VoIP Essay

Table of Contents


Introduction………………………...………... Page 3


Problem Statement………………………….... Page 4


Analysis…………………………………...…. Page 8


Recommended Solution and Implementation.. Page 11


Conclusion………………………………………………………….…Page 13


Works Cited………………………………………………………… Page  14











The traditional landline telephone was designed and constructed by Alexander Graham Bell March 10, 1876. No doubt, Graham’s invention has contributed to business, government, and social affairs due to ability to send and receive real-time voice over several hundreds and perhaps thousands of miles. Much as changed since invention of the telephone such as transfer of control of landline carriers from the Interstate Commerce Commission (ICC) to the Federal Communications Commission (F.C.C.)  Briefly  after  telephone service begun to penetrate few highly populated urban areas across North America, much  controversy sparked regarding capitalistic ownership and control of local-long distance carriers. The constant bickering between consumers business, governmental agencies, state public service commissions, over rates and monopolization eventually prompted revisions to be made to the original Federal Communications Act of 1934. Therefore, the Federal Communications Act of 1996 opened doors for smaller landline entities; both landline and cellular-wireless. The increase of available local telephone providers in urban and rural areas resulted in dramatic reduction of telephone rates due to completion between former and new local-long distance telecommunications providers. After premier of Second Generation (2-G) wireless telecommunications; Voice over Internet Protocol (VoIP) was introduced during 1995. This converging technology initially permitted individuals with Personal Computers (PC’s) to place calls over their computer to other individuals with a computer connected to broadband internet. The sound quality was far superior to that of 8Kilohertz traditional landline telephones, however, then (VoIP) was much less desirable for replacement of traditional landline telephone service. “Voice over-Internet Protocol (VoIP) is also recognized by many professionals in the telecommunications industry as Internet Protocol (IP) Telephony. This technology has ability to support real-time transmission of full-duplex (synchronous) digital voice audio over Internet Protocol (IP) over the public internet or private data network.” (Edwards, 2001) For example, VoIP converts the voice signal from older analog telephones into digital signals by analog-to-digital and digital to analog conversion utilizing an Analog Telephone Adapter (ATA). One of the most significant advantages of (VoIP) technology when compared to Public Switched Telephone Networks (P.S.T.N.) is ability to make long distance phone calls, thereby, avoiding expensive long distance telephone rates typically metered by geographical distance and time from origination. Too, utilizing VoIP over IP networks enables several calls to be made utilizing the same or less bandwidth, otherwise, required with legacy networks.






Problem Statement


Most importantly, VoIP technology has converged with Wireless Local Area Networking (WLAN) and WiMax. The advantages of VoIP include; improved sound quality, lower long distance rates; including flexibility to utilize call-forwarding, Voice Mail, Three-Way calling, adjustable bandwidth control to optimize sound quality, virtual phone numbers for a substantially lower rate when compared to traditional landline telephone service. Too, VoIP subscribers can take his or her own VoIP equipment and utilize it any where in the world where there is a broadband internet connection. Furthermore, there are a number of VoIP providers offering out-of-local area exchanges, therefore, permitting VoIP subscribers to utilize one or more local exchanges in nearly everywhere within the United States. For example, a VoIP subscriber residing in Columbus, Ohio can choose a Los Angeles, California number to enable their friends in that city to place calls to Columbus using a L.A. local exchange.


Voice Internet Protocol Technology (VoIP) was originally intended as a software program and architecture “developed by Vocaltech during 1995” (Theodorou, 2006), therefore, enabling digital, crisp, clear, voice telephone calls to be made and received over computers and networks  using a variety of protocols including network technologies. Then, two primary protocols were responsible for creation of an overlay network; Session Internet Protocol (SIP) and H.323 protocols. The (SIP) protocol SIP is an RFC standard (RFC 3261) from the Internet Engineering Task Force (IETF), a professional standards organization responsible for administering, and furthermore, developing mechanisms comprising the Internet. Meanwhile, (SIP) is currently progressing; including being extended as technology matures. Therefore, (SIP) products are advertised/ socialized in the marketplace. (Unspecified, 2010) Conclusively, the number of businesses and consumers utilizing VoIP technology is continuing on a steady incline. In contrast, as more telecommunications subscribers switch to (VoIP), consequently, (IP) infrastructure is lacking sufficient bandwidth and network throughput requirement to full-fill (VoIP) technician requirements and/or specifications.


