The Impact of VoIP on the Future of Telephony

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. A single network that is capable of carrying both voice and multimedia is preferable to having more then 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 Voice over 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 IP 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 telephony 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 protocol is 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, a 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 a 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 for the 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 a 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 QoS 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 QoS had to be controlled to fall within acceptable limits. One of the important challenges of VoIP was to 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 in fact 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 then ever before. Market reports have indicated that there is an increasing trend towards the full deployment of VoIP rather then 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 then the traditional voice and data networks. Security at the network infrastructure level is considered to be more important then 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 for the 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 is 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 erlangs. 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 a router being the diversion of packet data traffic to the appropriate devices on the network, with the correct IP addresses. Hence, while the PBX 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 a 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 of inter 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 a single wire to carry both voice and digital network services. ISDN too is 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 a single signalling or “delta” channel. Signalling channels are used to establish calls and perform call related signalling which permits the ISDN 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 QoS 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. For the gateway, converting the media signal to the required format is only a 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 is, 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 then 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 for the 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 a directory of the telephone numbers which have an associated IP address 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 or PCM. 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 re-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 on to 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 ARJ etc. The gatekeeper, therefore, acts as a zone manager by performing a 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 PC platform with an attached microphone for communications on the VoIP channel. The VoIP network may be classified as a logical switch that is a 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 QoS. 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 is 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 a 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 for the 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 a toll telephone quality voice connection can require 64 Kbps data stream per 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 and de-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 to be 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 for the equipment that is in use is able to deal with telephony protocols, dialling plans, compression algorithms, access controls, PSTN fallback 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/IP 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/IP 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 an ability 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, web as 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 the H.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 – 1890 that 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 RTP and whilst this protocol is not needed for the RTP to function, the RTCP 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 a tool 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 Megaco / H.248 protocol 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.323 component standards include the G. 711 which is a codec standard for the 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 at 5.3 and 6.3 Kbps. The G.729 which is also a part of the H.323 is a codec 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.248 is 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 a 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. A gateway is used to translate the VoIP packets into circuit-switched telephony signals for use over the PSTN network if the destination is a connection on the PSTN. The H.323 standard has specifications that exceed the requirements of the VoIP telephony and it is in fact a 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 IP telecommunications architecture. A clear channel approach provided a dedicated circuit, an internet backbone approach that permitted an IP transport mechanism and an IP “service bureau” approach that permitted other carriers to access the IP service backbone. A global IP network is able to offer a QoS that depends on the desired level of the QoS, 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 IP 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 a combination of the four previously mentioned schemes. A fully open IP 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, a 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 IP 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. IP 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 a network and tax communications. Hence, the VoIP networks which are a new technology are subject to government regulation and scrutiny and this is another hurdle that has existed in the rapid acceptance of a mass 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 ILD 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 an ability to provide international communications but the local charges are expected to be lower then the international charges. When considering the economics of communication services provision, the IP based systems have shown that they are the cheapest as compared to T1 lines, 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 a 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 to own 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 a 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 IP telephony has been expanded into a country is the land and build while expanding approach. When using this approach, IP network is landed 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 can also provide other services such as IP telecommunications, IP broadband internet access, ISP interconnectivity and Internet data services. An IP value chain is, therefore, possible for those who have an IP infrastructure. Earnings are possible through the provision of raw bandwidth, supported IP telephony, IP based ILD, in country ISP services, high speed internet, value added internet data, ISP contention and facilitation as well as e-commerce content. Hence, the IP 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 a 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 IP entry 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 a 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, on a 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 IP 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 for the 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 a 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 may be 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 then 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 a 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 for the 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 may be subcontracted. The numbers of corporations that are switching over to 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 WiFi 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. WiFi and 2G / 3G devices are beginning to be increasingly used but the area that is covered by WiFi 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 is, 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 IP based networks gradually. Despite this, some new network operators such as BT-Spain are already IP based.  

The availability of VoIP and IP telephony technologies has also made it possible to have instant messaging and presence management services made available, because of the IP networks and internet access. These services are converging with voice services and voice chat services that are being made available. Yahoo, MSN, AOL and others have made it possible to use chat and instant messaging with web cam services for those who have internet access and a computer. These services make it possible to contact others at great distances while avoiding the use of international telephony circuits to make a call. The services are available to both fixed and mobile users and although there are some drawbacks for business resulting from the difficulties involved with being able to establish the identity of the individual with who instant messages and chat sessions are being conducted, it is far cheaper to conduct routine matters using these very low cost services. Presence management is needed to determine if friends, business associates or others are available for contact i.e. if they are on or off line. These applications are still being developed and more refined services are likely to become available with features related to urgent calling as the technology develops. Although one of the VoIP standards that is being used for instant messaging, the SIP or RFC 3261 is highly relevant for instant messaging, the technologies that have been developed are still proprietary and not open standards. Even though Microsoft has shown some enthusiasm for SIP which has been built into the Windows XP operating system, users of instant messaging still have to use the client software that is associated with the instant messaging service that they intend to use. Corporate users have shown a keen desire to be provided with some sort of interoperability standards and have formed the Financial Services Instant Messaging Association. These services are currently free to the end users and are paid for by those who advertise on the messaging service provider’s web sites and the instant messaging companies are hoping that at some future date, they will be able to provide a product that will have the additional features which may be chargeable. Currently more then one billion messages are sent each day using instant services and there have been moves by MSN to try to licence the software that is required for instant messaging so that additional revenues may be generated. Instant messaging is, however, still growing rapidly and it does put some pressures on the income that PSTN operators are able to generate because the users do not need to make phone calls to exchange information.

