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Impact of VoIP on the Future of Telephony

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Any opinions, findings, conclusions or recommendations expressed in this material are those of the authors and do not necessarily reflect the views of UK Essays.

With the dawning of a new age of pervasive computing, there is a greater requirement for the exchange of data to be made possible between computing assets that are connected to a network. Interactions require an exchange of various multimedia formats as well as the provision of enhanced services including instant messaging and presence management.

There is, therefore, a need for a converged network that is capable of carrying both voice and multimedia in digitised form. Single network that is capable of carrying both voice and multimedia is preferable to having more than one networks because such a network is vastly more economical. Packet networks that use the internet protocol have emerged as a solution for this requirement.

These networks are capable of carrying all forms of data as well as voice over the internet protocol in real time. The networks use the internet protocol to provide a universal connectivity that was not previously possible. Despite the earlier problems involving latency, quality of service and reliability in the establishment of connections, VoIP or Voiceover the Internet Protocol has come to be accepted as a matured technology.

The proliferation of this technology is steadily increasing because of the economic considerations associated with its use as well as the futuristic services that are capable of being provided on I networks. It has been estimated that by the year 2015, VoIP will have captured about 50% of the global market share for telephony. VoIP has, therefore, proven to be a killer application for switched telephone networks and its advent has unleashed an unprecedented level of competition at all levels in the telecommunications industry. This dissertation takes a look at the impact of the VoIP technology on the future of telephony.

1.1 Introduction

Switched telephony networks have been responsible for carrying most of the world’s voice communications over the past decades, but with the advent of the relatively new communication technologies, there is likely to be a change towards a greater use of the telecommunications networks that carry voice as well as other information. The switched telephone networks and equipment were designed as fixed communications channels for bi-directional speech. In the old public switched network, a call that is initiated by a user establishes a connection between two users and once the connection has been established, no one else could use the connection.

Terminating the call frees the line for other users who can then initiate another call. With the evolution of computers, modems were used to modulate data streams over the voice telephony channels and over time, better modulation schemes were developed that resulted in higher data transmission rates. Developments in computing and multimedia have created a demand for new kinds of services and the telecommunications infrastructure that is in use is expected to satisfy this demand.

The development of internet and computer data networks along with the evolution of the Internet Protocol or the IP meant that it is now possible to send packets of data over the network. Voice can now be digitized after the speech signal is acquired from a microphone, encapsulated into packets and sent over the networks using the internet protocol. On the receiving side, these packets are de-encapsulated, processed and played over the speaker to present the information to the listener.

This method of transporting voice over the internet protocols called the voice over internet protocol or VoIP. It is also possible to send video and data from other shared applications to destinations using the internet protocol. A codec is used to encode and decode speech, audio and video over the IP network and there is no need to reserve a connection between parties to the call.

Signalling is, however, required to create and manage calls. Personal mobility, desire to communicate and availability can make the task of the required network signalling a complex one. There are several standards which have been developed for signalling over the new IP networks. The Session Initiation Protocol or the SIP which was developed by the Internet Engineering Task Force or the IETF manages the creation of a call as distinct from the ringers and switches in a switched network. For a more generalised exchange of data including video conferencing over the IP, the H.323 standard has been developed by the International Telecommunication Union, ITU for the management of network connections and the associated tasks of bandwidth allocation etc.

There has been growing acceptance of VoIP all over the world and a growing number of users including businesses, especially call centres, as well as network service providers have started to use this technology. A lower cost forth user is associated with the use of VoIP and this is the major factor in presenting a business case for the use of VoIP, along with the ability to send multimedia over a telecommunications link. IP makes more efficient use of the bandwidth that is available and inflated cross border tariffs are avoided.

Tariffs and regulations associated with VoIP telephony are, however, in a flux and it is difficult to predict how VoIP will be affected as a result of a possible implementation of new internet access charges. Adding a new media type on IP requires no change to the network infrastructure and the initiation of multiparty calls is only slightly different from a two-party call. IP also makes it possible to develop novel telecommunication devices and it is now possible for the world to progress beyond the simple voice telephone to the IP’s more exciting applications.

It is possible to use the public telephone network PSTN /IP Gateway Interoperability standard to feed IP encoded voice messages over the telephone network. This protocol coupled with the Resource Reservation Protocol, RSVP, makes it possible for an application to have a certain amount of bandwidth allocated with a maximum delay which assists in the implementation of a VoIP connection. Developments in new multimedia technologies has meant that there are two types of telecommunications networks which are in existence today, the old switched PSTN network with its reliability and quality along with the new packet based networks with cost efficiencies and an ability to provide the new types of services.

Although VoIP technology is developing and gaining a much wider acceptance, it is has not been without its problems. Because it is not possible to guarantee the arrival time of the data packets which have been sent over a packet network, there were problems with the voice quality when using VoIP. These problems could, however, be solved by using private networks and more internet bandwidth. Although VoIP does not use a large chunk of the internet bandwidth that is available, other applications that are running may result in a deterioration of the voice quality.

Hence, it was important to carefully consider how the internet connection was to be utilized and what bandwidth was required to be purchased. The security of VoIP communications was also considered to be a problem and it was thought that there was a need to compress voice and enhance security by using commercially available encryption products. The added latency or delay in voice communications was, however, considered to be unacceptable.

The best and the latest encryption devices are restricted items and their export is prohibited under United States Export regulations. There were, therefore, problems associated with implementing VoIP using either hardware or software and better quality of service or Qi’s was only possible with dedicated hardware. Although VoIP can hide costs associated with communications from the consumers, these costs could be returned in the form of service fees.