The disadvantages of VoIP include; inability to make and receive phone calls during internet and/or commercial power outages. Too, any technical abnormalities occurring on the internet can result in either poor VoIP performance; or in some situations complete outages. In addition, if VoIP is utilized with 802.11a, 802.11b, 802.11g Wireless Local Area Networks (WLAN’s) can sometimes inhibit co-channel and adjacent interference caused by router-Access Points (AP’s) in the same general vicinity; including devices such as telephones, toys and other low power communications devices operating in the 2.4 Gigahertz portion of the (UHF) radio spectrum. Also, the hidden node problem can cause co-channel interference if several AP’s are used for one large Wireless Local Area Network (WLAN). In essence, un-intentional interference can cause degradation of WLAN performance, and likewise, the VoIP equipment will perform poorly or be unusable. Therefore, as telecommunications technology expands and converges, likewise, will be requirement of (VoIP) to adhere to performance expectations. Security is a major issue associated with Wireless Local Area Networking (WLAN), therefore, data and voice are transmitted between Access Points (AP’s)/Routers and Wireless Network Interface Cards (NIC’s)  installed in Personal Computer; especially, laptop and notebook (PC’s). Likewise, it is important to remember 802.11 Wireless Ethernet Protocol devices are categorized as part-15 non-licensed devices by the Federal Communications Commission. Most importantly, the Federal Communication Commission’s part-15 rules and regulations has been intact for several years initially utilized as a yardstick for low-powered handheld citizens band transceivers operating in the 26-27 megahertz of the radio spectrum. The initial rule of thumb specified these low powered transceivers must not exceed 100 milli-watts (1/10) watt of output power fed into a vertical radiator in excess of 10’. Likewise, low power radio transmitters as such must not cause interference to licensed radio services, however, being subject to accept interference by licensed radio services. Today, in certain situations depending on; modulation, emission type, bandwidth deployed and frequency utilized; part 15 transmitters all permitted to transmit with more that 1/10 watt (100 mill-watts power. However, the similar rules apply; the transmitter must accept interference from licensed and non-licensed stations, however, must not cause interference to licensed radio stations. Regarding 802.11 security issues; the person who utilizes wireless internet must assume that his or her internet is subject by to be intercepted by non-licensed individuals, furthermore, utilizing administrative controls to lock the (WLAN). In addition, security of any wireless device is jeopardized when data and/or voice is transmitted over RF carriers. However, utilization of different types of digital modulation greatly enhances security.  To enhance wireless network security the following modulation schemes are deployed; Orthogonal Frequency Division Multiplexing (OFDM), Frequency-Hopping Spread Spectrum (FHSS), and Direct Sequence Spread Spectrum (DSSS). Orthogonal Frequency Division Multiplexing (OFDM). “Orthogonal frequency-division multiplexing (OFDM) sometimes referred to as discrete multi-tone modulation (DMT), is a complex modulation technique for transmission based upon the idea of frequency-division multiplexing (FDM) where each frequency channel is modulated with a simpler modulation. In OFDM the frequencies and modulation of FDM are arranged to be orthogonal with each other which almost eliminates the interference between channels. Although the principles and some of the benefits have been known for 40 years, it is made popular today by the lower cost and availability of digital signal processing components. Therefore, the theory behind (OFDM) is utilization of low-rate modulations. Consequently, low-rate modulations use relatively long symbols compared to the channel time, therefore, characteristics are less sensitive to multipath. Additionally, it is advantageous to send a number of low rate streams in parallel than sending one high rate waveform. Therefore, (OFDM) divides the frequency spectrum in sub-bands small enough so that the channel effects are constant (flat) over a given sub-band. Then a "classical" IQ modulation (BPSK, QPSK, M-QAM, etc) is sent over the sub-band. If designed correctly, all the fast changing effects of the channel (multipath) disappear as they are now occurring during the transmission of a single symbol and are thus treated as flat fading at the received.  Classical signal processing such as channel coding, power allocation, adaptive modulation and coding can be applied for a given sub-band or over the sub-bands. Multiuser allocation is also possible, either using time, coding or frequency separation of the users. Orthogonal Frequency-Division Multiplexing (OFDM) is almost always used in conjunction with channel coding, therefore, an error correction technique utilized to create coded orthogonal (FDM) and/ or (COFDM). Most importantly, it is a complex technology to implement, however, now widely used in digital telecommunications systems to facilitate ease of encoding and decoding such signals. Too, the system has been used in radio broadcasting as well as certain types of computer networking technology. This is particularly due to the fact that such signals show good resistance to multipath fading, best known as the source of "ghosting" on analog television broadcasts. An Orthogonal frequency-division multiplexing (OFDM)  carrier signal is the sum of a number of orthogonal sub-carriers, with baseband data on each sub-carrier being independently modulated commonly using some type of quadrature amplitude modulation (QAM) or phase-shift keying (PSK). This composite baseband signal is typically used to modulate a main RF carrier. The benefits of using OFDM are many; including high spectrum efficiency, resistance against multipath interference (particularly in wireless communications), and ease of filtering out noise (if a particular range of frequencies suffers from interference, the carriers within that range can be disabled or made to run slower). Also, the upstream and downstream speeds can be varied by allocating either more or fewer carriers for each purpose. Some forms of Rate-adaptive DSL use this feature in real time, so that bandwidth is allocated to whichever stream needs it most.” (Unspecified, 2007) Most importantly, Orthogonal Frequency Division Multiplexing (OFDM) is utilized for/with 802.11a, 802.11g, and 802.11n Wireless Local Area Networking (WLAN).


