DSL modem. Except for your normal monthly ISP

2. After dialing, call is routed through the switch at your local carrier to the party you are calling. 3. A connection is made between your telephone and the other party's line using several interconnected switches along the way. 4. When someone answers the call, the connection opens the circuit. 5. You talk for a period of time and then hang up the receiver, thus freeing the line. 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. If you look at a typical phone conversation, much of this transmitted data is wasted.

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. If we could remove the silent intervals, the file would be even smaller. Then, instead of sending a continuous stream of bytes (both silent and noisy), we send just the noisy bytes. Instead of circuit switching, 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:

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

The sending computer chops data into small packets, with an address on each one. 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. The nearby router sends the packet to another closer router and so on. When the receiving computer finally gets the packets, 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. 3

One of the hurdles that were overcome some time ago was the conversion of the analog audio signal your phone receives into packets of data. How it is that analog audio is turned into packets for VoIP transmission? The answer is codecs.

VoIP: Codecs
A codec (coder-decoder) converts an audio signal into compressed digital form for transmission and then back into an uncompressed audio signal for replay. It's the essence of VoIP. Codecs accomplish the conversion by sampling the audio signal several thousand times per second. For instance,

a G.711 codec samples the audio at 64,000 times a second. It converts each tiny sample into digitized data and compresses it for transmission. When the 64,000 samples are reassembled, the pieces of audio missing between each sample are so small that to the human ear, it sounds like one continuous second of audio signal. There are different sampling rates in VoIP depending on the codec being used:

64,000 times per second 32,000 times per second 8,000 times per second

A G.729A codec has a sampling rate of 8,000 times per second and is the most commonly used codec in VoIP.

Codecs use advanced algorithms to help sample, sort, and compress and packetize audio data. The CS-ACELP algorithm (CS-ACELP = conjugatestructure algebraic-code-excited linear prediction) is one of the most prevalent algorithms in VoIP. CS-ACELP organizes and streamlines the available bandwidth. Annex B is an aspect of CSACELP that creates the transmission rule, which basically states "if no one is talking, don't send any data." The efficiency created by this rule is one of the greatest ways in which packet switching is superior to circuit switching. It's Annex B in the CSACELP algorithm that's responsible for that aspect of the VoIP call. E.164 is the name given to the standard for the North American Numbering Plan (NANP). This is the numbering system that phone networks use to know where 4

to route a call based on the dialed numbers. A phone number is like an address: (313) 555-1212 313 = State 555 = City 1212 = Street address The switches use "313" to route the phone call to the area code's region. The "555" prefix sends the call to a central office, and the network routes the call using the last four digits, which are associated with a specific location. Based on that system, no matter where you're in the world, the number combination "(313) 555" always puts you in the same central office, which has a switch that knows which phone is associated with "1212." The challenge with VoIP is that IPbased networks don't read phone numbers based on NANP. They look for

IP addresses, which look like this: IP addresses correspond to a particular device on the network like a computer, a router, a switch, a gateway or a telephone. However, IP addresses are not always static. They're assigned by a DHCP server on the network and change with each new connection. VoIP's challenge is translating NANP phone numbers to IP addresses and then finding out the current IP address of the requested number. This mapping process is handled by a central call processor running a soft switch. The central call processor is hardware that runs a specialized database/mapping program called a soft switch. Think of the user and the phone or computer as one package -- man and machine. That package is called the endpoint. The soft switch connects endpoints. Soft switches know:

Protocols: As we've seen, on each end of a VoIP call we can have any combination of an analog, soft or IP phone as acting as a user interface, ATAs or client software working with a codec to handle the digital-to-analog conversion, and soft switches mapping the calls. How do you get all of these completely different pieces of hardware and software to communicate efficiently to pull all of this off? The answer is protocols. There are several protocols currently used for VoIP. These protocols define ways in which devices like codecs connect to each other and to the network using VoIP. They also include specifications for audio codecs. The most widely used protocol is H.323, a standard created by the International Telecommunication Union (ITU). H.323 is a comprehensive and very complex protocol that was originally designed for video conferencing. It provides specifications for real-time, interactive videoconferencing, data sharing and audio applications such as VoIP. H.323 is a large collection of protocols and specifications allowing it to be used for so many applications. The problem with H.323 is that it's not specifically tailored to VoIP. An alternative to H.323 is the Session Initiation Protocol (SIP) by the Internet Engineering Task Force (IETF). SIP is a more streamlined protocol, developed specifically for VoIP applications. Smaller and more efficient than H.323, SIP takes advantage of existing protocols because of its better traversal of NAT(Network Address Translation) and firewalls. Media Gateway Control Protocol (MGCP) is a third commonly used VoIP protocol that focuses on endpoint control. MGCP is geared toward features like call waiting.

Where the network's endpoint is What phone number is associated with that endpoint The endpoint's current IP address

VoIP: Soft Protocols



The soft switch contains a database of users and phone numbers. If it doesn't have the information it needs, it hands off the request downstream to other soft switches until it finds one that can answer the request. Once it finds the user, it locates the current IP address of the device associated with that user in a similar series of requests. It sends back all the relevant information to the softphone or IP phone, allowing the exchange of data between the two endpoints.

One of the challenges facing the worldwide use of VoIP is that these three protocols are not always compatible. VoIP calls going between several networks may run into a snag if they hit conflicting protocols. Since VoIP is a relatively new technology, this compatibility issue will continue to be a problem until a governing body creates a standard universal protocol for VoIP.

VoIP Call Monitoring

The greatest advantage of VoIP is price and the greatest disadvantage is call quality. For businesses that deploy VoIP phone networks, particularly those who operate busy call centers call quality issues are both inevitable and unacceptable. To fix call quality issues, most of them use a technique called VoIP call monitoring. VoIP call monitoring, also known as quality monitoring (QM), uses hardware and software solutions to test, analyze and rate the overall quality of calls. This is a key component of a business's overall quality of service (QoS) plan. Call monitoring hardware and software uses various mathematical algorithms to measure the quality of a VoIP call and generate a score called mean opinion score (MOS). The MOS is measured on a scale of 1 to 5, although 4.4 is technically the highest score possible for VoIP. A MOS of 3.5 or above is considered a "good call" To come up with the MOS, call monitoring hardware and software analyzes several different call quality parameters, the most common being:

round trip. Round-trip latency contributes to the "talk-over effect" experienced during bad VoIP calls, where people end up talking over each other because they think the other person has stopped speaking. A round-trip latency of over 300 milliseconds is considered poor. Jitter -- Jitter is latency caused by packets arriving late or in the wrong order. Most VoIP networks try to get rid of jitter with something called a jitter buffer that collects packets in small groups, puts them in the right order and delivers them to the end user all at once. VoIP callers will notice a jitter of 50 milliseconds or greater. Packet loss -- Part of the problem with a jitter buffer is that sometimes it gets overloaded and late-arriving packets get "dropped" or lost. Sometimes the packets will get lost sporadically throughout a conversation (random loss) and sometimes whole sentences will get dropped (burst loss). Packet loss is measured as a percentage of lost packets to received packets.

There are two different types of call monitoring: active and passive. Active call monitoring happens before a company deploys its VoIP network. It is often done by equipment manufacturers and network specialists who use a company's VoIP network exclusively for testing purposes. Passive call monitoring analyzes VoIP calls in real-time while they're being made by actual users. It can detect network traffic problems, buffer overloads and other glitches that network administrators can fix in network down time.

Latency -- This is the time delay between two ends of a VoIP phone conversation. It can be measured either one-way or 6

Another method for call monitoring is recording VoIP phone calls for later analysis. This type of analysis is limited to what can be heard during the call, not what's happening on the actual network. This type of monitoring is usually done by human beings, not computers, and is called quality assurance.