There was a need for call service capability to be brought to packet switching and the Qi’s had to be controlled to fall within acceptable limits. One of the important challenges of VoIP waste construct a converged VoIP and PSTN network that will permit VoIP and PSTN connectivity, with calls originating from one network and terminating into the other network. The SIP protocol which establishes the call in VoIP uses multiple messages with multiple parameters to initiate a call session and this protocol could fail because messages were not transmitted in the proper order with proper parameters and configuration.

A miss-configured user proxy address for the user can result in host unreachable messages being presented to the client. The Internet Control Message Protocol and the INVITE messages which are a part of the SIP protocol could be dropped when attempting to conduct a session using VoIP due to traffic, resulting in there being no connection to the remote system. SIP did not work well when tried from behind firewalls. Hence, with VoIP, call traffic becomes data traffic and this traffic is exposed to threats related to confidentiality, availability and integrity.

Hence, care needed to be taken when implementing VoIP in organisations, to provide for good design to prevent cost overruns, misalignment with strategic objectives and inadequate benefit realisation. IP networks must be able to meet strict performance criteria and perform for real time traffic. Packets travelling on a network will pass through a heterogeneous network with varying quality of service and bandwidth, but a reasonably good end-to-end quality of service is expected for voice communications. Signalling or the passing of messages for correct call setup, progress and termination is also important on the network. Hence, the implementation of VoIP was associated with the solution of important technical problems.

Despite the above problems that have been improved upon, VoIP today can match the features that were available in the legacy PBX systems and infect provide an enhanced set of features. The Internet today is an essential business tool and Internet connections are considered to be essential fixtures for any business premises. VoIP telephony systems have been designed to utilise the advantages of IP telephony in order to present a flexible communications infrastructure which businesses can use in order to simplify the business process and enhance productivity.

Many manufacturers of legacy telephony products have also accepted that IP telephony is the future and that the technology provides better communications equipment with enhanced features. VoIP has been showing a far greater level of proliferation in business organisations than ever before. Market reports have indicated that there is an increasing trend towards the full deployment of VoIP rather than its mere implementation.

Because there is an increased level of satisfaction and familiarity with VoIP technology, converged networks that blend VoIP and other technologies are considered to be more strategic in nature rather than the traditional voice and data networks. Security at the network infrastructure level is considered tube more important than voice security, with the level of satisfaction associated with the technology remaining about the same.

The new networks, which have new equipment that is in demand in the market includes IP PBXs or IP enabled traditional PBXs, Voice Enabled Routers,IP Phones, IP Centrex’s and Soft Phones etc. The new technology has changed the network components and the nature of the equipment that has been associated with telephony. IP PBXs indicated a 15% growth rate while IP Centrex indicated a 54% growth rate in usage from previous years according to market reports. A Centrex is essentially a scaled down PBX with features that are supported by the service provider.

Adoption of IP telephony presents advantages related to an enhanced and converged business process as well as advantages related to costs of adoption or changes. It is easier to deploy new integrated applications which may benefit the enterprise. Costs of calls within an organisation, between different sites are substantially reduced and enhanced features become available. Other advantages that result from the adoption of IP telephony include reduced staff costs, lowered costs associated with wiring, lower international call charges as well as reduced costs associated with the upgrading and maintenance of telephony equipment, including the PBX.

Because VoIP is a more complex and sophisticated technology as compared to the legacy telephony networks, instrumentation systems that are required for troubleshooting and managing VoIP have been cited as a barrier to its implementation. It has also been claimed that there is a shortage of trained people forth design and maintenance of VoIP networks. Because VoIP networks are so very different from the legacy telephone networks, substantial investments can be required to implement large projects, even though financial instruments are available to sustain a growth in the adoption of VoIP. Sophisticated upgrade of the legacy networks involving the purchase of new network equipment, servers, IP phones, management software and diagnostic tools may be involved to acquire a network with acceptable levels of latency, jitter and the number of lost packets.

VoIP Architecture

An obvious question that arises with regard to VoIP telephony is how it’s different from the legacy telephone networks? In the legacy telephony networks, voice communications had been handled by the proprietary PBX platforms providing circuit connection and circuit switched calling features such as call transfer and hold along with voice applications such as call accounting, voice mail and automated call distribution. The PBX ensured that savings were made by avoiding having to provide a line to each telephony user for connection to the organisation’s central office.

The PBX acted like a small central office with switching being made possible to users as required over a number of shared external telephone lines. The number of external telephone lines that were needed depended on the number of users that had to be connected to the PBX and the expected telephone traffic into the connection in elands. The PBX which could be considered to have the telephony switching intelligence was connected to the dumb telephone terminals or the telephones which merely passed digital keystrokes to the PBX for switching and voice application related decisions to be made. PBX systems in switched telephony can be networked together, but such efforts are likely to be expensive.

It was most likely that key telephone systems could not network with other key telephone systems and peripheral devices such as a Centrex could not interconnect with a PBX or another system. Hence, the legacy telephone systems were plagued with connectivity problems along with being expensive. The IP telephone system changed all this by adopting the router instead of the PBX as the distributor of traffic on the all data packet network. The routers connect not just one network together, but hundreds of thousands of networks, with the essential function of arouser being the diversion of packet data traffic to the appropriate devices on the network, with the correct IP addresses.

Hence, while thebe in the legacy system used to divert voice traffic to telephone numbers, the router diverts data packets of various kinds including voice, multimedia or video etc. to the data network equivalent of telephone number or an IP address. Interconnection problems are minimised because there is a standard IP protocol which is used to transport packets over the IP network and all IP protocol compatible devices may be interfaced with each other. The IP protocol is able to connect equipment manufactured by many different vendors over different types of media such as the twisted pair, coaxial or other data links such as the Ethernet or Token Ring and even the wireless connections.

