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

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

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

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

1.1 Introduction

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

VoIP Architecture

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

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

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

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

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

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

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

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

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

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

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

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

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

2.1 Products, Services and Issues Related to VoIP

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

Mere telephony will cease to exist in the future and will be replaced with capabilities for multimodal integration involving speech, text, pictures and web interactions that can take place through instruments that will replace the simple telephone of the days gone by. It will be possible for organisations and call centres to interact at a much superior level, with those who interact with them and such interactions can involve quick access to