PSTN vs VOIP

Introduction

Inherently, innovations in technology such as the Voice over Internet Protocol (VoIP) are coming up with the aim of replacing the former technology, specifically, public switched network PSTN. More importantly, the innovations target at reducing cost in addition to introducing new features in the communication industry.

Public switched telephone network (PSTN) refers to the global collections of interconnections, and consists of telephone lines, microwave transmission links, communication satellites, fiber optic cables, together with cellular networks. Consequently, it gives a chance for any telephone in the world to connect to any other. Voice over Internet Protocol (VoIP) also referred to as internet telephony, describes a transmission of a two way voice over the internet in real, or near real time (Federal Communication Commission 2006). Most important, VoIP is relatively newer technology as compared with PSTN, and in fact it is a better substitute of PSTN. Voice over IP interconnects with PSTN, where the VoIP users are in a position to call other people on standard phones and vice versa. Inherently, the idea here is the fact that the voice data are transmitted through the internet to a point which is near to the end of a phone call, consequently, it offers a relatively cheaper mode of communication.

Objective of the Study

The objective of this paper is to conduct a comparative study between the Voice over Internet Protocol (VoIP) and public switched network PSTN

PSTN

Public switched telephone network (PSTN) is the global collections of interconnections which consists of microwave transmission links, telephone lines, communication satellites, fiber optic cables, together with cellular networks. Therefore, it gives an opportunity for any telephone in the world to be connected to any other telephone.

Technological Infrastructure

Most importantly, PSTN makes use of standards that were created by the ITU-T, which gives a chance for different networks from a number of countries to interconnect seamlessly. Evidently, the PSTN infrastructure has evolved over the years in a bid to support the increasing number of subscribers, connections to other countries and calls (Linden 2004). A good number of automated telephone exchanges utilize digital switching. Similarly, the trunks that connect these exchanges are digital, and they are referred to as circuits or channels.

For instance to transmit a phone call, the analogue audio signal will first need to be digitised at an 8 kHz frequency while using 8-bit resolution with the help of a nonlinear pulse code modulation that is called G.711. Consequently, the call is later transmitted through a telephone exchange, where it is controlled with use of call set up protocol in the telephone exchange. More importantly, the call is transmitted over the PSTN with use of a 64 bit channel.

Inherently, PSTN is comprised of three components: customer premises equipment (CPE), transmission facilities, and the switching system. A pair of copper wires in form of a local loop runs between a local central office and the subscriber. Similarly, the local central office then connects to the local tandem switch, which then connects to the higher tandem switch. More importantly, the switches are connected by the use of trunks (Hardman 2003). It is also worthy to note that some portion of the PSTN goes through a number of levels of hierarchy, and can even be as many as five. In PSTN making a call will obviously need a communication circuit between the two people communicating. Moreover, both the set up and the release of connection are activated through the use of signals.

Bandwidth Requirements

Basically, PSTN supports a bandwidth of 64kbps, although traditional dial up modem supports even 56 kbps bandwidth.

Quality of Service

This actually rates the performance, which is normally described through real-time quality of services which put into consideration the bandwidth, packet loss, jitter and latency. Inherently they effectively portray the aggregate quality of service and for PSTN the QoS has been refined for circuit switches (Linden 2004).

Quality of Experience

Moreover, due to congestion in the network, the quality of connection at the end user is significantly affected, specifically, it is low.

Traffic Engineering

This describes a method through which the performance is optimised through analysing and regulating behaviour of data transmitted dynamically. In PSTN traffic engineering is executed through analysing the bandwidth, packet loss, jitter and latency (Bush and Meyer 2004).

Billing Aspects

In a prepaid PSTN, one will need to have a certain minimum balance to be in a position to make calls. The cost of a call in PSTN is deducted from the credit which is available every time one makes a call (Broom and Hollier 2003).

