The Impact of VoIP on the Future of Telephony Essay Example

Enhancements in technology improved transmission techniques and boosted data speeds. The progress of computing and multimedia necessitates new services that can be accommodated by existing telecommunications infrastructure.

The development of cyberspace and computer information networks, along with the emergence of the Internet Protocol (IP), has enabled the transmission of data packets over the web. By capturing voice signals with a microphone, they can be converted into digital form. These digitized voice signals are then packaged and sent over the internet using IP. At the receiving end, these packages are unpacked, processed, and played through a speaker for the listener. This method is known as Voice over Internet Protocol (VoIP). Additionally, IP can also facilitate the transmission of video and data from various applications.

A codec is utilized for encoding and decoding address, sound, and picture over the IP web without requiring a connection between callers. However, signaling is necessary for establishing and managing calls. The complexity of performing the necessary signaling on the web is influenced by personal mobility, communication desire, and convenience. Various standards have been established for signaling on the new IP networks.

The Session Initiation Protocol or SIP, developed by the Internet Engineering Task Force (IETF), manages the creation of calls in a switched network, separate from tollers and switches. The International Telecommunication Union (ITU) has developed the H.323 standard for a more general exchange of information, including video conferencing over IP, which includes tasks such as managing network connections and bandwidth allocation. VoIP has been increasingly accepted worldwide, with more users including businesses, call centers, and web service providers adopting this technology. The use of VoIP offers lower cost for users and the

ability to transmit multimedia over a telecommunications connection, making it a favorable option.

IP enables more efficient utilization of available bandwidth and avoids excessive cross-boundary line duties. However, regulations and laws regarding VoIP telephony are constantly changing, making it difficult to predict how VoIP will be affected by potential implementation of new internet access charges. The integration of a new media type on IP does not require any modifications to the underlying web infrastructure, and initiating multiparty calls is only slightly different from initiating two-party calls. Additionally, IP allows for the development of new communication devices, expanding beyond traditional voice telephony to the more exciting applications offered by IP technology.

The PSTN/IP Gateway Interoperability criterion can be used to transmit IP encoded voice messages over the telephone web. By combining this protocol with RSVP, an application can allocate and hold a specific amount of bandwidth for a VoIP connection. The advancements in multimedia technologies have resulted in two types of telecommunications networks: the traditional switched PSTN network known for its reliability and quality, and the newer packet-based networks known for their cost efficiencies and ability to offer new services. However, despite the growing acceptance of VoIP technology, it has faced challenges due to the inability to ensure timely delivery of data packets over a packet network, leading to issues with voice quality.

These issues could be resolved by utilizing private networks and increased internet bandwidth. While VoIP does not consume a large amount of available internet bandwidth, other running applications can affect the quality of voice. Therefore, careful consideration of internet connection utilization and required bandwidth was important. The security of VoIP communications was also a concern, prompting

the need to compress voice and enhance security through commercially available encryption products.

The unacceptable nature of potential voice communication delays led to challenges in implementing VoIP using hardware or software due to restrictions on the export of advanced encoding devices under US laws. As a result, achieving high-quality service (QoS) was only possible with specialized hardware. While VoIP can conceal communication costs from consumers, these expenses may be transferred as service fees. Therefore, it was necessary to integrate call service capability into packet switching and maintain QoS within acceptable boundaries.

One of the major challenges in the field of VoIP was to establish a unified network that allows for both VoIP and PSTN connectivity. This network enables calls to originate from one system and terminate in the other. The SIP protocol, used in VoIP, utilizes multiple messages with various parameters to initiate a call session. However, this protocol was prone to failure when messages were not transmitted in the correct order or with the proper parameters and configuration. In cases where user configuration was incorrect, messages signaling that the host was unreachable would be presented to the client. Additionally, attempting to conduct a session using VoIP could lead to dropped Internet Control Message Protocol and INVITE messages, resulting in a lack of connection to the remote system. It was also observed that SIP did not function effectively behind firewalls.

Henceforth, when VoIP is implemented in organizations, it is crucial to be cautious about confidentiality, accessibility, and integrity issues as call traffic is transformed into information traffic. A well-designed approach is essential to prevent excessive costs, misalignment with strategic objectives, and inadequate benefits. IP networks must adhere to

rigorous performance standards and operate in real-time for actual traffic. As packets traverse a diverse network with varying quality of service and bandwidth, it is expected that voice communications maintain a relatively good end-to-end quality of service. Additionally, proper signaling for call setup, progress, and termination is significant on the network.

