Virtual Call Centers O Voip Networks example essay topic

Voice over Internet Protocol Before we begin our discussion on transporting voice over a data networks, it is important to understand the concept and terms associated with a traditional telephone network. One of the most common piece of telecommunication equipment used today is the telephone. When you plug an ordinary analog telephone into the wall jack installed by the local telephone company, you are connected to the telephone network and are able to place calls. 1), but how is this process performed? In early telephone networks, call completion was dependent on an operator to complete the calls. When you wanted to place a call you would pick up the receiver and be connected to an operator at the Central Office (CO).

The CO is the local telephone office which all local lines connect too and where circuit switching of subscriber lines occurs. The operator would determine which two lines need to be connected for the parties to talk and connect the lines. To connect the two lines the operator would plug a cord connecting the two ports on a cord board to bridge the two ports together. The cord board used by the operator to connect the two parties was the first example of an early switching system. In today's telephone networks a CO switch or a private branch exchange (PBX) provides the same service once performed by an operator. A PBX is a premise switching system, serving a commercial or government organization, and usually located on that organization's premise.

PBX's provide telecommunication services at the location and access to public and private telecommunications network services. The PBX switch is connected to a similar switch located at the CO. Now we will discuss some of the key components of any telephone network. These components are loops, lines and trunks. "Loop transmission facilities connect switching systems to customer premises equipment throughout the serving area. A loop is a transmission path between a customer's premises and a local exchange carrier (LEC) central office" (Loop Transmission).

A line is a communication connection between your telephone and the local phone company's switch, such as a PBX or a CO switch. No that we know that a line connects a telephone to the switch, the next stage is the trunk. A trunk is a shared communication channel that connects multiple telephone switches together and has the ability to transfer a telephone call from one location to the next. It consists of wires that are twisted together to minimize the electromagnetic radiation created by the current flowing through them. "A trunk is a communications path connecting two switching systems used to establish end-to-end connections between customers (Trunks)". A trunk assigns connections case-by-case when a number is dialed.

When using a PBX everyone has a telephone line connected to the PBX, but the actual connection is only established when in use. When there is a need to access an outside line, an access code is dialed, such as 9 and the PBX connects the telephone to an outside line to the Public Switched Telephone Network (PSTN). Another component of any telephone network is the transmission media. The basic transmission media are: twisted-pair cable, coaxial cable, radio frequencies, and fiber. Twisted-pair cabling is made up of two insulated wires twisted in a spiral pattern. The wires can be shielded twisted-pair (STP) or unshielded twisted-pair (UTP).

Twisted-pair cable is common in telecommunications networks to carry analog signals from the user to other parts of the network. Integrated Services Digital Network (ISDN) is an international communication standard offered by telephone companies for sending data, voice, video, and other traffic. Both ISDN and digital subscriber line (DSL) can carry digital signals at a higher rate across the same twisted-pair cabling used by a standard telephone. DSL includes a variety of technologies being used to get higher digital bandwidth out to the customer's location. Another transmission media is cellular technology, which use radio waves for network access. Radio gives you the convenience of not having to install wiring in the ground to provide the service.

Inside the network, radio is usually used with microwave technology such as Local Microwave Distribution System (LMDS) and Multipoint Microwave Distribution System (MMDS) to carry higher-rate digital systems on a line-to-site basis between points. Fiber Optics is a method of transmitting data in the form of laser light over bundles of glass fibers. This method has numerous advantages over such traditional techniques using twisted pair wires, coax, and radio waves. Fiber optics offers greater bandwidth and is virtually unsusceptible to electromagnetic interference. The North America format standard for transmission over fiber is typically Synchronous Optical Network, or SONET.

SONET has the ability to deliver very high speeds, up to 2.5 Gbps. Types and Uses of Various Transmission Media Media User to Network Network to Network Speed Twisted Pair Analog voice ISDN T 1 / E 1 Digital subscriber line (xDSL) T 1 E 1 1.544 M pbs / 2.048 Mbps Coaxial Cable Cable TV T 3, T 4 E 3 44.736 / 34.368 Mbps Radio Cellular WLL, LMDS, and MMDS T 3, T 4 E 3 44.73 Mbps 34.368 Mbps Fiber SONET Cable Television SONET 2.5 Gbps For a telephone call to be completed, several forms of signaling must occur. In a voice network signaling is used to establish a connection. A voice connection usually begins when the phone is taken off hook, which sends a type of signal called an access signal. The access signal determines when a line is off hook or on hook. Foreign Exchange (FX) trunk signaling is another form of signaling that can be provided over analog or T 1 lines.

FX is a term used in a trunk network that has access to a distant CO. FX uses either loop-start or ground-start signaling methods. A loop-start is a type of access signal that is used in a PSTN. Ground-start is another access signaling method used on trunk lines or tie lines between PBXs to indicate on-hook / off -hook status to the CO. A Foreign Exchange Station (FXS) is how standard residential phone lines are configured for signaling. A FXS interface connects basic devices such as phones, modems, and faxes and must provide voltage, ring generation, off-hook detection, and call progress indicators.