It can, therefore, be concluded that the introduction of VoIP services or internet telephony has introduced a new era of competition in telecommunications and this competition has been beneficial to the users of telecommunications services, who along with enjoying the benefits of low tariffs are also enjoying the fruits of the new technologies and services being offered. However, the limitations with financing and the ability of operators to deploy new equipment have slowed the proliferation of the new technologies. The PSTN or the public switched telephone network had become old technology, having served mankind for 130 years and the new technology offers universal connectivity at all levels. The new technology has not just offered new services and devices but it has also posed new questions about how the new telecommunications arena should be regulated. It may even be said that the new telecommunications connection that is desired for an organisation or household is a broadband connection rather then the PSTN connection and it seems that the world may be moving this way.  The regulatory and policy issues that are associated with the regulation of IP networks and VoIP are discussed in the next section. 

5.1 Regulatory and Policy Issues Associated with VoIP and IP Networks

Internet telephony has opened up telephone services to competition like never before and it is possible to select the organisation which will be supplying the internet access to a home or organisation. The monopolies that controlled telephony are slowly being led out and amongst the reasons why the big telecommunications organisations like Bell, BT, AT&T as well as others are still there is because they are at the leading edge of technology and are likely to remain there because of their research programs as well as massive investments. Also, these organisations control a lot of backbone and installed capacity. Attempts have been made by governments and regulatory authorities to impose the old telephony regulations and controls on the new internet service providers with rules for market entry, exit and taxes. However, internet telephony by its universal acceptance and interconnectivity is going to discourage monopolies and hence the old regulatory framework is not necessary for the new technologies. It is generally agreed that government will have to ensure consumer protection and privacy, along with being able to monitor communications if it becomes necessary, there is no requirement to fix prices because of the competition that is available. There is a lot that has been said about the need to set the appropriate regulatory and tax framework in order to facilitate the availability of communications to all including the rural areas. Telecommunications is currently the most heavily regulated and taxed in many countries including the United States and although residential circuit switched systems benefit from a number of subsidies, VoIP systems do not gain the same benefit. Price regulation, quality of service and reporting as well as market entry requirements do not make much of a sense in a competitive marketplace, although it is the responsibility of the government to protect consumers. There have been debates about whether VoIP services should be subjected to universal service charges, access charges, emergency service requirements, law enforcement access and a host of other regulatory requirements such as the obligation to inform consumers that the services may not work in the event of a power failure. It has, however, been generally agreed that regulation should not hinder the spread of VoIP or the competition in the industry.  Hence, the wider applicability of VoIP has generated a lot of regulatory and policy debate about the applicability of the PSTN regulatory framework to the new era of telecommunications and many around the globe are thinking about how to best regulate this advance in telecommunications for the national benefit. As an example of the dilemmas that can be generated due to the introduction of a new technology the example of allowing wiretapping access to VoIP networks can be considered.  Wiretapping of switched telephony equipment has been allowed in the United States for quite a while. However, the Department of Justice had been urging the legislature to enact clear laws to permit the tapping of VoIP communications after refusal by VoIP operators to permit this.  

In the United States of America, the Federal Communications Commission or the FCC has the regulatory responsibility to oversee communications services at the Federal level. However, state utilities, corporations and commissions have an input into the development of a national policy and this input is influential. Currently, VoIP is classified as an “information service” and it is, therefore, exempt from telecommunications service regulations. The FCC has not handed down very clear cut rules in relation to the handling of VoIP services and the new technology has, therefore, caused policymakers to think harder about its regulation. Inter-carrier compensation systems in the United States which relate to the payments that have to be made between carriers for routing calls over other networks have become complex. However, because VoIP is classified as an information service, therefore, there is no requirement to pay inter and intrastate access charges for VoIP carriers. VoIP operators are also not subject to other public interest regulations or charges and this makes the technology and its use even more attractive. VoIP providers are also not required to pay a certain proportion of their income into the Federal Universal Service Fund which had been established to assist in providing telephony to rural communities and expensive to serve areas. Hence, VoIP operators cannot receive funding from this resource for their requirements. IP networks and VoIP has generated a considerable regulatory debate with operators providing services based on the new technology lobbying and asking for the establishment of voluntary compliance standards and guidelines rather then regulatory standards. This debate coupled with the rapid proliferation of VoIP will ensure that the regulatory mechanisms that currently govern telephony will be overhauled, with the result that those who are thinking of investing in new networks will have to consider their options in a new light and with a higher level of freedom with regard to financial engineering.   