The packets are transported in a reliable manner with the IP protocol running on devices ranging from PCs to mainframes. IP is everywhere and it carries packet traffic faithfully from anyone sending this traffic to anyone who is required to receive it. There is, therefore, a global standard that is understood anywhere in the world and unprecedented connectivity is made possible for all kinds of devices.

Amongst the other advantages of VoIP include provision of directory services over the telephone by which it is possible for ordinary telephones to be enhanced in order to act as internet access devices, availability offender office trunks for inter office communications, ability to access the office from a remote area such as the home and the ability to interact with the large number of customers who may want to make enquiries after having visited the corporate web site through IP based call centres. Fax over IP is also made available through the VoIP connection and it is possible to send fax data that has been converted into packets over long distances without having to deal with problems related to analogue signal quality and machine compatibility.

In the present scheme of things, the Integrated Services Digital Network or the ISDN represents the all-digital network that uses single wire to carry both voice and digital network services. ISDN tools an improvement on the old switched telecommunications network and this network too has been improved upon over the years to include new features. The ISDN uses the existing switched network with digital signalling and media transmission being used, which makes it possible for the subscriber to access a number of services through a single access point.

A number of different ISDN connections are available, but the most widely and commonly used connection is the basic rate interface or the BRI which consists of two 64 kbps media channels and single signalling or “delta” channel. Signalling channels are used to establish calls and perform call related signalling which permits theist network to be connected to networks with standard SS & signalling. ISDN is the subject of an International Telecommunications Union or ITU specification, the ITU-T recommendation which results in standardisation. However, this network is not as versatile as the packet switched network that has an all-digital approach with no analogue signalling whatsoever and which also has universal connectivity.

Switched – circuit networks rely on a fixed routing over the network to establish a connection. However, VoIP networks do not need to follow a fixed routing path and there is an adaptive routing algorithm that is employed to establish the best possible route under varying conditions of traffic. There is, therefore, a decentralized environment and the network is flexible enough to accept the deployment of new applications. Intelligence is important and this can be stored anywhere on the new IP networks.

VoIP does not provide a guaranteed quality of service or Qi’s when compared to the PSTN. However, PSTN uses expensive components and resources, whereas VoIP is able to provide connectivity at a reduced cost. It is the VoIP gateway which is responsible for connecting or interfacing the IP network to the rest of the telephony network.

Forth gateway, converting the media signal to the required format is only matter of transforming an input signal to an output signal. However, signalling and control translation requires conversion of semantics as well as syntax and there is a requirement for conveying the meaning of signals and control information from one network to the other. Hence, the evolution of VoIP telephony has made it necessary to provide an interface between various telecommunications networks and newer VoIP networks are connected to the older networks by means of interfacing equipment such as the gateways.

It can, therefore, be concluded that the emergence of IP telephony and VoIP have significantly changed telephony and it is very likely that the enhanced pace of VoIP adoption that has been witnessed in the business sector will continue to accelerate because of the convenience and cost savings that are offered by the relatively new technology.

It’s, therefore, worth investigating how VoIP technology will evolve and how this technology will change the future of telephony. The growth of VoIP has been phenomenal and Gartner estimates that the sale of consumer products for VoIP will grow by more than 40% in the United States in the year 2007. The advantages, disadvantages and the impact of VoIP on telephony are discussed below.

2.1 Products, Services and Issues Related to VoIP

In this section, it will be appropriate to discuss how VoIP technology has changed networks and network components and also how telephony services that are available have evolved as a result of the availability of VoIP technology. Products that use the VoIP technology are also discussed.

Network devices have evolved and changed as a result of the development of VoIP technology. The telephony switches, ringers and colour coded cables are likely to be replaced by the data network components. The heart of a VoIP phone system is the call processing server which is also known as the IP PBX into which all VoIP control connections are terminated. Call processing servers do not handle the actual VoIP payload, however, conferencing functionality, routing of voice traffic to another call processing server and music on hold features are provided by the call processing servers.

The VoIP payload traffic flows in a peer-to-peer fashion from one VoIP terminal to every VoIP terminal. VoIP control traffic, however, flows in a client –server model with VoIP terminals being the clients that communicate with the call processing servers. Call processing servers are usually software based but they may also be implemented as a dedicated appliance or be a part of a router platform and there may be a single server, a cluster of servers or a server farm. This server caters forth signalling mechanism that is required for a VoIP call establishment. Gateways are devices which act as the link between telephone signals and the IP endpoint.

The functions that are performed by gateways include the search function, connection function, digitizing function and the demodulation function. The gateway contains directory of the telephone numbers which have an associated Padres and a search is performed by the gateway to convert a dialled telephone number into an IP address upon a call being received to establish a connection. A connection is established between the calling party and a destination gateway through an exchange of information that is related to call setup, option negotiation, compatibility as well as a security handshake. The gatekeeper also digitizes any analogue signals that are received from the incoming trunk into a form that is useful for the gateway.

The incoming analogue signals are usually digitized into a 64 Kbps data stream which is pulse code modulated orca. The gateway is, therefore, required to be able to interface to a number of telephone signalling conventions so that the VoIP network can be interfaced to another network when required. Sophisticated gateways can accept both voice and fax signals and the fax signal is usually demodulated into a 2.4 – 14.4 Kbps digital format that is transmitted in the form of IP packets on the VoIP or IP network.

A remote gateway-modulates any fax related data into the fax format and this is relayed to the remote fax machine. Gateways on the IP network are connected to gatekeepers, which are LAN endpoints and these gatekeepers perform a discovery on being switched on to find out what IP addresses are connected to the LAN. This discovery information is then passed onto the gateway and the gatekeeper synchronises with the gateways to exchange data traffic if required. A collection of a gatekeeper and its registered endpoints are called a zone.