Despite the significant technical challenges, implementing VoIP has now resulted in modern day systems that offer the same features as traditional PBX systems and provide additional enhancements. In today's business environment, having internet connections is crucial, making VoIP phone systems essential for any business establishment. These systems take advantage of IP telephony to create a flexible communication infrastructure that businesses can use to streamline their operations and improve productivity.

Many manufacturers of legacy telephone products have also recognized that IP telephone is the future and that the technology provides improved communications equipment with enhanced features. VoIP has been demonstrating a significantly higher level of adoption in business organizations than ever before. Market studies have indicated that there is a growing trend towards complete implementation of VoIP rather than just its implementation. Due to the increased level of satisfaction and familiarity with VoIP technology, converged networks that combine VoIP and other technologies are considered to be more strategic in nature instead of the traditional voice and data networks.

Security at the web infrastructure level is considered to be more important than voice security, while the level of satisfaction associated with the technology remains the same. The new networks, which require new equipment that is in demand in the market, include IP PBXs or IP enabled traditional PBXs, Voice Enabled Routers, IP Phones, IP Centrex’s and Soft Phones etc.

This new technology has changed the network components and the nature of the equipment that has been associated with telephony. According to market studies, there has been a 15% growth rate in the use of IP PBXs and a 54% growth rate in the use of IP Centrex from previous years. Centrex is essentially a scaled down PBX with features that are supported by the service provider. The adoption of IP telephony offers advantages related to an enhanced and converged business process as well as advantages related to the costs of adoption or modifications.

The adoption of IP telephony offers several advantages for businesses. Integrated applications can be deployed more easily, leading to potential profits. Costs of calls within an organization and between different sites are reduced, and improved features become available. Accepting IP telephony also results in decreased staff costs, lower wiring expenses, reduced international call charges, and lower costs for upgrading and maintaining telephone equipment, including the PBX. However, there are some barriers to implementing VoIP, such as the complexity of the technology and the lack of trained personnel for designing and managing VoIP networks. Implementing large projects may require significant investments, despite the availability of financial instruments to support the growth of VoIP adoption.

The purchase of new web equipment, waiters, IP phones, direction package, and diagnostic tools may be necessary to create a web with acceptable levels of latency, jitter, and lost packages.

VoIP Architecture

VoIP telephone systems differ from legacy telephone systems in several ways. Legacy systems used proprietary PBX platforms to handle voice communications, providing circuit connections and circuit switched calling features. These systems also included voice applications such as call accounting,

voice mail, and automated call distribution. The PBX allowed for cost savings by not requiring a dedicated line for each telephone user to connect to the organization's central office. Instead, a small central office (the PBX) facilitated connections as needed over a shared pool of external telephone lines. The number of external lines required depended on the number of users who needed to be connected to the PBX and the expected telephone traffic volume in Erlangs.

The PBX, which held the intelligence for telephone exchanging, was connected to telephones that transmitted digital key strokes to the PBX for voice applications and other determinations. Networking PBX systems in switched telephones was costly, and cardinal telephone systems couldn't network with each other or with peripheral devices such as Centrex. This resulted in connectivity issues and high expenses for legacy telephone systems. However, the IP telephone system solved these problems by using the router instead of the PBX to distribute traffic on the data network.

Routers not only connect multiple networks, but thousands of networks. Their main function is to redirect package data traffic to the appropriate devices on the network, using the correct IP addresses. Formerly, in the legacy system, PBXs were used to redirect voice traffic to phone numbers. However, routers now redirect data packages of various types, including voice, multimedia, or video, to the data network equivalent of a phone number or IP address. Interconnection issues are minimized because there is a standardized IP protocol that is used to transport packages over the IP network. Any IP protocol compatible devices can be connected to each other. The IP protocol can link equipment from different manufacturers and using

different types of media, such as twisted pair, coaxial, Ethernet, Token Ring, and even wireless connections. Packages are transported reliably with the IP protocol running on devices ranging from PCs to mainframes.

IP is ubiquitous and reliable in carrying package traffic from senders to receivers. It provides a global standard for connectivity, enabling various devices to connect. One advantage of VoIP is its ability to offer directory services over the telephone, allowing regular phones to function as internet access devices. Inter-office communication is made convenient through this technology, as well as accessing the office remotely. Additionally, IP-based call centers facilitate communication with numerous clients who have visited the corporate website. Fax over IP allows sending fax data as packages over long distances, eliminating issues with analog signal quality and machine compatibility. The Integrated Services Digital Network (ISDN) is an all-digital network that uses a single wire for voice and digital services. It is an improvement on previous telecommunications networks and has been enhanced over time with new features.