One of the good things that existed with the old switched network was that the telephone number that was associated with a name and a residential address was fairly unique and could not be readily changed. IP telephony and VoIP have network addresses and not telephone numbers that are associated with telephony devices and these IP addresses can be readily duplicated on networks. Any uncoordinated allocation of addresses by an authority presents a risk of undermining the most fundamental characteristic of an address which is its unique correlation with an identity. The uniqueness of telephone numbers has also been ensured by collective action in the ITU at an intergovernmental level. The Internet Corporation for Assigned Names and Numbers or the ICANN maintains a unique register of registered domains. A coordinated approach is required to maintain an unambiguous system of IP addresses that are associated with telephone numbers, names and addresses so that there are no problems that arise in the future. Legal certainty, competitive neutrality and coordination are, therefore, essential to identify and associate IP devices with owners on networks. This can be more difficult then was the case with switched telephone systems.   

6.1 Discussion

The advances in telecommunications technologies all add up. Better networks have gradually evolved from simple things like the availability of better cabling, to latest networking technologies which have become available into faster networks with higher bandwidths as well as better interconnectivity. The new IP based networks are capable of much faster throughputs then the switched networks of the days gone by. Some estimates state that the modern networks are capable of at least ten times the throughput of the older networks with much enhanced services and these networks along with the services that they are capable of providing have become the norms of the day. The next generation means better connectivity, intelligent devices and higher productivity using these devices at a lower cost. Hence, the world is slowly switching over to the new networks and the older networks are either being upgraded or are still in use as parts of the converged networks. Networks that are in use within the companies today are capable of Gigabit transfer rates on the Ethernet standard and cater to the requirements of pervasive computing. The older networks could neither match the speed nor provide the connectivity. A modem on a switched PSTN network cannot even come close to being able to transfer at Gigabit speeds and wireless connectivity was impossible. Hence, the gradual progress in technology along with enhancements in different areas related to telecommunications has all added up into the present business norms. Organisations today don’t just need voice connectivity in order to be able to perform their tasks but they also need internet connectivity and access to the World Wide Web along with an ability to use their computing resources which have also evolved to the optimal. Organisations are now doing more with less and this means that workers have to deal with more data and require better connectivity. In order to reduce travel expenses associated with face-to- face meetings while still being able to perform the job, a relatively cheap combination of voice and data is required over networks and this is only possible with IP networks.     

As a result of the advent of VoIP and IP communications, the whole concept of a telephone call is changing. A simple switched telephone is now increasingly being considered to be an antiquated instrument and is being replaced with other equipment in which wireless connectivity is increasingly becoming the norm and mere voice interactions are considered to be insufficient. Video or video interactions in real time, the ability to quickly exchange photos and scans of documents along with a variety of other features including the means of exchanging messages such as text or voice and web access or interaction are becoming increasingly common as a result of the IP enabled networks. The digital and packet type interactions have resulted in a thinking that is more oriented towards Wide Area Networks or WAN rather then the jumble of copper wires that have been associated with the PSTN. Real time communications, open systems and an ability to bridge people, systems and technologies is what the new communications infrastructure is all about. The old network just could not have survived in the modern age of pervasive computing. File sizes are increasing and with the slow network connectivity, it would have been a nightmare trying to do things that are possible today. Hence, there has been a swing towards IP networks because they are cheaper, they are required and they can offer additional features which were not possible on the older networks.  