A gatekeeper performs the function of bandwidth management upon receiving a request for bandwidth allocation, translates alias addresses into transport addresses and performs the admission control function to the LAN, based on admission requests and confirms or rejects messages including ARQ / ARC and Arête. The gatekeeper, therefore, acts as a zone manager by performing variety of functions for its zone and the associated gateways as well as other devices in the zone. IP telephones have replaced the conventional telephony sets and the IP phones provide enhanced services suited to VoIP, while retaining the features that were available with the conventional instruments in order to keep the users who were used to the conventional phones comfortable.

Soft phones are software packages that may be installed on a PC and the user may use the Platform with an attached microphone for communications on the VoIP channel. The VoIP network may be classified as a logical switch that Isa packet network and it is different from the circuit– switched infrastructure of the legacy networks. Voice and data traffic have to be treated differently and if both types of traffic is to flow on the same network, then there has to be a capability for prioritisation. VoIP networks, unlike the circuit switched networks, can be considered in terms of statistical availability in which priority is given to packets of a specific application with a certain class of service or Qi’s. VoIP traffic is, therefore, given priority over other traffic flowing on the networks in order to ensure that the real time applications related to speech communications are met.

Regardless of what type of equipment is being used to receive VoIP packets, there can be a substantial packet loss over the network and this can degrade the quality of speech that is played out on the speaker. To improve the situation a “jitter buffer” is employed. This jitter buffer is a stack area in memory in which packets are stored prior to being played on the phone’s speaker. The jitter buffer adds to the overall delay that is involved in the VoIP speech transport but it’s necessary to allow for lost packets and to implement error correction schemes. Forward error correction schemes or FEC schemes are employed to check for corrupted packets.

In the intra-packet error correction scheme, additional bits of data are added to the packet in order to make it possible for the receiving end to determine if packet has become corrupted. Uncorrupted packets are played out while corrupted packets are rejected. Another scheme that is utilised to cater for packet loss is the extra packet FEC in which additional information is added to each of the packets which makes it possible forth receiving end to extrapolate voice if a packet is lost or becomes corrupted. Hence, unlike the analogue telephony equipment in which only filtering and amplification of the received analogue signals was performed, there is a substantial amount of digital signal processing using microprocessors that is conducted in the VoIP packet based equipment.

The error correction and detecting codes can be quite powerful, depending on the computing power that is available and hence the quality of the received voice can be improved. Delay is, however, introduced due to the digital processing of the packets and this can become an annoyance. For delays in excess of 600 Ms, voice communications is impossible while delays of 250 Ms disturb the communication considerably. Delays of 100 Ms do not show up as delays in the conversation and hence there is an upper limit that has to be observed when processing the packets on the VoIP networks.

High voice quality on the VoIP channel is bandwidth intensive and atoll telephone quality voice connection can require 64 Kbps data streamer call. However, it is not possible to conduct a call of this quality on the VoIP networks because of the bandwidth limitations. Speech compression is, therefore, used using different compression ended-compression codec’s in order to bring the required data rates to what can be sustained on the VoIP networks. Using codec techniques such as the G. 729 and silence suppression in which the areas of speech in which nothing is said are not converted into packets reduce the bandwidth substantially to about 5 – 6 Kbps for a voice conversation tube possible on the VoIP channel.

This is a remarkable achievement of digital signal processing considering that the overheads that are required by the routers on the network can run into about 7 Kbps. Silence suppression techniques can make the listener uncomfortable and to add to the natural flow of conversation, the ambient noise is periodically sampled and regenerated at the receiving end in between the pauses in the active speech so that the listener can feel more comfortable. All the digital signal processing, handshaking and coordination that is going on behind the scenes is transparent to the user of the VoIP channel and the user should be able to use the VoIP instrument naturally as a phone was used.

The management interface forth equipment that is in use is able to deal with telephony protocols, dialling plans, compression algorithms, access controls, PSTN fullback features, port interactions and management of the configuration for the instrument that is being used on the VoIP channel. Telephone numbers and IP address need to be handled transparently to the user and personal computers making voice calls will require telephone numbers to make the calls possible.

The packets that are sent over the VoIP network are encoded for the UDP/IP protocol instead of the TCP/I protocol so that retransmission of packets is not possible. TCP/IP is, however, a better choice for fax messages so that if packets are lost while attempting to transmit a page, the fax can be terminated. Retransmission of packets is hidden from the fax machine if TCP/I encoding is used for fax messages.

The widespread use of the TCP/IP protocol has resulted in a move towards what are known as converged networks. Convergence may be defined as one structure or one network architecture that will end up supporting all kinds of information media on all available network technologies. This means that it should be somehow possible to bring together all kinds of telecommunications technologies and interface them to each other in order to provide universal connectivity and inability to send and receive just about almost anything which may be required to be sent or received. Such universal connectivity has been made possible as a result of the widespread adoption of the IP protocol and this is the glue which binds all networks and applications.

Apart from VoIP, the other building blocks of convergence include unified messaging which attempts to integrate all forms of messages, computer and telephony integration which makes it possible to intelligently identify and route calls as well as automatically present information related to the caller, XML which provides a standardised format for data storage and interchange, Voice XML which makes it possible for an application to hear key tones that are encoded in DTMF.

SALT, which stands for Speech Application Language Tags make it possible for existing mark-up languages such as XML to access telephony related applications. SIP or the Session Initiation Protocol makes it possible to provide signalling for voice applications on IP as well as making it possible to initiate a voice call from an instant messaging application. Convergence promises to make it possible to interact with computers and other computing devices with intelligence and individuals can interact with others in ways that were never dreamt of before.