Future Prospects

Actually, PSTN is currently faced with challenges of transitioning to VoIP, owing to the fact that VoIP is more technologically advanced in addition to being a cheap method of communication as compared to PSTN.

VoIP

Voice over Internet Protocol (VoIP) also referred to as internet telephony, describes a transmission of a two way voice over the internet in real or near real time. Most important, VoIP is relatively newer technology as compared with PSTN, and in fact it is a better substitute of PSTN. Essentially, according to Zubey, Wager and Otto (2002) voice over IP interconnects with PSTN, where the VoIP users are in a position to call other people on standard phones and vice versa. Inherently, the idea here is the fact that the voice data is transmitted through the internet to a point which near to the end of a phone call, consequently, it offers a relatively cheaper mode of communication.

More importantly, Voice over Internet Protocol mostly target transferring of voice based messages together with applications through the use of diverse protocols and transmition through the internet. Basically, the main steps that are involved in transmission of voice data via the internet include the conversion of the voice signal into analogue and consequently to digital signal. Subsequently, there is compression together with conversion of the signal to the internet protocol in order to broadcast it over the internet (Amirah 2006). It is also worth noting that VoIP system assumes a different session control protocol while commanding over set up, tearing down of calls, and special audio codec that gives a chance to encode the voice signal in order to make transmission possible.

Notably, the technologies that are used in implementing VoIP include H.323, IP Multimedia subsystem (IMS), Real time Transport Protocol (RTP), as well as Session initiation Protocol (SIP). A VoIP network topology consists of a combination of various equipments which also include a gatekeeper, VoIP gateway, and VoIP clients. A VoIP gatekeeper is actually a routing manager along with central manager in H.323 IP telephony surrounding. This offers an option in VoIP system that will manage the end points of a sector. More importantly, VoIP gatekeeper is very crucial while managing calls, terminal together with gateways. In addition, the VoIP gatekeeper allows access control, address translation and bandwidth control. Besides, VoIP entry or gateway converts the voice calls into a genuine instant in the middle of Public switch telephone Network (PSTN) and IP networks. Moreover, its basic functionalities include compression, decompression, packetization, signal controlling along with call routing. Further, VoIP clients represent the phones and multimedia PC’s (Zubey, Wager and Otto, 2002).

Quality of Service (QoS)

Inherently, the quality of service is a normal process of VoIP. It enables VoIP to deliver a good quality of service to the users and in fact it is advantageous to the users since it allows them to save money, instead of spending more money through other communication services. More importantly, the security issues are known to degrade the quality of service of the VoIP (Kos, Klepec and Toazic 2004). The main security issues include latency, jitter, packet loss together with bandwidth problem. Latency is actually the delivery time of the voice transmission from its source to the preferred destination. The inherent issues that affect latency can be summarized to security measures, packetization, composition and decomposition, voice data encoding as well as decoding. On the other hand jitter represents the non-uniform packet that is associated with packet delivery delay which is normally caused by insufficient bandwidth. Similarly packet loss relates directly to both latency and jitter, and is associated with data loss. Subsequently, low bandwidth will actually delay packet delivery, consequently, degrading the quality of service through increasing latency and jitter.

Bandwidth Requirements

Jaiswal, and Raghav (2004) claim that in voice over IP the standard method that is used to transport voice data though an IP network will normally require IP, UDP, as well as RTP. Inherently, while an IP is 20 octets, and a UDP header is 8 octets, an RTP header is normally 12 octets. Consequently, the total length for header adds up to 40 octets (bytes) equivalent to 320 bits and these headers are actually sent every time the voice data is transmitted. Besides, the additional bandwidth that is occupied by the information is identified by the number of packets sent every second. For instance, if one packet transmits voice data representing 20 milliseconds, then 50 such samples will be required every second. Now since every sample transmits 320 bits, it then translates that every second 16000 header bits are transmitted and consequently it can be presumed that the header information will include 16 kbps to the bandwidth requirement for VoIP.