The IP networks are different, both in terms of the network components and also in terms of what they can offer. New devices such as gateways, gatekeepers, routers, hubs, switches, proxies, firewalls and servers etc are now increasingly a part of the corporate LAN and the WAN. Despite some problems that are associated with quality of service that needs to be carefully controlled on VoIP networks, these networks can become good revenue generators if efforts are made to control the additional delays and distortions that are likely to be introduced into these networks. Echo control is very necessary in packet networks and two or more echo controllers may be required in the voice path in order to be able to control echoes. Failure to control echoes on the VoIP networks can mean that the mouth to ear delay tolerance will make it impossible to achieve PSTN quality. It is also important to select a good codec for voice coding into packets and a performance level that is better then 32 kbits / s is essential for PSTN quality voice to be made possible. It is also important that transcoding should not be used when attempting to maintain a high level of voice quality. Packet loss is a fact of life on IP and packet based networks. The use of de-jittering buffers is essential to control the quality of voice deterioration that is likely to occur as a result of packet losses. Packet loss concealment techniques can considerably enhance performance but care should be taken when employing these techniques that the computation time involved in processing for de-jittering should not exceed the maximum delay time associated with acceptable quality of speech. This time is ideally about 100msec and if the delay time approaches 600msec, there can be serious deterioration in the ability to decipher the speech conversation that is being attempted. There are various techniques that may be employed for de-jittering on VoIP networks but adaptive de-jittering mechanisms are more appropriate as compared the static de-jittering mechanisms. Adaptive de-jittering mechanisms are more likely to adjust their correction mechanism for packet loss with changing conditions of ambiance noise and the statistical properties of the IP network as compared to the static de-jittering schemes. However, despite these problems that are likely to occur on VoIP networks, reputable manufacturers such as Alcatel, Cisco and Juniper Networks etc have demonstrated that it is possible to manufacture equipment that can deliver PSTN speech quality on VoIP networks.  Greater freedom is possible in setting gateway parameters with attempts being made to maintain a good control over echoes on the network and it is quite possible to have VoIP networks with comparable quality to PSTN networks if the proper design expertise is employed and network equipment is selected carefully.  Delays in voice processing are usually introduced at the VoIP telephone, IP router and switches, IP to PSTN gateway, the wires and other delays in the PSTN network system. Security on VoIP networks and the ability to designate unique telephone numbers that are linked to a unique address and individual are also concerns that have been associated with the new technology. However, because there is an ability to overcome the potential problems that have been associated with VoIP technology, it can be assumed that the technology has matured and it is ready for even greater penetration into the business or corporate world.  

According to IDC, 37% of the large and medium sized firms in the United States of America have already deployed VoIP based equipment in their organisations. The trends in Europe usually follow technology trends in the United States of America and the rest of the world also follows these trends. Hence, it can be argued that the markets are seeing a significant level of VoIP penetration which is going to continue to increase significantly when it is considered that it has also been predicted that the global revenues from sales of VoIP equipment is very likely to exceed £ 2.5 billion in the year 2007, up from £ 0.7 billion in 2003. Such trends indicate that the telecommunications networks associated with speech will slowly become VoIP based with an end to the switched network telephony. Standardised equipment which is becoming available helps businesses in gaining confidence in a new technology and VoIP helps reduce capital investments and operating expenses for telecommunications operators. The old technology was just not capable of being used in the pervasive computing age because it could not satisfy the requirements that were presented related to the need for multimedia interactions and transfers for computing. With requirements for greater data transfers as a result of the need to transfer larger files between businesses which had come about as a result of the increasing size of computer files for applications and a need for computers to constantly interact with each other to support corporate level interactions such as supply chain management, web services interactions and automated business processes, the premium has shifted towards data communications. It was wasteful to have two or more networks which would have fulfilled the needs associated with data transfers, voice transfers or other multimedia transfers such as video or which were capable of supporting web interactions. Hence, one converged network architecture was needed that could transport everything faster and also offer universal connectivity. There is also a trend away from wired networks and wireless is being preferred because it is less messy, more flexible and offers mobility for the users. The way in which technology evolved and which was most supportive of the requirements related to wireless flexibility, speed, connectivity and the ability to provide additional features was towards the use of IP networks. Apart from IP, there were other standards which were also involved. The acceptance of XML related standards for data storage and data encapsulation along with the use of Voice XML all needed the abilities that are associated with the internet protocol or IP. The switched networks were energy inefficient, with cumbersome equipment which could not sustain the speeds that were required even with the use of high speed modems that were able to be developed as a result of new QPSK modulation schemes. Hence, the impact of VoIP and IP based networks on switched telephony has been one of a killer application which will ultimately make the old technologies associated with network switching to become redundant. Computers need digital data networks and not analogue voice networks. An acceptable way has been found to transport analogue signals along with nearly everything else in real-time over the digital data networks which are required for the pervasive age.  