Voice-over-Internet Protocol (VoIP) is expanding and converging with other technologies as time progresses. Most importantly, (VoIP) Quality of Service (Q.O.S) must meet or exceed that of traditional landline carriers; including cellular-wireless providers in order to be cost effective in highly competitive telecommunications markets. “Psytechnics Ltd., provider of voice, video and multimedia quality testing tools has conducted considerable research pertaining to (VoIP) Quality of Service (Q.O.S.), therefore, finding an increase of subscribers as time progresses. In contrast, studies conducted by the same research tend to indicate inconsistent or poor (Q.O.S) amongst those who utilize (VoIP) in businesses and residences. Specifically, the research firm found that nearly 60% of the respondents were concerned about quality levels of (VoIP) technology. Too, the study indicated most respondents placed blame on unreliable service providers and multi-vendor deployments, therefore, lacking abilities of managing-vendor networks as the primary cause. This research firm utilized the Quality of Experience (Q.o.E) Methodology to describe/evaluate call quality. Meanwhile, the company’s (Q.o.E.) tools are said to be capable of monitoring network applications to quickly identify and diagnose issues that could potentially affect quality, therefore, permitting brief resolutions to maximize both IP network infrastructure.” (Hickey, 2007)


Therefore, innovation through development of design and troubleshooting tools will most likely enable professionals in research, development; including engineering to make dramatic improvements to (VoIP). For example, it does not require a rocket scientist to realize that utilization of (VoIP) over existing Internet Protocols such as IEEE 802.3 Ethernet has increased exponentially over the past decade. Most importantly, there are a greater number of businesses and consumers utilizing (VoIP) over IP infrastructure, therefore, in essence accounting for just a small amount of allocated bandwidth by Internet Service Providers (ISP’s). Perhaps, more concentration should be initially placed on maintaining a prescribed Quality of Service (Q.o.S.) amongst Internet Protocol (IP) infrastructure including networking prior to establishing an expected (Q.o.S.) for all protocols in both infrastructure and P2P (VoIP) overlays. “In an Infrastructure overlay, the overlay nodes primarily perform roles of both server and router. Essentially, the clients are the same nodes external to the overlay. Whereas, P2P overlays differ from infrastructure variants in that the clients also form and function as part of the network. Most importantly, each node can essentially perform any of the communication roles. This enables the overlay to become infrastructure free from perspective of the service provider since end-to-end hosts themselves perform roles of servers, whereas, other nodes functioning as routers.” (Ganguly, 2008) Fundamentally, it is extremely important to emphasize that both Internet Protocol (IP) and Voice-over-Internet (VoIP) are independent, however, function together to enable P2P and/or Infrastructure overlay. For example, there are three types of nodes within the VoIP architecture. Firstly, are the End nodes, thus, responsible for termination and initiation of (VoIP) calls, consequently, communicating with one-another by the Session Initiation Protocol (SIP). End nodes are fundamentally involved in VoIP overlay infrastructure, additionally, utilized to monitor overlay path quality; including ability to access overlay nodes. The second type of node associated with (VoIP) infrastructure is Overlay nodes within the VoIP architecture. These nodes are responsible for monitoring path quality over adjacent overlay nodes; including serving role as routers within overlay network. Lastly, Control nodes are the third type of node within the VoIP infrastructure. Most importantly, Control nodes are necessary for provision of auxiliary data required to initiate (VoIP) calls.