Mere telephony will cease to exist in the future and will be replaced with capabilities for multimodal integration involving speech, text, pictures and web interactions that can take place through instruments that will replace the simple telephone of the days gone by. It will be possible for organisations and call centres to interact at a much superior level, with those who interact with them and such interactions can involve quick access to information stored on computers, text, webs well as interactions while being mobile. Hence the capabilities of the simple telephony have been very much enhanced as a result of the evolving computing technologies and the internet protocol.

VoIP networks have a more complicated set of signalling protocols as compared to the switched networks. It is these signalling protocols which determine the features and the functionality that is available on the networks and how VoIP components will interact with each other. The International Telecommunications Union is the leading organisation which has played a role in the standardisation of VoIP protocols, although the Internet Engineering Task Force has also been a leading player in such efforts. Some equipment vendors have also come up with their own proprietary signalling schemes and hence the issue of signalling protocols has been a contentious issue on VoIP networks.

This issue is, however, being resolved as VoIP continues to become more readily accepted. Various protocols have their own set of strengths and weaknesses and some are more suited to a particular application as compared to the others. The most prevalent set of protocols include theH.323 which is a ITU recommendation related to packet based multimedia communication systems, which defines signalling functions as well as signalling formats related to packets for audio and video, the Real-time Transport Protocol or RTP includes RFC- 1889 and the RFC – 1890that provide end-to-end delivery services for packets that have real-time characteristics such as interactive audio and video.

The Real-Time Transport Control Protocol or the RTCP acts as a companion to the Rutland whilst this protocol is not needed for the RTP to function, thatch provides a feedback related to the quality of data distribution that is being accomplished by the RTP. Feedback from the RTCP can only indicate that problems are occurring on the network and not where they are occurring, but despite this limitation, the RTCP can be used as atoll to diagnose a problem. The MGCP or the Media Gateway Control Protocol is used to coordinate the action of media gateways. This protocol attempts to break the function of the traditional voice switches into elements related to the action of the media gateway, the media gateway controller and the signalling gateway functional units.

The Session Initiation Protocol or the SIP has been put forward by the Internet Engineering Task Force as a powerful client – server protocol that is used to manage multimedia sessions between speakers. Invitations are used by SIP to exchange lists of capabilities between the parties to the contact and control of the channel use. Another protocol that has emerged relatively recently is the Macao / H.248protocol which defines the media gateways that control the source of the calls and provide media conversion capabilities. This protocol implements a simple minimal design and functions very similarly to the MGCP, allowing for a wide range of telephone devices to be defined in order to support sophisticated business telephony features.

The H.323 is a standard that in effect brings together various sub-standards to be grouped into a single specification. The main H.323component standards include the G. 711 which is a codec standard forth conversion of voice frequencies into the pulse code modulated signal. The G. 723.1 standard is a codec standard for dual rate speech conversion for multimedia applications which makes coding possible at5.3 and 6.3 Kbps. The G.729 which is also a part of the H.323 is acidic standard for coding at 8 Kbps using the Conjugate-Structure-Algebraic-Code-Excited-Linear-Prediction method or the CS-ACELP. The H.255.0 is a substandard of the H.323 which deals with call signalling conversion into packets and multimedia communications.

The H.245 is a control protocol for multimedia communications, which supports supplementary services in the H.323 and it is a generic functional protocol that supports the H.323. The H.248is another protocol which works within the H.323 and it is the ITU equivalent of the IETF MEGACO. When a user picks up a phone and dials desired number, the H.245 protocol is used by the H.323 enabled phone to negotiate a channel and exchange the capabilities that are possible with the destination. After this, the H.225.0 negotiates call signalling and call set-up. This is followed by another component that is called RAS or the Registration / Admission / Status channel signalling the gatekeeper to coordinate the call within its zone.

Agate way is used to translate the VoIP packets into circuit-switched telephony signals for use over the PSTN network if the destination is connection on the PSTN. The H.323 standard has specifications that exceed the requirements of the VoIP telephony and it is in fact standard for video conferencing and multimedia transport. However,H.323 capabilities indicate that a network is versatile and capable of transmitting most information streams that may be required to be communicated between users.

It can, therefore, be seen that what goes on when a VoIP or multimedia connection is established between two users on a converged network is very different from what used to take place in the old switched networks. Hence, the use of IP has transformed the manner in which networks function and calls are established. Not only have the converged networks become much more sophisticated and complex, but the manner in which they operate is very different from the way in which the switched networks worked.

The standardised protocols that have been developed for VoIP and the cost as well as service benefits that are made available by using the technology mean that innovative telecommuting and multimedia conferencing as well as unified messaging are possible.

It is also possible to have location scheduling which makes communications to be uploaded anywhere on the network and simplified relocation, involving an easy change in the location of communications equipment is possible. High powered call centres which are capable of providing a more focused attention to customers are possible with less financial outlay. With such advantages, it has become obvious that the future of telephony has a different projection to what would have been possible with switched networks.

Having discussed the VoIP protocol and the major advances and differences in telecommunications network that have resulted from the advances in IP technology, it is now appropriate to discuss how these advances have had an effect on telecommunications around the world. This is done in the next section.

3.1 The Internet Protocol and the Global Telecommunications Transformation

In this section, it is appropriate to discussed how the internet protocol or IP has revolutionised telecommunications around the world.

As a result of the evolution of the internet technology and the widespread acceptance of the internet protocol or IP many new technologies, products and services were developed and many new marketing, distribution and technology companies came into existence. This is indicative of the interest that was able to be generated in the internet technology, which is a facilitator for communications, permitting many users to be interconnected together at the same time. The internet as a whole was seen to be a facilitator of commerce around the world because of its ability to provide a means for swift communications over vast distances and rapidly access information.