VoIP Market Share

If the tributes that have been presented by satisfied organisations who have shifted to VoIP networks and converged telecommunication systems are to be considered, then it can be safely concluded that the voice echo, jitter, delay and quality issues that have been mentioned previously in connection with IP telecommunications have indeed been overcome and the technology can be safely recommended for use, even by the large organisations with a large number of staff and offices that are separated by very large distances.     At Cisco Corporation, it is claimed that all employees have IP phones for their use. These phones must be working correctly otherwise Cisco would not be doing as well as it is. It has been said that with the right mix of legacy systems and IP telephony equipment and networks in an organisation, a 70% return on investment is possible with 30% of the investment. When considering the adoption or changeover to VoIP networks, it is important to think beyond the immediate horizon. The technology appears to have achieved a proven track record and the services that it offers are the future. These enhancements could not have been made possible with switched telephony. Cost savings that can be made as a result of the adoption of a new technology consist of what have been termed as “hard” or easy to quantify cost savings and the so called “soft” savings for which it is difficult to put a value on. The replacement of a PBX with a server can certainly save an organisation a certain amount every year but the ability to upgrade the software on the server and to have unified messaging are benefits which are difficult to quantify, but which are really there because of the ability of an organisation to take a sudden leap forward when it is so desired. Even though the cost of suddenly switching to VoIP or converged networks may appear to be daunting because of the need to acquire a lot of new equipment, leasing options make it possible to spread the cost over the years and the leasing company is very unlikely to permit an investment on useless equipment. Hence, even though the PSTN network operators tried to put up a fight by reducing long distance call rates very substantially at the time of the introduction of VoIP, they could not kill the new technology because the sums and the advantages added up. Cost savings in VoIP occur as a result of savings that are made on equipment and maintenance, network carrier costs and network management. The regulatory environment also classifies VoIP as being different from switched telephony and hence, the taxes, access charges and payments that operators are required to make for IP networks are lower then what would have been incurred as a result of investments that would have been required for switched telephony. Having one network for both voice and data certainly adds up to cost savings and hence investment. Having VoIP can not only reduce the cost of ownership, but this technology also reduces the incremental cost associated with having additional VoIP phones and network users being added to the network. Incremental costs are not only for single users but also for group of users. Thus adding new corporate offices to existing VoIP networks is cheaper then doing the same with switched telephony networks. Wiring costs are reduced because there is no requirement for providing two sets of wires and a single pair will be sufficient for both voice and data. The IP network that has been installed in an office can also readily accept a wireless infrastructure which may be required in an office for flexibility and convenience as well as enhanced productivity.     

Centralised call processing is provided and readily implemented in VoIP, which makes it possible for many organisations to install core processing capabilities for their network in one or several sites from which VoIP facilities may be extended throughout the organisation. Centralised call processing allows standardisation of services that may be required to be provided within organisations and reduces the overall costs associated with equipment, maintenance and network management. It is possible for one team to manage the entire organisation’s network from one centralised site with assistance from automated fault location systems that can narrow faults on the IP network. The job functions of those who look after the PBX and those who look after the data network in an organisation merge and a single job function emerges. Even though the personnel who are likely to be assigned to look after IP networks are likely to be more qualified and hard to find as compared to those who are experts in switched telephony, there are savings on personnel because fewer individuals are required. The Dynamic Host Control Protocol or the DHCP enables a VoIP device to automatically reconfigure itself when moved. Hence the cost associated with moving a piece of equipment on the IP network is considerably lower then that for switched telephony systems. Costs associated with setting up new sites are reduced and portability is made available, permitting an individual user to access any services or applications from any corporate phone in the organisation. Easy addition of new features advanced routing and integration into business etc have already been mentioned as the added benefits of VoIP telephony. Despite the training costs associated with VoIP, which can be high and the business risks, the benefits and the costs associated with VoIP by far outweighs the obstacles in the implementation of IP networks. The enhanced competition in telecommunications that has been unleashed at all levels is also good for telecommunications, the users of telecommunications and those who are the service providers because they can formulate the right strategies in an attempt to tap a much larger consumer base. 

Hence, it can only be concluded that any future additions or implementations of telephony networks are very likely to be using IP based networks with a converged voice and data network. The advantages and the potential of VoIP are great both in terms of future requirements, features that are available and the costs savings that are possible. VoIP is a killer application for switched telephony networks which are gradually going to be phased out from use after having become obsolete and economically unviable. Market reports have indicated that VoIP telephony has grown from nothing to a multi-billion dollar industry since its introduction and that it is expected that IP telephony will be taking up about 35 - 50% share of the industry by the year 2015. The total telephony market is worth about 100 billion dollars per annum and the only reason why VoIP will not be capturing a greater share is because old networks which have substantial investments associated with them just cannot be discarded.  Also, global telephony consists of much more then just voice telephony circuits with substantial investments in wireless and cellular communication systems, undersea cables and fibre optic communication systems as well as radio and microwave telephony. There are very good chances that a lot of corporate and business users will have switched to VoIP telephony by the year 2015 because of the cost effectiveness and added services. IP networking is suited for use in backbone networks for telecommunications in addition to corporate networks, whereas PSTN, ISDN and cellular GSM channels permit fixed bandwidth connectivity. However, IP is the glue that binds it all and IP protocols are capable of being implemented on all connections including ISDN, Broadband, mobile and cable etc.  