Overall, performance of (VoIP) can vary due to performance of the (IP) infrastructure, and furthermore, technical metrics within either P2P or Infrastructure overlay (VoIP) architecture. For example, (VoIP) Quality of Service (Q.o.S) is contingent on routing of packets between nodes in an overlay network. Too, in an overlay network, the nodes are connected by overlay links that are logical paths enabling establishment of physical links within the infrastructure. Therefore, any variation or obstruction of packet transfer between nodes can seriously result in a reduction of (Q.o.S.).


There are many variations of (IP) network conditions that affect overall (Q.o.S) of Voice-over-Internet Protocol (VoIP) networks. For example, available bandwidth, latency-delay, jitter, packet loss, and talk overlap are all determinates of (Q.o.S) over wired network infrastructure, with emphasis placed exclusively on the (IP)  generic/internet infrastructure, however, not withstanding/considering variations that must and do occur in both P2P and Infrastructure overlays. Again, reiterating importance of realization that Internet Protocol (IP) and P2P/Infrastructure are independent, however, most function together in order to achieve expectations of Voice-over-Internet (VoIP (Q.o.S).


Packet loss is an important factor to consider when attempting accomplish an expected level of (VoIP) performance. “Packet loss occurs due to lost packets along the data path in the (IP) internet infrastructure.” (Ganguly, 2008) Packet loss can severely degrade voice quality such as clipping audio such as an improperly adjusted audio Automatic Gain Control (AGC) and limiter. “Packet loss is said to occur when a router’s queue/buffer limit is exceeded during times of excessive network throughput.” (Ganguly, 2008) Especially, packet loss is an important aspect to consider when utilizing (VoIP) over 802.11a, 802.11b, 802.11g, and 802.11n Wireless Local Area Networking (WLAN) or wireless (VoLAN) as some telecommunications professions would prefer to recognize. Likewise, packet loss can occur due to co and adjacent channel interference; including inadequate signal strength when utilizing (VoIP) over Wireless Local Networks (WLAN’s). “Packet loss too can occur due to excessive jitter buffering; including if end-to-end delay for the (VoIP) packet expectations is not met.” (Ganguly, 2008)


Too, inadequate bandwidth over either or both guided and non-guided network infrastructure can result in dramatic deterioration/degeneration of (VoIP) performance. Most importantly, there are several reasons for insufficient internet bandwidth over Wired and wireless networks. For example, excessive lengths of coax wireless cable between the residence terminal block can introduce high attenuation, consequently, resulting in low up-stream and down-stream signal levels. Too, damaged coax and/or connectors can cause insufficient bandwidth. In addition, excessive or damaged couplers and connectors can cause less that acceptable internet bandwidth.


 Likewise, packet transmission delay in the (IP) architecture plays an important role in network latency, equally important, affecting (VoIP) Quality of Service (Q.o.S). Specifically, delay is time required for a packet to travel between nodes in a network. Most importantly, there are three types of delay in (IP) network infrastructure; packetization, propagation, transport, and jitter buffer delay. “Propagation delay is proportion to speed of light, and furthermore, is dependent on physical distance between two nodes. Worthy of discussion too is transport delay time. Transport delay occurs due to network devices such as routers and firewalls utilizing methods to route and filter packets. Whereas, packetization delay is function of codec speeds utilized in conjunction with protocols such as G.723. On the other hand, jitter buffer delay is caused by jitter buffering utilized to reduce/minimize variations of arrival times of voice datagrams.” (Ganguly, 2008) Most importantly, excessive delay can cause abnormal buffering, therefore, reducing quality of performance; including resulting in unintelligible voice quality and in certain situations lost/dropped calls. 


Conclusively, there are many factors to consider while attempting optimization of (VoIP) performance when utilizing IP over (WLAN) and (LAN) networks


Equally important is latency of a data transmission on networks. “Latency is the total time (delay) a packet faces while in transient from the source to destination. Most importantly, there are multiple contributors to latency. The foremost of these contributors is the physical limit imposed by speed of light in free space. Meanwhile, the second contributor to latency is queuing delay. This form/type of delay occurs when a due to packets of data being cued by a router. Too, transmission delay is a form of latency, thus, caused primarily by bandwidth limitations. Importantly, delay is said to be an additive quality, therefore, all types of delay-latency accounts for a segment of overall transmission delay.” (Ganguly, 2008) Here’s my analogy of how this theory could be applied when attempting to evaluate (VoIP) performance; as mentioned, latency is most often caused by; sorting- queuing of packets in routers, line attenuation, and free space loss; including packet loss. Therefore, an expression that could be used to determine external influence towards latency could be; Latency Proportion= (Router Delay+ Attenuation Delay+ Free Space Loss). 