It was, therefore, thought that the internet could become a key element in the economic and telecommunications infrastructure and a precipitator of commerce. The internet protocol TCP / IP as well as other associated protocols were at the heart of the success of the internet. Software and the systems agreements that permitted diverse pieces of equipment to communicate and interact with each other assisted with the success of the internet.

The TCP / IP protocol may be regarded as a set of agreements that permit the exchange of communications between telecommunications equipment and networks. TCP operates above IP and provides the best effort transmission service with an end to end recovery that leads to sequencing. Flow control over the network required that both ends uniquely agree and it was possible for a packet to circulate, with every process being able to engage in multiple conversations. It is the IP that gets packets across the network and the TCP is responsible for bringing the packet stream into context.

The evolution of the IP presented three alternatives for the I telecommunications architecture. A clear channel approach provided dedicated circuit, an internet backbone approach that permitted an I transport mechanism and an IP “service bureau” approach that permitted other carriers to access the IP service backbone. A global IP networks able to offer a Qi’s that depends on the desired level of the Qi’s, the demand for the service and its actual use. These factors also determine the level of network congestion. The service provider becomes the IP backbone of the network and others who are interested must connect to this backbone at the IP level.

This approach is directed towards the service provider and not the majority carrier. The I channel could, therefore, be made available to the users by charging as a dial up clear channel usage, an IP utility, a dedicated IP or an ISP interface with IP telephony. Other variants were also possible with combination of the four previously mentioned schemes. A fully open I channel is likely to take a long time to implement and will require significant regulatory and political hurdles to be overcome because there is a need to have some sort of a definition for interfaces and standards that are acceptable.

A diversified IP community, open competition, globalisation and open markets with the lowering of costs on a global scale have stood in the way of rapid adoption of standardised technologies throughout the world. The installed telecommunications network have to be utilised in order to generate revenues and this has to be done by making use of raw bandwidth, leased bandwidth, TCP/IP carriage, voice unit carriage and providing services to the customers. Hence, the proliferation of technology is not merely determined by the availability of a superior technology, but also by the commercial agreements and monitory considerations related to investments that have already been made.

Making an international telephone call involves a number of service providers and players who may be providing services at various levels. The circuits that may be involved in between the international parties may include access tandems, dedicated IP networks, dedicated backbone networks, shared IP networks and some other old technologies. The networks that are used to affect calls are not selected merely by technology considerations but are also influenced by cost considerations, political considerations and other commercial considerations that exist. There may even be considerations related to network traffic or congestion.

A new technology gradually becomes widely accepted when superior performance, products and the requirements of the consumers coupled with persuasion and promotion result in others accepting and preferring this new technology. Acceptance by large players has to be translated to acceptance by the smaller customers who think carefully before re-investing in different equipment, new telephone numbers and schemes for generating incomes based on the benefits that new technologies have to offer.

Hence, better communications technology acceptance can result in displacement of business bases, creation of offshore distribution and sales, shifting of sales infrastructure to other locations / countries where customers may exist resulting in a loss of economic business bases and an economic dislocation of businesses as a result of changes in cost and distribution structures.

Along with purely commercial considerations, there are also regulatory considerations that have to be considered. These regulatory considerations are mostly influenced by national governments as well as international organisations that facilitate international communications. The drivers related to policy making are related to national interest, privacy and security of communications as well as the ability to generate revenues from taxation. The internet and I networks are relative vulnerable to threats of attack from adversaries and hence it is important to be able to protect a vital piece of economic infrastructure from attacks by adversaries.

The threats that may exist consist of active attack in which it is intended to inflict direct and measurable harm to the infrastructure, passive monitoring of communications traffic, active monitoring and use of the network as well as the covert use of a network for conducting transactions. I networks may be attacked by attempting to attack the transmission path, the endpoint consisting of system software or router, attack on switching and attempts at conducting silent or embedded attacks on the network in which destructive software may attempt to destroy software systems if a certain chain of events occurs.

Governments may demand that network providers work closely with them in order to protect national communications and attempt to enforce standards for protection. Attempts may also be made to take over private networks under attack or to conduct surveillance of such networks. Governments may also want a slice of the income that is being generated by network and tax communications. Hence, the VoIP networks which are anew technology are subject to government regulation and scrutiny and this is another hurdle that has existed in the rapid acceptance of amass communications technology. Because IP is an open network, therefore, it is relatively less secure and vulnerable to attacks and this has been a drawback in its greater proliferation as decision makers as well as policy makers attempt to better understand the issues.

Blueprint for VoIP Migration

Communications technologies have to prove themselves to be economically competitive in terms of International Long Distance or the IL economics and Competitive Local Exchange Carriers or CLEC economics that cater for local communications for residential and business customers. When providing CLEC type services, there has to be inability to provide international communications but the local charges are expected to be lower than the international charges.

When considering the economics of communication services provision, the Abased systems have shown that they are the cheapest as compared to T1lines, fibre optics and RSM based switched systems with concentration. The long distance international costs associated with the IP based systems are inconsequential as compared to other technologies. Hence, IP telephony has a capability to drive all costs down to a minimum base level, causing many telecommunications service providers to accept its usage.

AT&T, for example eliminated the use of circuit switches in its domestic network, preferring to rely on an IP backbone. International IP connectivity is provided by British Telecom Joint Ventures and Bell Atlantic amongst others and there are indications of greater usage and proliferation in Europe, the Americas as well as Asia. IP networks drive costs down and increase the usage of networks. Government concerns for privacy and the ability to identify those who are attempting to communicate as well as concerns related to taxation and a general regulation of VoIP channels have been an impediment to faster proliferation and adoption of this technology.