The ISDN uses the bing switched web with digital signalling and media transmittal being used, which makes it possible for the endorser to entree a figure of services through a individual entree point. A figure of different ISDN connexions are available, but the most widely and normally used connexion is the basic rate interface or the BRI which consists of two 64 kbps media channels and a individual signalling or “delta” channel. Signing channels are used to set up calls and execute call related signalling which permits the ISDN web to be connected to webs with standard SS ; signalling. ISDN is the topic of an International Telecommunications

Union or ITU specification, the ITU-T recommendation which consequences in standardization. However, this web is non every bit versatile as the package switched web that has an all digital attack with no parallel signalling whatsoever and which besides has cosmopolitan connectivity.

Switched circuit networks require a fixed routing path on the web to establish a connection. In contrast, VoIP networks do not need to follow a fixed route and instead use an adaptive routing algorithm to find the optimal path based on changing traffic conditions. This creates a decentralized environment where new applications can be easily added to the flexible network. Intelligence can be stored anywhere on these new IP networks. However, it is important to note that VoIP does not guarantee the same quality of service (QoS) as the traditional public switched telephone network (PSTN).

However, PSTN utilizes expensive components and resources, while VoIP is capable of providing connectivity at a reduced cost. The VoIP gateway is responsible for connecting or interfacing the IP network with the rest of the telephone network. For the gateway, converting the media signal to the required format is simply a matter of transforming an input signal to an output signal. However, signaling and control translation involves not only transitioning semantics but also sentence structure. There is a need to convey the meaning of signals and command information from one network to another.

The development of VoIP telephone has made it necessary to connect older telecommunications networks with newer VoIP networks through interfacing equipment such as gateways. This shift has significantly changed the telephone industry, with the increasing adoption of VoIP in the business sector expected to continue due to its convenience

and cost savings. Therefore, it is important to investigate how VoIP technology will evolve and impact the future of telephone systems. The growth of VoIP has been remarkable, with Gartner estimating a more than 40% increase in the sale of consumer products for VoIP in the United States in 2007.

The text discusses the advantages, disadvantages, and impact of VoIP on telephone. In this section, we will explore how VoIP technology has transformed networks and web components, as well as the evolution of telephony services due to the availability of VoIP technology. Products utilizing VoIP technology are also examined. The development of VoIP technology has brought about changes in network devices, potentially replacing telephone switches, tollers, and color coded cables with information network components. The core of a VoIP phone system is the call processing server, also known as the IP PBX, where all VoIP control connections terminate.

Name processing waiters handle conferencing functionality, routing of voice traffic, and music on hold features for VoIP calls. The VoIP warhead traffic travels between VoIP termini in a peer-to-peer manner. On the other hand, VoIP control traffic follows a client-server model, with VoIP termini acting as clients communicating with the call processing waiters. Name processing waiters can be software-based or implemented as dedicated devices or part of a router platform. There can be single servers, server clusters, or server farms for this purpose. These servers facilitate the signaling mechanism for establishing VoIP calls. Gateways serve as intermediaries between telephone signals and IP endpoints.

The gateways perform several maps including the hunt map, connexion map, digitising map, and demodulation map. The gateway has a directory of telephone numbers with associated IP

references. When a call is received, the gateway performs a hunt to convert the dialled telephone number into an IP reference to establish a connexion. The gateway exchanges information with the naming party and a destination gateway to establish a connexion, including name apparatus, option dialogue, compatibility, and security handshaking. The gatekeeper also digitizes incoming linear signals into a useful form for the gateway.

The incoming parallel signals are typically transformed into a 64 Kbps information stream that uses pulse codification modulation (PCM). To connect the VoIP network to another network, the gateway needs to be able to interface with various telephone signaling protocols. Some advanced gateways can handle both voice and fax signals. The fax signal is usually converted into a digital format with a speed of 2.4 - 14.4 Kbps and transmitted as IP packets on the VoIP or IP network. A remote gateway reconverts any fax-related data back into fax format and sends it to the remote fax machine.

Gateways on the IP web are connected to gatekeepers, which are LAN end points. These gatekeepers perform a search on being switched on to find out what IP addresses are connected to the LAN. This discovery information is then passed on to the gateway. The gatekeeper synchronizes with the gateways to exchange data traffic if necessary. A combination of a gatekeeper and its registered end points is referred to as a zone. The gatekeeper carries out the tasks of bandwidth management, translating alias addresses into transmission addresses, and enforcing access control based on access requests. It also approves or rejects messages such as ARQ/ARC and ARJ. Therefore, the gatekeeper acts as a zone director by

performing various functions for its zone and the associated gateways, as well as other devices in that zone. IP telephones have replaced traditional telephone sets. These IP phones offer improved services suitable for VoIP, while still providing the features that were available with traditional instruments to ensure comfort for users accustomed to conventional phones.