Even though VoIP telephony has been accepted as a universal standard, its implementations can be different. Various manufacturers have used their own approaches to making VoIP telephony into a practical reality and these varied approaches can be important when considering the upgrading of a switched corporate network into a VoIP or converged network. Judicious selection can result in economies and a better fit for the new equipment with the legacy telecommunications infrastructure, resulting in economies. Some VoIP implementations may converge better with existing switched installations to be found in organisations, while other manufactured equipment may be better suited to the other users. Cisco system’s implementation of VoIP for use in corporate networking is called AVVID or Architecture for Voice, Video and Integrated Data which has a more centralised approach to implementing VoIP as compared to some other manufacturers such as Shoreline Communications, which offer a more distributed approach. It has been claimed that Cisco’s architecture is more complex and can be expensive to implement when upgrading existing corporate networks. However, performance and a centralised VoIP implementation may offer savings when considering the costs associated with maintenance of corporate networks. The important thing is that the International Telecommunications Union or the ITU’s VoIP standard recommendations can be implemented in a number of ways. Whereas Cisco’s AVVID approach requires a gateway, gatekeeper and terminal devices, other manufacturers have ventured along the path to a more distributed approach in their product designs. In the Shoreline Communications design approach, for instance, the functions of the gateways and the gatekeepers have been incorporated into each of the VoIP devices that may be connected to the network. Carefully selecting the right architecture for corporate needs can result in lower implementation costs, maintenance costs, LAN infrastructure costs and lower revenues lost due to downtime. Hence, having established the major benefits of the VoIP protocol, it is very likely that the product designers and manufacturers will compete on the basis of the relative advantages and disadvantages of their designs and implementations of VoIP. Such competition is by itself healthy because better products and ideas will emerge as a result of the efforts associated with different approaches.  

In the light of the available comments and reports that are available from published sources including those presented by business, manufacturers and academic investigations, it can only be concluded that VoIP is a standard in telephony which has come to stay. Its adoption is accelerating and it is turning into a killer application for switched telephony. A very significant level of penetration of voice over IP networks is expected to be completed by the year 2015 and business, network service providers as well as users around the globe have realised the cost effectiveness and added functionality of VoIP as well as IP networking. VoIP is the technology for the new age of pervasive computing and the switched networks were just not up the tasks that may very well be required in the future on a mass scale.     

7.1 Conclusion

In the decades gone by, the most important requirements for communication in the society at large were associated with personal and business communications, without there being a great need for the exchange of data. However, with the evolution of computing and the dawning of the pervasive computing age, exchange of data between computing assets has emerged as an important requirement for business as well as individuals. Hence, data networks in which all types of media can be exchanged over packet based networks have assumed a very considerable importance. With such an evolution, it was important to find a telecommunications solution that will permit universal connectivity over a single network, without there being a need to have two separate networks for voice communications and data. The internet protocol or the IP emerged as the glue that could transport just about any media after converting the content into a stream of packets to be sent over packet networks. Voice over the internet protocol or VoIP refers to the real time transport of speech which has been encoded into packets using the internet protocol, in real time. This technology has had profound implications for the future of telephony and switched networks are likely to be converted into IP networks as VoIP penetrates into the global telecommunications market. Many of the problems that had existed with VoIP including those related to the quality of service and acceptance have been ironed out and the technology has now matured. It is very likely that VoIP based products will account for as much as 50% of the global telecommunications requirements by the year 2015, worth about £ 50 billion. The technology is relatively cheap and offers advantages for connectivity, added services and productivity. Hence, VoIP has come to stay and is proving to be a killer application for switched telephony.

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References Related to VoIP and Telephony from British Libraries