Foreign Exchange Office (FXO) is a type of signaling that is used primarily to communicate with CO switching equipment or PBXs. There are two forms of technology that can be used to transmit voice across a traditional telephony network, analog and digital. In the PSTN, the person speaking into the phone modulates the signal via the mouthpiece. The caller's voice changes the frequency and the amplitude of the electric current flowing in the telephone circuit, thereby imparting a signal. The alternating electric current is represented by a series of sine waves which fluctuate over time. In analog transmission over the PSTN, the input voice signal and any noise on the line is used to generate the signal for transmission.

The quality of the voice signal depreciates over time and distance as the noise accumulates on the path. While analog transmission allows the amplification of the signal at points along the path, the system cannot differentiate between voice and noise. As a result, the signal-to-noise ratio (SNR) decreases because amplified noise obscures the voice signal at the receiver's end. A digital signal consists of individually distinct states that can easily be encoded to binary values forming a series of 1's and 0's for transmission across a network. The structure of a digital waveform is described as a square wave because its form is defined by instantaneous transitions between discrete levels.

The discrete nature of the digital signal and its binary format transmission means that digital signals are not subject to the interference, signal loss, and noise of analog signals. As long as the stream of bits gets to its destination, it can be reconstructed into a perfect replica of the original source. Voice over Internet Protocol, commonly called Voice over IP, or simply VoIP is the ability to make telephone calls and send faxes over IP-based data networks with a suitable quality of service (QoS) and superior cost / benefit. In its simplest form, when a user picks-up their phone to make a call, the call audio travels over the Internet instead of through Plain Old Telephone Service (POTS) lines.

This is where the main difference lies. Whereas the traditional phone line is physically connected to the phone company's CO, a VoIP call bypasses this altogether, and routes directly through the Internet. Calls travel from the user premise via their Internet connection to their Internet Telephony Service Provider (ITSP), which then connects them to the Public Switched Telephone Network (PSTN), where the call is usually terminated via POTS. VoIP enables routers and switches to carry telephony-style voice traffic, that is, live, packet ized voice traffic such as telephone calls over IP-based data rather than the PSTN. Since VoIP is IP based it is considered to be a best-effort transport technology when used with the User Datagram Protocol (UDP). UDP which is data gram-based is the preferred protocol for use with voice transmission, due to its connection less characteristics.

The connection-oriented protocol, Transmission Control Protocol (TCP), has built in error checking and ensures reliable transmission of data, but since connection less transmission is preferred for voice traffic UDP is usually deployed. Voice packets are split in to two parts prior to bring sent over IP connections. The first part is the control signaling which runs on the TCP and the second is the actual voice packets runs on UDP. The following figure shows the VoIP packet. VoIP Packet Structure. You may be asking yourself why TCP is not used to transmit voice over IP.

When we look at the fact that TCP ensures reliable transmission of data or the data is retransmitted, we can see why this is not useful with live voice transmissions. If the original voice signal did not reach the end-user, there would be no use of trying to resend the signal because by the time the transmission had completed the signal would be useless. Retransmission is useful with the control signaling, so TCP is used for this purpose, but UDP handles the actual voice signal. We will now look at some of the advantages to switching to a VoIP network, compared to traditional voice systems. o IP provides contiguous connectivity that is separate from the transmission media it rides on. o The use of VoIP has a tremendous cost savings over high priced leased lines. o VoIP traffic is easily integrated with traffic from modern applications such as unified messaging or virtual call centers. o VoIP networks are less expensive to build and maintain over traditional circuit-switched networks. o You can get rid of the need for separate voice and data networks by combining the two. o Reduction in long distance charges by using VoIP compared to PBX technology.

So, what is involved in actually making VoIP work? The user typically uses a hardware device that connects on one end to a regular analog phone, and on the other end to their Local Area Network (LAN). The user's network is connected to the Internet by any means necessary. The VoIP hardware converts the analog signals into a digital signal, which is then transmitted to the Internet telephony service providers (ITSP) by way of the Internet. The call travels over the Internet using a special protocol, the most robust these days being the Session Initiation Protocol (SIP). The data being sent over the Internet is packet ized, meaning that it is split into small packets, which are then reassembled when they reach their destination (Packetizer, n. d.

). While this concept breaks-away completely from the traditional telephone system, the advantage of it is that not only is it secure, but if there is a routing problem on the Internet, the packets will automatically be rerouted efficiently. For instance, a router on the Internet between the user and the ITSP is down, the Internet by its very nature will reroute traffic to an adjacent router that is fully functional. This is a benefit that VoIP is able to exploit without the need to have this sort of technology built-in. For traditional systems, copper phones are not at all secure, as a line can easily be tapped in many places with minimal effort or knowledge. In addition, if the connection to the Central Office is broken at any point, the call will not go through until someone is sent physically to repair the problem.