Soft phones are software packages that can be installed on a Personal computer, allowing users to communicate on the VoIP channel using their computer platform and an affiliated microphone. The VoIP network is a logical switch and differs from traditional circuit-switched networks. The handling of voice and data traffic is different, and prioritization is necessary if both types of traffic are to flow on the same network. Unlike circuit-switched networks, VoIP networks can be viewed in terms of statistical availability, where packages of a specific application with a certain quality of service (QoS) receive priority. To ensure that real-time speech communication applications are met, VoIP traffic is given priority over other traffic on the network. Regardless of the equipment used to receive VoIP packages, there can be significant package loss over the network, resulting in a degradation of speech quality.

The use of a "jitter buffer" helps improve the current situation. This buffer, stored in memory, holds packages before they are played on the phone's speaker. The jitter buffer increases the overall delay in delivering the VoIP address, but it is necessary to account for lost packages and employ error correction strategies. Forward error correction (FEC) strategies are used to check for corrupted packages. In the intra-packet error correction strategy, additional data bits are added to the package, allowing the receiving terminal to

determine if a package has been corrupted.

Uncorrupted packages are rejected, but corrupted packages are played out. Another strategy used to address package loss is the excess package FEC. This strategy involves adding extra information to each package, allowing the receiving terminal to interpret voice data even if a package is lost or corrupted. In contrast to standard parallel telephone equipment that only filters and processes parallel signals, VoIP package-based equipment relies on extensive digital signal processing using microprocessors. The level of error correction and monitoring codes can be highly effective, depending on the available processing power, ultimately leading to improved voice quality.

The digital processing of packages introduces delay, which can be irritating. Communication becomes impossible with holds exceeding 600 MSs, while holds of 250 MSs disrupt communication. Delays of 100 MSs do not appear as holds in the conversation, but there is an upper limit that must be observed when processing packages on VoIP networks. High voice quality on the VoIP channel requires a bandwidth of 64 Kbps per call for toll telephone quality voice connections. However, bandwidth restrictions prevent conducting calls of this quality on VoIP networks.

The utilization of various compaction and de-compression codecs enables the conveyance of necessary information rates within sustainable levels on VoIP networks. Techniques like G.729 and silence suppression significantly reduce bandwidth by converting countries of address with no speech to compressed packages. This allows voice conversations on VoIP channels with a bandwidth requirement of approximately 5-6 Kbps. Digital signal processing plays a vital role in achieving this accomplishment, as it helps to minimize operating expenses for web routers to around 7 Kbps.

Silence suppression techniques can make the

listener uncomfortable and disrupt the natural flow of conversation. To address this, ambient noise is periodically sampled and recreated at the receiving end during pauses in the active conversation, providing a more comfortable experience for the listener. Despite all the digital signal processing, handshaking, and coordination happening behind the scenes, the user of the VoIP channel should be able to use it just like a regular phone. The management interface for the equipment being used covers various aspects such as telephone protocols, dialing programs, compression algorithms, access controls, PSTN disconnection features, port interactions, and configuration of the VoIP instrument. The handling of telephone numbers and IP addresses should be seamless to users, as personal computers making voice calls will require telephone numbers to initiate calls.

The VoIP web uses encoded packages for the UDP/IP protocol instead of the TCP/IP protocol to prevent retransmission of packages. However, TCP/IP is considered a better option for facsimile messages because it allows termination of the facsimile if packages are lost while transmitting a page. When TCP/IP encryption is used for facsimile messages, the facsimile machine is unaware of any retransmission of packages. The widespread adoption of the TCP/IP protocol has led to the development of converged webs.

Convergence refers to the integration of various telecommunications technologies and web architectures in order to provide universal connectivity and the ability to send and receive all types of media. This is made possible by the acceptance of the IP protocol, which serves as the foundation for all networks and applications. Convergence includes elements such as VoIP, unified messaging, computer and telephone integration, XML, Voice XML, and SALT. These components enable seamless communication and access

to telephone-related applications using standardized formats and languages.SIP, also known as the Session Initiation Protocol, enables signaling for both voice applications on IP and the ability to initiate a voice call from an instant messaging application.

Convergence enables the possibility of interacting with computers and other devices intelligently, allowing individuals to connect with others in unprecedented ways. Traditional telephones will no longer be present in the future and will be replaced.