Since development of Voice –over-the- Internet Protocol (VoIP) technology; there has been considerable deliberation pertaining to Quality-of-Service (Q.o.S) of this newer technology. Most importantly, “(VoIP) Quality of Service is an important consideration for businesses transitioning from landline to Internet Protocol (IP) telephony. The reason for this is because many businesses for decades have taken telephone (Q.o.S). Enterprises that switch to a low-cost (VoIP) service are often surprised to discover having entered a world where service quality can be elusive including difficult to achieve. Since few enterprises wish to subject their employees and clients to poor phone call quality (Q.o.S), it is critical to pay close attention to service quality issues when designing a (VoIP) system. Despite widespread perception to the contrary, (VoIP) actually has potential to deliver higher (Q.o.S) than older Public Switched Telephone Networks (P.S.T.N.’s).” (Edwards J. , 2007) As mentioned, abnormal network bandwidth including excessive throughput can result in poor (VoIP) performance. Therefore, “(Q.o.S.) doesn’t stop at the point the (VoIP) service enters and leaves residential and/or business premises. For example, a business enterprise’s internet network also plays a major role towards overall (Q.o.S.).” (Edwards J. , 2007) Equally important, switchers, routers, hubs, and other devices play an important role in  determining internal (Q.o.S.) amongst Local Area Networks (LAN’s), Metropolitan Area Networks (MAN’s) and Wide Area Networks (WAN’s). Therefore, it is essential for hubs, switches, routers, and other devices supporting (Q.o.S) standards, such as those defined in RFCs published by the Internet Engineering Task Force (IETF), including other professional organizations to meet and/ or adhere to industry standards.


 Currently, “there are multiple types of deployment scenario for (VoIP) service.  Most importantly, these scenarios can be categorized as the following types; P2P (VoIP) over Internet, Managed (VoIP) over Internet, (VoIP) over managed carrier Internet Protocol (IP) networks, including (VoIP) over Enterprise Networks. Specifically, P2P (VoIP) enables users with a computer, modem, sound card, and microphone to place and receive high-quality digital telephone calls, exclusively, over Personal Computers (PC’s). Whereas, Managed VoIP over the Internet would/could involve techniques as mentioned, in addition to, “methods of locating users on-line a user-id; including; how to support NAT transversal, and furthermore, utilization of Distributed Hash Tables (DHT’s). Voice over the Internet Protocol (VoIP) over Managed carrier IP networks would involve (VoIP) between Public Switched Telephone Networks (P.S.T.N.’s), including direct (VoIP) to users.” (Ganguly, 2008) For example, (VoIP) telecommunications is an excellent solution for the consumer at home due to requirement of connecting an Analog Telephone Adapter (ATA) between a traditional landline telephone and either a modem or router connected to a ISP/DSL or satellite modem. “Voice over Internet Protocol (VoIP) is becoming widely popular amongst many business enterprises. Typically, business enterprises enables (VoIP) to reside over existing (IP) infrastructure, furthermore, utilized with existing Private Branch Exchange (PBX) networks.



 “An SIP signaling protocol is utilized for establishment of sessions over the internet most often utilizing the IEEE Ethernet 802.3 Protocol. Most importantly, the flexibility and capabilities of SIP enable it to be an attractive choice for integration of various types of telecommunications/communications application into existing data networks. Additionally, SIP is a simple protocol responsible for functioning as a mechanism to establish sessions. The SIP operation is independent of the underlying network transport protocol, and furthermore, indifferent to the content/type of session being established such as SMS, Instant Messaging (IM), streaming audio and video (video-conferencing).” (Ganguly, 2008)


Today, “Internet Service Providers (ISP’s) are deploying packet telephony technologies as an alternative to traditional circuit-switched telephone networks in order to shift traditional voice services to packet- based networks and to create new services that combine data, voice, and video information. The lower cost associated with converged data and voice networks are a prime driver for deployment of packet telephony. Too, both SIP and H.323 protocol is said to encompass a distinct set of advantages and disadvantages within a packet voice network. However, it is possible to use both protocols within the same network, and it is definitely necessary to interconnect networks using one or the other. The H.323 protocol has been available for several years, and furthermore, telecommunications industry has made significant investments to construct and outsource several huge H.323-based networks.  Furthermore, Session Initiation Protocol (SIP) is being used modestly to its utilization, therefore with similarities to other Internet technologies such as HTTP. However, H.323 is more widely deployed and the standard is more mature.” (Unspecified, H.323 and SIP Integration, 1991-2002)  For example, the SIP protocol can support small payloads such as text messages, however, without architecture capable of supporting high bandwidth media applications required for high-speed data, voice, and video. In essence, (SIP’s) ability to be used as a signal protocol is due to its computer language plain-text utilization to send and receive words such as text messages over an overlay. Most importantly, “flexibility and capabilities of (SIP) enable this technology to be an excellent choice for integrating various types of communications-telecommunications applications into existing networks. Too (SIP) is a layered protocol meaning the protocol serves solely as control mechanisms for more complex protocols with ability to support substantially higher payloads of data, audio, and video.” (Ganguly & Bhatnagar, 2008)