Once the voice packets arrive at the ITSP, it can be routed accordingly, based on the destination. For instance, an ITSP can have multiple long distance providers, besides its connection to the local PSTN. If the user is calling Europe, the call is routed through the ITSP's long distance provider, and conversely, if a local call is dialed, it goes directly to the PSTN. The advantage of multiple long distance providers is that Itsp can choose which specific area codes are routed to which provider. In other words, the ITSP could have a better rate to Switzerland via AT&T, yet have a better rate to Japan via Global Crossings. The ITSP's switch automatically routes the calls through the least expensive provider for each area code, thus providing them with the highest margins, which often results in better savings for the user.

Once the call leaves the ITSP's switch, it is routed to the PSTN, or through a long distance provider. The PSTN is the local phone company's network, which eventually connects the user to their destination. Typically, the call recipient is connected to the PSTN through a traditional POTS line, such as a regular twisted pair lines. If the call is long distance, the long distance provider connects the ITSP to the destination country's PSTN. In the end, the recipient of the call usually cannot tell the difference between a call that came through the Internet versus a call over the traditional POTS / PSTN. Sometimes, the quality of a VoIP call exceeds that of a traditional phone call.

This is best experienced when two VoIP users call each other. In which case the call never reaches the PSTN, it goes from on ITSP to another over the Internet, making it a true VoIP call. The SIP protocol has helped bring VoIP mainstream thanks to its ability to traverse Network Address Translation (NAT). NAT is a technology that allows several computers to share a single IP Address on the Internet by assigning networked computers private IP addresses. The purpose behind sharing a single IP address is because IP Address space is scarce, and having a public IP for each computer, and device is prohibitively expensive for the residential through medium-sized businesses.

Right now, this market is in the process of rapid expansion. Last year saw the introduction of VoIP for mainstream residential and business use, which has led to the technology's explosion. There are several protocols currently in use to carry VoIP phone calls, which predate SIP. The main ones are H. 323 and Media Gateway Control Protocol (MGCP).

These protocols lack the features and most importantly flexibility that is offered by SIP. For instance, H. 323 can present configuration challenges, because there is a wide range of ports, about 10,000, that must but opened on firewalls to allow calls. On the other hand, SIP requires at minimum 2 ports, which can be selected by the user. MGCP is typically found in Voice over Asynchronous Transfer Mode (ATM) and Integrated Access Devices (IAD), which are typically used by medium-sized companies. I ADs are devices that act as the DSL modem or T 1 CSU / DSU, the router, and the VoIP device, all in one. They are typically very expensive, and are not very reliable.

VoIP will become more of a household name in the years to come as many new options are presented. Cable companies are entering the area of VoIP to offer its users the option to have Cable, Internet and VoIP all with the same service, on one convenient bill. In just the VoIP market some reports predict that by the end of 2005 somewhere between 2.8 million and 6.7 million residential VoIP lines will be in use and 27 million within 5 years (VoIP News, 2005, P 4). What will become of the current providers of traditional telephone services? Many have already begun to move into the world of VoIP and those that have not, will have some tough decisions to make in the years to come. It is hard to image life without traditional phone lines and service, but that could very well become a reality.

VoIP will continue to improve and expand with advancements in technology and this is just the beginning of what will be the future of the telephony industry. Currently VoIP is an unregulated service and is not required to follow the same guidelines as traditional phone service. But the Federal Communications Commission (FCC) is looking into this very issue. The main discussion is whether VoIP should be as highly regulated as traditional telecommunication services or left largely unregulated and classified as information services (Cisco Systems, 2005). The United States is not the only country looking at this issue.

In Europe there has been several reports address this same topic on regulating VoIP. Many of the topics revolve around 911 services, emergency assistance, and universal services to name a few. It is definitely a hot topic to keep a close eye on. Regulations will not only impact VoIP, but I believe it will have a ripple effect into other areas, such as Internet sales. Obviously there will eventually be more regulation on VoIP and the Internet.

With its rapid expansion and more and more consumers switching to IP based services and transaction, it just a matter of time before the state and federal governments want a piece of the pie. IT departments are typically at the forefront of technology. VoIP itself is not such a new technology for the telecommunications industry. Long distance providers, such as AT&T have deployed it for years. In fact, most long distance traffic today is transported via VoIP instead of physical connections. This allows carriers to expand their capacity without expanding their physical capacity.

Since VoIP by nature is compressed, more calls can be stuffed into a single satellite channel or fiber strand than a traditional analog signal, making VoIP less expensive to deploy and maintain, and can all be controlled in-house via a browser-based interface, thus eliminating the need for a separate PBX administrator. The learning curve for VoIP technology is very small, and the interfaces that have been designed are typically geared toward IT departments. Reference sCABS CG Interconnect Telecom Dictionary. (n. d. ). Retrieved June 27, 2005, from web Systems. (January, 2005).

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(2005, May 16). Learning Guide: SIP. Retrieved July 6, 2005 from web.