It can, therefore, be observed that IP networks and IP telephony has forced a convergence of networks with requirements related to being able to integrate any network with another network. There are requirements related to a complete integration of multimedia, voice, data and any other services. Openness of networks and markets has made it possible to have global marketing and distribution related to telecommunication and other commercial transactions.

The way in which transactions and tariffs were viewed has changed along with issues related to which country a transaction occurs, whose taxes are to be paid and whose law has to be obeyed. Global markets have emerged and new electronic marketing channels have emerged. The communications market has been greatly opened up for new entrants as a result of lowered barriers to entry and long term price reductions as well as the introduction of new services have been made possible. New products taking advantage of the new technologies are also likely to revolutionise telephony and communications all over the globe.

At the time of its introduction, internet telephony only offered lower rates. When using internet as a backbone, the voice quality was bad and the call set up time was not determinable, with a relatively low percentage of calls being completed successfully. These problems have, however, been resolved. Another approach that has evolved since then is one of attempting to integrate IP services while owning the market, very similar to Net2Phone which has tried to be selective but wants town the IP market.

From a business perspective, two extreme strategies have been used to connect IP traffic to a country. The first expansion strategy into a country is the land and expands strategy which involves connecting to a specific country through the use of internet as medium and then using a local ISP or a comparable player to terminate calls to customers in that country. The costs involved with the land and expand approach include the internet access charges and the charges associated with terminating calls. The other approaches through which telephony has been expanded into a country is the land and build while expanding approach.

When using this approach, IP network islanded into a country and then expanded in the form of a backbone network. This approach is more appropriate for a country in which there is no IP infrastructure in existence and the approach has the benefit of developing an all IP network that is capable of providing maximum benefits of the new technology. The owner of IP telephony backbone canals provide other services such as IP telecommunications, IP broadband internet access, ISP interconnectivity and Internet data services. Any value chain is, therefore, possible for those who have an Infrastructure.

Earnings are possible through the provision of raw bandwidth, supported IP telephony, IP based ILD, in country Disservice’s, high speed internet, value added internet data, ISP contention and facilitation as well as e-commerce content. Hence, they network can stimulate a lot of other economic activities that are conducive to the further growth of the internet related industries and this is by itself a stimulator for the enhanced proliferation of IP in country or region. Usually a common strategy has been to offer multiple services by establishing a “beachhead” in each country and then using this presence to attempt to sell other services.

Once an Pantry point has been established into a country, a variety of methods and technical configurations may be used to further expand into the country by buying and selling IP related services to and from the local operators. When making a decision to establish an IP presence into country, economic data related to telecommunications growth rates, telephone lines, population, GDP, economic climate and security of investment as well as other similar indicators related to the ability to generate a reasonable return on investment are considered. Hence, one global scale, the proliferation and acceptance of IP telephony is also determined by the overall economic growth potential of regions and countries as well as their future growth requirements. Most of the time, a national government is interested in permitting an expansion.

Telecommunications infrastructure which may be installed when expanding may include originating meet points where communications meets with other carriers which may be an international carriers and the originating local interconnection which consists of transmission or signal handling facilities that transfer signals to facilities for local telecommunications processing. Originating local switch and equipment which may consist of routers, VoIP and processing equipment as well as signal conditioning equipment may also be required to be made available for connection to domestic private lines, local ISPs and other telecommunications interconnections.

Various strategies may be used by commercial telecommunications operators in order to better position themselves in a country or region which is being expanded in order to supply services that are in demand and make a profit on the investment. Hence, apart from the mere availability of technology, the expansion of VoIP and I telephony is also dependant on opportunities, investment capital, and selection of the best opportunities for return on investment as well as the desire of telecommunications operators to take risks and expand into new areas of operations.

With a demand in existence for telecommunications as a vital tool for business and a requirement forth post-modern society of the ubiquitous age, opportunities certainly have abounded for expanding VoIP networks into new regions of the globe and the older technologies are being gradually left behind in favour of the newer technologies involving mobile wireless connectivity and multimedia interconnectivity. Services can be provided at cheaper rates as compared to what is possible with the existing telecommunications infrastructure and arbitrage opportunities become available.

Enhanced services are made available to what has been previously available resulting in consumers and customers switching over to the newer technologies along with completely new services for which there is a demand. Countries are certainly interested in attracting new operators because the economics of VoIP are strongly dependant on the availability of access and the bandwidth that is available.

Hence, it is in the interest of the public at large to have large number of telecommunications players getting involved and taking part in the reconstruction of the national telecommunications infrastructure. VoIP has changed the markets, commercial operations and regulatory remedies that have been associated with switched telephony. Things that had been closely associated with the existing switched telephone network such as addresses and network access have been decoupled and new component and convergent services have been introduced.

In the next section the ways in which VoIP has been implemented in the real world are discussed with a view to presenting the flexibility that is possible in implementing VoIP as a result of the availability of internet access.

4.1 Implementations of VoIP Telephony and Impact on Telecommunications

In this section, an attempt has been made to present the various ways in which VoIP telephony is being implemented in the real world. Access to an internet connection is the major pre-requisite for being able to make VoIP calls.

There are various ways in which VoIP is being used and implemented in the real world. There are about five different types of VoIP implementations including self-provided consumer VoIP, independent internet access, VoIP provided by broadband access service provider, corporate internal use of VoIP through the use of LAN / WAN and internal use through a carrier. Hence, VoIP has been made possible wherever there is an internet connection. Self-provided or do it yourself VoIP makes it possible for a PC user to place VoIP calls through a soft phone software application running on the PC.