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12. Bates, Juliet. Optimizing Voice in ATM/IP Mobile Networks. New York ; London : McGraw-Hill, 2002.
13. Bayer, Michael. Computer Telephony Demystified : Putting CTI, Media Services, and IP Telephony to Work. New York ; London : McGraw-Hill, 2001.
14. Bellamy, John. Digital Telephony. 3rd Ed. New York ; Chichester : John Wiley & Sons, 2000.
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17. Berridge, Lucy, Tawanda Chihota, and Olivia Gibney. Africa and the Middle East : Mobile, Fixed Telecoms and Internet Opportunities. 2nd ed. London : Baskerville Strategic Research, 2001.
18. Beyh, Sébastien and University of Salford. Computer & Communication Engineering : Internet Protocol Telephony in Construction. Salford : University of Salford, 2004.
19. Billingnews.Com : the Leading Billing, IP Telephony and Switching Print and Online Publication. London : Rock, 2001.
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24. Blenheim, Online. Personal Communication Networks and Digital Cordless Telephony : Proceedings of the Conference Held. Pinner : Blenheim Online.
25. Blery, Evangelia and University of Surrey. Factors Influencing Customers' Repurchase Intentions in the Greek Mobile Telephony Sector. Guildford : University of Surrey, 2003.
26. Blowers, Mark, et al. Communications Convergence : Evolving to a Next Generation IP-Based Network. [Hull] : Butler Direct Ltd., 2004, 2004.
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29. Broadband Local Access Sourcebook : Analysis and Contacts for the Broadband Local Loop Across Europe. London : Baskerville, 2001.
30. Brown, Kevin. IP Telephony Unveiled. Indianapolis, Ind. : Cisco Press, 2004.
31. Bwalya, Yese Williams and London Panos. Completing the Revolution : the Challenge of Rural Telephony in Africa. London : Panos Institute, 2004.
32. Camarillo, Gonzalo and Miguel A. García-Martín. The 3G IP Multimedia Subsystem (IMS) : Mergining the Internet and the Cellular Worlds. Chichester : Wiley, 2004.
33. Camarillo, Gonzalo and Miguel A. García-Martín. The 3G IP Multimedia Subsystem (IMS) : Mergining the Internet and the Cellular Worlds. Chichester : Wiley, 2004.
34. Camarillo, Gonzalo. SIP Demystified. New York : McGraw-Hill, 2002.
35. Camp, Ken. IP Telephony Demystified. New York, N.Y. ; London : McGraw-Hill, 2003.
36. Caputo, Robert. Cisco Packetized Voice and Data Integration. New York ; London : McGraw-Hill, 2000.
37. Chan, Yiu Ting, Silvia Massini, and School of Management. Mobile Telephony Diffusion : Case Studies of the UK & Hong Kong. Manchester : UMIST, 2002.
38. Chorleywood, Consulting. GPRS and 3G Billing. London : Chorleywood Consulting, 2001.
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56. Downing, Jeremy and Welsh Consumer Council. Mobile Telephony in Wales = Teleffoni Symudol Yng Nghymru. Cardiff : Welsh Consumar Council, 2002.
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58. EBSCO Publishing (Firm). Communications Convergence. San Francisco, Calif. : CMP Media.
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60. Ellis, Juanita, Charles Pursell, and Joy Rahman. Voice, Video, and Data Network Convergence : Architecture and Design, From VoIP to Wireless. Amsterdam ; London : Academic Press, 2003.
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62. Evans, D. R. Digital Telephony Over Cable : the Packetcable Network. Boston ; London : Addison-Wesley, 2001.
63. Faynberg, Igor, Lawrence Gabuzda, and Hui Lan Lu. Converged Networks and Services : Internetworking IP and the PSTN. New York ; Chichester : Wiley, 2000.
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66. FT Media & Telecoms. Internet Telephony. London : Financial Times Media & Telecoms.
67. Giralt, Paul, Addis Hallmark, and Anne Smith. Troubleshooting Cisco IP Telephony. Indianapolis, Ind. : Cisco Press, 2002.
68. Global Telephony. Overland Park : Intertec Publishing.
69. Great Britain. Business Statistics Office and Great Britain. Central Statistical Office. Business Monitor. Report on the census of production: telecommunication equipment, electrical measuring equipment, electronic capital goods and passive electronic components ed. London : HMSO.
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71. Great Britain. Dept of Trade and Industry. Radiocommunications Agency. Cordless Telephony : the Future of Analogue and CT2 Cordless Telephony in the United Kingdom: a Consultative Document. Radiocommunications Agency, 1999.
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73. Great Britain. Office of Telecommunications. Local Loop Unbundling : the Term of the Access Network Facilities Agreement. London : OFTEL, 2001.
74. Great Britain. Office of Telecommunications. Protecting Consumers by Promoting Competition : Consultation on Oftel's Review of the Fixed Telephony Market : 31 January 2002. London : Oftel, 2002.
75. Great Britain. Office of Telecommunications. Statement and Determination on Local Loop Unbundling 'Bow Wave Process'. London : OFTEL, 2000.
76. Greenblatt, David J. The Call Heard 'Round the World : Voice Over Internet Protocol and the Quest for Convergence. New York ; London : AMACOM : McGraw-Hill, 2003.
77. Greetham, D. C. and Bob Emmerson. Computer Telephony and Wireless Technologies : Future Directions in Communications. Charleston, S.C. : Computer Technology Research Corp, 1997.
78. Grigonis, Richard. Computer Telephony Encyclopedia. New York : CMP Books, 2000.