 H.323 can provide essential protocols other than requirements necessary for session establishment. For example, H.323 can support both Real-time Transport Protocol (RTP), and Real-time Transport Control Protocols (RTCP).




Most importantly, “Real-time Transport Protocol (RTP is mostly utilized with Internet Protocol networks, and furthermore, this protocol was/is designed to provide end-to-end network transport functions over applications transmitting real-time data such as audio and video, via multicast or unicast network services.” (Unspecified, 2010) Therefore, we have a robust protocol that functions in an architectural type suite capable of supporting numerous network monitoring statics that are second to none over traditional network logging , thus, fundamental towards network Quality of Service (Q.O.S.) including security. In essence, data collected by this method might be used with a packet analyzer and other instruments to optimize network throughput.


On the other hand, Real-time Transport Control Protocol (RTCP) is considered a sister protocol of the Real-time Transport Protocol (RTP), therefore, functioning in accordance with (RTP) in delivery and packaging of multimedia data, however, (RTCP) is not responsible for transmitting data like its (RTP) counterpart. Most importantly, the (RTCP) protocol is primarily responsible for providing positive feedback on the (Q.O.S.) of service provided by the (RTP) protocol.” (Unspecified, RTCP, 2010)


Therefore, here we have a complimenting technology that perhaps enables (RTP) to be more useful for collecting real-time network statistics to improve network quality of service; including enhancement of security amongst the network infrastructure.


Likewise, (RTP) and (RTCP) utilized together can be utilized to collect data such data transfer rate, lost /recovered packets, jitter, latency; including real-time monitoring of network status condition.


Meanwhile, (VoIP) over computer internet infrastructure (broadband) phone service continues to impact the way we work live due to flexibility and cost of this newer technology with ability to provide worldwide telecommunications at a much lower cost than traditional landlines.


Recommended Solution and Implementation


Voice –over-Internet Protocol (VoIP) performance (Q.o.S) can initially be improved by assuring there is sufficient internet bandwidth. Although, most (VoIP) service providers provide user friendly access to bandwidth controls, consequently, there are many factors within the internet architecture that directly affects performance of P2P and/or Infrastructure overlays which are independent of the typical (IP) infrastructure or  IEEE 802.3/802.11 Ethernet-Wireless Ethernet (OSI) layers. If poor (VoIP) performance is observed the user should utilize a personal computer (PC), and furthermore, access on-line servers that measure internet bandwidth and latency of packets over networks. If there is an acceptable level of bandwidth then the user should check his or her down-stream, up-stream levels by accessing the modem parameters entering their IP address utilizing Hyper-Text-Transfer Protocol (HTTP) IP addressing. For example, the down-stream, up-stream levels and signal-to-noise s/n ratio reaching the modem from the (ISP) can be accessed by entering; in the author of this essay’s browser. Acceptable levels are; -4.8 to -10 dBmV downstream power level (DPL) reading is a snapshot taken at the time this page was requested, whereas, up-stream power levels should be between 47.5 to 56 dBmV. Likewise, the ratio of line signal to noise s/r level should measure at near 30 dB at the modem. Most importantly, the understanding of line attenuation and noise is directly proportion according to distance between the modem to the gateway, and furthermore, from the gateway to the head-end (node) should be around 30dB. Any readings below approximately 20 dB would indicate that a high-level of noise is entering the line. Line noise can be reduced by reducing the amount of coax between the last-mile to the residence, and furthermore, from outside the residence to the modem’s (F-Connector). Equally important, is assurance that coaxial cable utilized is 75 ohms, in addition to, replacing all RG 58u cable between the outside terminal block with new RG 6u cable. Industrial grade cable utilized by telecommunications providers today is highest quality RG-6 with a good shield. We should realize that the purpose of utilizing shielded coax or wire is to prevent electronic or magnetic fields from entering or leaving the center conductor. Too, good grounding at the residence terminal block can potentially reduce static-line noise, in addition to, protecting connected equipment such as modems, routers, hubs, switches, and computers from static discharge. A method of grounding the coax is to utilize a grounding coupler; or perhaps connecting a splitter enclosure as close as possible to the residence. The splitter or coax ground female-to-female F-type connector should electrically connected to a  6’ copper ground rod driven into soil with a piece of solid conductor (not-stranded)  # 10-12 AWG electrical type solid copper wire. I cannot over emphasis requirement to keep coaxial runs as short as possible between the modem and outside terminal block, and likewise, keeping coax runs short between the terminal block and nearest utility pole. Furthermore, modern (ISP) last mile connections between the line at the residence to the nearest fiber optic node should be minimized to avoid signal loss and including minimizing noise. Unlike digital repeaters that amplify signals by tearing down and reconstructing signals; line amplifiers along a last mile run amplify both signal and noise. Therefore, subjected noise amplified between sections of coax located prior to the nearest fiber optic node. We should realize that utilization of minimal splices including using fewest line couplers will theoretically reduce both line loss and noise.