Quality of Experience

Inherently, the quality of experience is gradually gaining popularity. Most importantly, it is worth noting that QoE is closely associated with human perception and actually serves as a more valuable indicator from the user perspective. According to a research conducted by Jaiswal, and Raghav (2004) the quality of experience evaluation of VoIP revealed the consideration of human affective attributes in the modelling of users experience.

Traffic Engineering

The main traffic engineering issues include jitter and packet loss. Jitter represents the non-uniform packet that is associated with packet delivery delay which is normally caused by insufficient bandwidth. Similarly packet loss relates directly to both latency and jitter, and is associated with data loss (Zubey, Wager and Otto 2002).

Billing Aspects

VoIP users experience lower recurring transmission charges in addition to reduced long-term ownership cost. More importantly, through directing voice calls through a corporate data network, corporations are able to significantly reduce their communication bills. The fact that these calls consume relatively little bandwidth, the calls are typically a fraction of the recurring cost that is normally charged by carries in order to transmit the same calling volume (Kos, Klepec and Toazic 2004). Additionally, the coverage network architecture similarly cuts down the on-going cost of owning two different networks: one is for the data while the other one is for the voice.

Future Prospects

More importantly, the next level of VoIP implementation will put into consideration various VoIP protocols together with their features and at the same time configuring on over WLAN along with Ethernet LAN. Inherently, this will ensure that the complete mechanism improves the system throughout the packet delivery process since it will apply some input statistics as in the simulation of object modeller (Zubey, Wager and Otto 2002).

Comparison

Whereas PSTN is considered to be reliable, VOIP is not reliable as compared to PSTN.
Moreover, PSTN does not only offer excellent audio quality, it also has dedicated lines that are required from Telco provider. On the other hand, VoIP quality is dependent on the internet bandwidth availability as well as the calls can be transferred over the internet connection (Gansola and Heyer).

Similarly, it is more costly to maintain and upgrade PSTN PBX than to maintain VoIP PBX network.
PSTN can only transfer audio calls while VoIP can transfer audio, video as well as data during a call.
PSTN is more suitable for emergency calls since the location can be traced and in contrast, VoIP is not suitable for emergency calls since the location cannot be traced.

In PSTN, each analogue telephone uses a bandwidth of 64 kbps and in all direction while VoIP can use bandwidth of even 10 kbps in each direction.

It is more complex to both expand and upgrade PSTN since it will require additional hardware and provisions of new lines, but on the contrary, VoIP only needs more internet bandwidth in addition to software upgrades, and hence it is cheaper.
PSTN calls are relatively costly, while on the other hand, VoIP calls are cheaper.

Recommendations

Essentially, with advancing technologies there is a need for more than just effective voice communication, and this demand invites in VoIP. VoIP is hence expected to offer high QoS to the users, despite the fact that the technology will need to work in conjunction with PSTN and not really to replace it.

Amirah (2006) asserts that performance is an important factor in the VoIP system; therefore there is a dire need for the system to improve on the quality through comprehending the network traffic, along with security issues and other peripheral issues which may include packet delay, together with queuing mechanism. Despite this, there is also a need to analyse packet delivery, bandwidth sharing, and network throughput without forgetting to improve latency, as well as jitter and packet delivery. This can be done through analyses of the physical network together with routing information and decoding. Inherently, the codec is necessary to decode the voice signals, while the packet delay on the other hand is related to both queuing delay and network delay.

Consequently, it will be required to improve packet delay through propagation of network traffic as well as packets, moreover, the security process will also enhance jitter, together with packet delay and latency while at the same time the encryption process will increase the size of the packet which will subsequently increase the network payload.

Conclusion

It is clear that PSTN is very effective in carrying voice traffic although it does not fit the current requirements of the society. Consequently, with advancing technologies there is a need for more than just effective voice communication, and this demand invites in VoIP. VoIP is hence expected to offer high QoS to the users, despite the fact that the technology will need to work in conjunction with PSTN and not really to replace it.

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