VoIP Cell Phones

VoIP-enabled cell phones are just entering the consumer market. Dualmode cell phones contain both a regular cellular radio and a Wi-Fi (802.11 b/g) radio. The Wi-Fi radio enables the cell phone to connect to a wireless Internet network through a wireless router and helps to make VoIP calls. Here's how it works:

1. When the cell phone is in range of a wireless Internet network, the phone connects to the network. 2. Any calls you initiate on the wireless network are routed through the Internet as VoIP calls. 3. If the phone is out of range of a wireless Internet signal, it automatically switches over to the regular cellular network and calls are charged as normal. 4. Dual-mode phones can hand off seamlessly from Wi-Fi to cellular (and vice versa) in the middle of a call as you enter and exit Wi-Fi networks.


Similar to dual-mode cell phones are Wi-Fi phones. Wi-Fi phones aren't technically cell phones because they only have a Wi-Fi radio, not a cellular radio. Wi-Fi phones look like cell phones, but can only make calls when connected to a wireless Internet network. That means all Wi-Fi phone calls are VoIP calls.

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. VoIP can facilitate tasks and provide services that may be more difficult to implement or expensive using the more traditional PSTN. Examples include: The ability to transmit more than one telephone call down the same broadband-connected telephone line. 3-way calling call forwarding, automatic redial, and caller ID; features that traditional telecommunication companies (Telco) normally charge extra for. Secure calls using standardized protocols (such as Secure Real-time Transport Protocol.) Most of the difficulties of creating a secure phone over traditional phone lines, like digitizing and digital transmission are already in place with VoIP. It is only necessary to encrypt and authenticate the existing data stream. Location independence. Only an internet connection is needed to get a connection to a VoIP provider. Integration with other services available over the Internet, including video conversation, message or data file exchange in parallel with the conversation.

data networks already understand the technology. By migrating to this technology, telephone networks immediately gain the ability to communicate the way computers do.

Disadvantages of Using VoIP

The current PSTN is a robust and fairly bulletproof system for delivering phone calls. On the other hand, computers, email and other related devices are still kind of flaky. 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 be one of the major flaws in VoIP: reliability.

Probably one of the most compelling advantages of packet switching is that 8

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 (unless it is a cordless) still works. With VoIP, no power means no phone. A stable power source must be created for VoIP. 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 consideration is that many other systems in your home may be integrated into the phone line. There's currently no way to integrate the products (Digital video recorders, digital subscription TV services and home security systems) with VoIP. Emergency 911 calls also become a challenge with VoIP. As stated before, VoIP uses IPaddressed phone numbers, not NANP phone numbers. There's no way to associate a geographic location with an IP address. To

fix this, perhaps geographical information could somehow be integrated into the packets. 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.

monthly surcharges. Cable companies too, have taken on VoIP and now offer faster, better broadband services. With the development of VoWiFi and VoLAN (voice over wireless LAN) even cell phones are becoming VoIP enabled. The future of VoIP doesnt show signs of slowing down either. As companies continue to develop phones, services and plans that incorporate this technology, the potential of VoIP will only continue to evolve.

When it comes to VoIP technology for corporations, small businesses, and individuals, long-distance phone calls, conferencing, e-mail, and other communications needs are now affordable and flexible. The future of VoIP technology is looking very promising as new dimensions are being explored and enhancements offered to customers. When VoIP technology was first developed, many people were skeptical, mainly because it sounded too good to be true. However, as time has passed and this technology has proven itself time after time, these same people are now realizing that the future of VoIP technology is indeed solid. Already VoIP technology has made advancements that are catching business owners interest. In addition to making long-distance calls all around the world for little to no money, this technology is always on the move, introducing new features. Over the last few years VoIP developers have sought new methods for optimizing phone quality and technology that would surpass traditional fixed-line capabilities. In doing so, they have been able to offer elimination of costly telephone bills for those running businesses and for those homeowners wanting to avoid extra 9

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