Too, if traditional telephones are connected to ISP Ethernet via a Wired or Wireless router, it is recommended to connect the (VoIP) Analog Telephone Adapter (ATA) box to a port on the router as opposed to connecting it to a hub port if a hub is utilized in either 10base-T or 100 base-T configuration connected to a router, thus, to enable more than two computers and VoIP to be utilized.


“Running on top of heterogeneous (IP) networks while dealing with distinct characteristics of each of them, and furthermore, going through transcoding at the gateway of the network boundaries results in significant performance issues (VoIP) deployments must face. Most importantly, (VoIP) is susceptible to underlying network conditions such as; delay, jitter, and packet loss, therefore, degrading the (Q.o.S) to the point of being unacceptable to the average user and un-acceptable to the business enterprise. Maintaining (VoIP) quality can be easier if networks are designed to better handle (VoIP) overlays by provisioning strategies, and furthermore, utilization of voice codecs designed to provide robustness to variation in network conditions by meet higher expectations- demands of (VoIP) consumers in both business and residences. Furthermore, the quality of (VoIP) can be significantly improved by utilization of proper codec. As opposed to traditional network codec(s), current wideband codec(s) provide natural and crisp sound quality. New mechanisms are also being used by current (VoIP) codec(s) that have ability to tolerate packet loss and jitter. Most importantly, multi-rate codec(s) operating at multiple bit rate(s) are available today that will easily adapt to network congestion or variation in available bandwidth. Meanwhile, different networks with heterogeneous and unpredictable behavior are challenging today as Research and Development (R&D) teams continue to protocol, design-develop, and test (VoIP) codec(s) in an attempt to reach and maintain a (Q.o.S) level of performance comparable or better than Public Switched Telephone Networks (P.S.T.N.)./ TDM technology. (Ganguly, 2008)



As previously mentioned, there are additional complexities when attempting to overlay either P2P or Infrastructure over all types of wireless internet. For example, if RF signal strength is abnormally low between an Access Point(s) and or client(s), certainly there will be considerable packet loss, thus, resulting in considerable reduction of bandwidth. Low signal strength can be caused by free space loss (excessive distance- between Transmitters and Receivers) utilized by both Access Points (AP’s) and Wireless Network Interface (NIC) cards, obstructions to line-of-sight propagation associated with Ultra High Frequency Signals 2.4 and 5 Gigahertz Bands in the  (UHF) radio spectrum, low current/ amperage if (NIC) cards utilize consume power from batteries. Another important factor to consider is utilization of 802.11n as opposed to previous (WLAN) standards. Most importantly, 802.11n (WLAN) utilizes the 2.4 and 5 Gigahertz portions of the radio spectrum, therefore, being less susceptible to adjacent and co-channel interference. Too, 802.11n (WLAN) deploys Orthogonal Frequency Division Multiplexing (O.F.D.M.), therefore, enabling Multiple-Input-Output technology (M.I.M.O) to be utilized.