79. Grigonis, Richard. Voice Over DSL : Understanding and Implementing VoDSL. Gilroy, Calif. ; [London : CMP Books : McGraw-Hill], 2002.
80. Groom, Frank M., Kevin M. Groom, and International Engineering Consortium. The Basics of Voice Over Internet Protocol. Chicago, Ill. : International Engineering Consortium, 2004.
81. Hardy, William C. VoIP Service Quality : Measuring and Evaluating Packet-Switched Voice. New York ; London : McGraw-Hill, 2003.
82. Hersent, Olivier, David Gurle, and Jean Pierre Petit. IP Telephony : Packet-Based Multimedia Communications Systems. Harlow : Addison-Wesley, 2000.
83. Hersent, Olivier, Jean Pierre Petit, and David Gurle. Beyond VoIP Protocols : Understanding Voice Technology and Networking Techniques for IP Telephony. Chichester : Wiley, 2005.
84. Hersent, Olivier, Jean Pierre Petit, and David Gurle. IP Telephony : Deploying Voice-Over IP Protocols. Chichester : Wiley, 2005.
85. Hopkins, Margaret, Catherine Dyer, and Analysys Research Limited. IP Voice Services : European Corporate Market Forecasts, 2002-2007. Cambridge : Analysys, 2002.
86. Huang, Tingxue and University of Stirling. Dept.of Computing Science and Mathematics. Policies for H.323 Internet Telephony. Stirling : University of Stirling, Dept. of Computing Science and Mathematics, 2004.
87. Hubbard, Sue and Consulting Chorleywood. IP Mediation. London : Chorleywood Consulting, 2001.
88. Ibe, Oliver C. Converged Network Architectures : Delivering Voice Over IP, ATM, and Frame Relay. New York : Wiley, 2002.
89. Informa. Informa Telecoms Group. Mobile Media : Messaging, Content and Applications : MM. London : Informa Telecoms Group.
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91. IP Wireline & Wireless Week. London : Euromoney Institutional Investor.
92. Jin, Young Il. The Study on Delay Variance Control of Internet Telephony. 2001.
93. Johnston, Alan B. SIP : Understanding the Session Initiation Protocol. 2nd ed. ed. Boston ; London : Artech House, 2004.
94. Kee, Richard and Ovum (Firm). Ovum Forecasts : Global Telecoms and IP Markets. London : Ovum Ltd., 2000.
95. Khasnabish, Bhumip. Implementing Voice Over IP. Hoboken, N.J. ; Chichester : Wiley, 2003.
96. Kolon, Matthew C. and Walter Goralski. IP Telephony. New York ; London : McGraw-Hill, 2000.
97. Koolwaaij, Johan and Katholieke Universiteit Nijmegen.  Automatic Speaker Verification in Telephony : a Probabilistic Approach. [Nijmegen? : Katholieke Universiteit Nijmegen?], 2000.
98. Kumar, Vineet, Markku Korpi, and Senthil Sengodan. IP Telephony With H.323 : Architectures for Unified Networks and Integrated Services. New York ; Chichester : Wiley, 2001.
99. Laino, Jane. The Telephony Tutorials : a Practical Guide for Managing Business Telecommunications Resources. San Francisco, Calif. : Telecom Books, 2000.
100. Lane, Nick. IP-Core Networks : Legacy Networks Versus Next Generation Networks. London : Informa, 2001.
101. Lane, Nick. IP-Core Networks : Legacy Networks Versus Next Generation Networks. London : Baskerville Communications, 2001.
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103. Lehr, William, Lee W. McKnight, and David D. Clark. Internet Telephony. Cambridge, Mass. ; London : MIT Press, 2001.
104. Ling, Rich and Telenor. The Adoption of Mobile Telephony Among Norwegian Teens, May 2000. Kjeller : Telenor R & D, 2000.
105. Lovell, David. CISCO IP Telephony. Indianapolis, Ind. ; [Great Britain] : Cisco Press, 2002.
106. Massini, Silvia, University of Manchester. Centre for Research on Innovation and, and Competition. The Diffusion of Mobile Telefony in Italy and the UK : an Empirical Investigation. Manchester : Centre for Research on Innovation and Competition, University of Manchester, 2002.
107. Mathenge, Peter, Dale Littler, and School of Management. What Comes Next? : Consumer Lifestyles, Product Symbolism and Third Generation (3G) Mobile Telephony. Manchester : UMIST, 2002.
108. McCarthy, Helen and Paul Miller. London Calling : How Mobile Technologies Can Transform a City. London : Demos, 2003.
109. McKeown, Max. Call Centres in a Digital World. London : LLP Professional, 2000.
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114. Miller, Mark A. Voice Over IP : Strategies for the Converged Network. Foster City, Calif. : M&T Books, 2000.
115. Mina, Andrea. Knowledge Co-Ordination and the Emergence of Standards: the Development of the Gsm-Based Market for Mobile Telephony. Manchester : University of Manchester, 2004.
116. Minoi, Jacey Lynn, P. Green, and Computation. Navigation and Orientation With Mobile Telephony. Manchester : UMIST, 2001.
117. Minoli, Daniel and Emma Minoli. Delivering Voice Over IP Networks. 2nd ed. Indianapolis, Ind. : Wiley, 2002.
118. Mobile Internet : Content, Commerce and Applications. London : Informa Telecoms Group, 2002.
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126. Odom, Wendell and Michael J. Cavanaugh. IP Telephony Self-Study : Cisco QOS Exam Certification Guide. 2nd ed. Indianapolis, Ind. : Cisco, 2004.
127. Oftel. Competition in the Provision of Fixed Telephony Services. London : OFTEL, 2001.
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131. O'Neill, Martin, Great Britain. Parliament. House of Commons. Trade and Industry, and Committee. Mobile Phone Masts : Report, Together With the Proceedings of the Committee, Minutes of Evidence and Appendices : Tenth Report Session 2000-01. London : Stationery Office, 2001.
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