Calls maybe made free of charge if the user has a flat-rate internet access plan. This service cannot be used for PSTN calls and has the attraction of zero cost for a call. In the independent internet access model of VoIP, the user can enter into an agreement with an IP telephony company that is distinct from the ISP that the user is connected to. Examples of such IP telephony companies include Net2Phone, Packet 8 and Vonage etc. End users of services are charged a retail fee for making callas and PSTN companies are paid termination and originating fees depending on the agreement that has been entered into for different types of calls for various geographic regions and distances that are involved.

Charges for the originating service are paid for by the ISP at rates that are lower than those for carrier selection because there are no originating charges. Free on-network calls can be offered without exposing the service provider to financial risk and the technology appeals to those with a broadband connection. Independent access can offer mainline replacement services, second-line services which may offer outgoing calls only or have a non-geographic number. Additional services that may be offered include call waiting, call barring and voice mail as well as voice conferencing.

Access to emergency services is possible and there may be a minimum monthly charge that is associated with the use of such services. VoIP services may also be provided by broadband services provider such as Yahoo BB in Japan. The broadband service provider uses a gateway to connect to the PSTN network and quality of service guarantees is offered.

The users can use soft phone or a an analogue terminal adaptor to place cheap phone calls over the PSTN without becoming a burden on PSTN revenues while the service provider can have an opportunity to add to their revenues. However, this way of offering VoIP has only been a success in Japan and some Asian countries such as Singapore and the broadband access providers elsewhere do not seem to find this concept attractive. There is a low minimum monthly service charge that is involved and the service aims to replace main line services although the main line still exists because it is used to provide DSL access and it is not possible to dial in the event of a power failure.

The corporate LAN or WAN can also be used to provide VoIP services forth internal use of an organisation. Corporate expenditure is reduced and these services may be enabled using an IP-enabled private branch exchange or PBX. No external service providers are involved in this model, although the corporate LAN / WAN operations and management maybe subcontracted. The numbers of corporations that are switching overdo this model are increasing gradually. A carrier may also use VoIP internally by replacing the circuit switches with software switches.

With this approach, network management is made possible through H.323,SIP servers, gatekeepers and the media gateway control protocol or the MGCP. VoIP telephony is also possible over Wi-Fi phones and this possibility creates a new class of telephone that exists between the services offered by fixed PSTN and the mobile carriers including GSM providers. Wi-Fi and 2G / 3G devices are beginning to be increasingly used but the area that is covered by Wi-Fi phones is small. Hence, these devices do not pose a significant threat to any other operators. However, mobile telephony operators compete against this type of threat by bundling minutes into their tariff structures.

VoIP can also be used over an IP data connection that is provided by a mobile phone operator, which may also provide native IP to provide voice services. Business models that can be sustained through the mobile phone include independent do-it-yourself consumer model, independent internet access, corporate use over LAN / WAN and carrier internal use. Mobile operators are especially concerned about routing calls over IP due to the reduction in their average revenue per user which can be detrimental to their debt position that was incurred after the construction of their mobile network.

There is also a desire on the part of the mobile operators to charge for the value of service that they are providing and not on the basis of a flat rate that they are sending. Thus, mobile users are at present cautious over the use of IP and their pricing of such abilities. VoIP solutions are, however, attractive on the mobile because business users prefer to have their fixed phone functionality being made available over their mobile phones.

Various business models that have been presented result in payment flows to providers of different services depending on how a call is routed and hence, VoIP is progressively finding a place in the overall scheme of telecommunications picture with the older networks and technologies being replaced by the more recent ones. VoIP can exist at a path on the overall route of a call.

It is now appropriate to discuss the impact of VoIP on the telecommunications market through the use of the various business models which have been described. The self-provided consumer type applications that depend on the sales of voice applications or the bundling of such applications with other software being sold by companies such as Microsoft, Apple and Skype etc. do not directly affect the revenues of other telecommunications operators.

However, this software based telephony approach will have a tendency to reduce the earnings of companies providing telecommunications services because the number of users who can use VoIP through internet access can run into millions. Independent internet access providers are relatively low barriers to entry services and have a tendency to put pressure on the prices being charged by the PSTN operators who have to compete. However, in response to this threat, many PSTN services have designed their tariff structure in such a manner that they remain competitive in the face of independent internet access providers providing a VoIP service.

Although PDA users may want to use the services as a cheaper form of mobile access, mobile operators compete against this threat by bundling minutes into their subscription with the result that the cost of incremental inbound traffic is reduced. VoIP services that are provided by broadband access service providers also puts pressure on the prices that are being charged by the PSTN services providers. However, this threat is limited to Japan and some South East Asian countries and the barriers to entry for this type of VoIP being provided are rather steep with high infrastructure costs being involved and the requirement for building access networks and the user must want VoIP, broadband access as well as virtual private networks or VPN.

Hence, those organisations or individuals who are interested in purchasing broadband and using VoIP services must want at least two of the offerings that come with broadband access providers. However, once again as a result of the competition and the services that are being made available by the broadband access providers, the PSTN services providers are incapable of raising their prices. In Japan, the success of broadband VoIP access is attributed to the fact that the broadband service providers there have extensive networks with unbundled fibre being made available by NTT.

There are about five million broadband subscribers who use VoIP services in Japan, but this segment is small in Europe and America. On the corporate front, it makes sense to replace a separate voice and data network with converged networks. It’s, however, not possible to replace the PBX and the desk telephones with IP phones overnight because such a proposition does not make financial sense. However, VoIP will start to have greater inroads into the corporate communications infrastructure when the equipment depreciation and replacement cycle requires the purchase of new equipment.

The new equipment that is VoIP compatible promises to bring with it new services such as presence aware routing, click dialling and unified messaging etc. Cost savings are, however, the biggest motivator for corporate changeover to VoIP services. The networks of most carriers are widely dispersed and can only be changed to


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