“Orthogonal Frequency Division Multiplexing (O.F.D.M is a digital transmission technique that uses a large number of carriers spaced apart at slightly different frequencies. First promoted in the early 1990s for wireless LANs, OFDM is used in many wireless applications including Wi-Fi, WiMAX, LTE, ultra-wideband (UMB), as well as digital radio and TV broadcasting in Europe and Japan. This distinct modulation is also used in land-based Asymmetric Digital Subscriber Line (ADSL), however, Frequency Division Multiplexing (FDM) implies multiple data streams, orthogonal FDM carries only one data stream broken up into multiple signals. Essentially, hundreds or thousands of carriers known as subcarriers are used for a single data channel. Too, the multiple subcarriers enable (NIC) receivers to more easily detect the signals in environments with multipath and other interference. In addition, each subcarrier can transmit a lower-speed signal; all of which are aggregated at the receiving side into the original high-speed signal. Lower speed signals are also more easily deciphered at the receiving end.” (Unspecified, OFDM, 2010) As mentioned, Multiple-Input-Output technology (M.I.M.O) is the method of utilizing multiple antennas for wireless communications. For wireless networking, (M.I.M.O.) technology appears in some WiFi routers, greatly enhancing their capability over single-antenna routers.


Multiple-Input-Output technology (M.I.M.O) 802.11n WiFi routers utilize the same network protocols and signal ranges that non-(M.I.M.O.) routers do. Specifically, (M.I.M.O.) technology achieves higher performance by aggressively transmitting and receiving data over 802.11n (WLAN) channels. (M.I.M.O) signaling technology can increase network bandwidth, range and reliability at the potential cost of interfering with other wireless equipment. The exact number of antennas utilized in a (M.I.M.O) WiFi router can vary. Typical MIMO routers contain three or four antennas instead of the single antenna that is standard in all earlier forms of consumer WiFi routers. Most importantly, (M.I.M.O) is a key element of the 802.11n WiFi networking standard.


In addition, signal multipath can cause jitter, slower bandwidth, and delay. “Signal multipath can occur when a radio frequency (RF) signal is transmitted towards the receiver; therefore, general behavior of the RF signal tends to grow wider as it is transmitted further. On its way, the RF signal encounters objects that reflect, refract, diffract or interfere with the signal. When an RF signal is reflected off an object, multiple wave-fronts are created. As a result of these new duplicate wave-fronts, there are multiple wave-fronts that reach the receiver. Multipath propagation occurs when RF signals take different paths from a source to a destination. A part of the signal goes to the destination while another part bounces off an obstruction, then goes on to the destination. As a result, part of the signal encounters delay and travels a longer path to the destination. Most importantly, multipath is defined as the combination of the original signal plus the duplicate wave fronts that result from reflection of the waves off obstacles between the transmitter and the receiver. Multipath distortion is a form of RF interference that occurs when a radio signal has more than one path between the receiver and the transmitter. This occurs in cells with metallic or other RF-reflective surfaces, such as furniture, walls, or glass.” (Unspecified, Multipath and Diversity, 2008)




This term paper has discussed Voice-over-Internet Protocol with emphasis placed on performance (Q.o.S). It is important for a newer or converging technology as such to be constantly improved through (R&D), therefore, to enable it to be competitive with similar existing technologies. Most importantly, the author of this term paper has worked diligently for several hours locating sources that adhere to academic standards.


Conclusively, it is my best intention to be graded for this accomplishment; however, this essay would be excellent for anyone wishing to learn about Voice-over-Internet Protocol (VoIP) technology.



Works Cited

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Theodorou, V. (2006, February 11). The History of Voice over Internet Protocol. Retrieved November 08, 2010, from Ezine Articles: http://ezinearticles.com/?The-History-of-Voice-over-Internet-Protocol&id=143336


Unspecified. (1991-2002). H.323 and SIP Integration. Retrieved December 06, 2010, from Cisco: http://www.cisco.com/en/US/tech/tk652/tk701/technologies_white_paper09186a0080092947.shtml


Unspecified. (2008, January 21). Multipath and Diversity. Retrieved December 11, 2010, from Cisco: http://www.cisco.com/en/US/tech/tk722/tk809/technologies_tech_note09186a008019f646.shtml#multipath


Unspecified. (2007). OFDM. Retrieved December 10, 2010, from Imran: http://www.mobileisgood.com/ofdm.php


Unspecified. (2010). OFDM. Retrieved December 09, 2010, from TechEncyclopedia: http://www.answers.com/topic/ofdm-technology


Unspecified. (2010). RTCP. Retrieved November 03, 2010, from Cisco: https://supportforums.cisco.com/docs/DOC-1116


Unspecified. (2010). RTP. Retrieved November 03, 2010, from Cisco: https://supportforums.cisco.com/docs/DOC-1119


Unspecified. (2010). What Is SIP Introduction. Retrieved November 09, 2010, from SIP Center : http://www.sipcenter.com/sip.nsf/html/What+Is+SIP+Introduction