VOIP SOLUTION

VoIP solution



VoIP solution

Introduction

This proposal will attempt to address the benefits and the limitations to the implementation of a Voice over Internet Provider (VoIP) telephony communication system. The advantages include a new versatility never known to the traditional Public Switched Telephone Network (PSTN) and a cost savings due to the efficient nature of packet-switched telephony. VoIP applications can satisfy many features of the PSTN system such as voice and fax, but can also expand communication into new areas for business telephony. Voice over Instant Messenger to expand the IM medium and Push-to-talk for lower bandwidth calls where quality is not a priority. Perhaps the most beneficial application for business applications will be in conferencing. The redundant nature of VoIP calling will allow voice message information to be encoded and then copied to every recipient while keeping bandwidth requirements on the network low (Weinstein, 2009).

This efficiency in the VoIP arena is necessary in order to maintain a high quality of service (QoS) for packet-switched communication. While VoIP technology provides a versatile medium that can go beyond the means of PSTN, there exist trade-offs inherent to the nature of packet -switched transfer over the existing Internet designed for data routing. The proposal will consider the Quality of Service standards and protocols needed to combat the performance issues that are introduced to a packet-switched voice transmission. To ensure that a communication is of a high quality and can be understood, there are three areas that have to be controlled; latency, packet loss and jitter. Latency refers to the concept that all VoIP packets must be received at the same time in order to for the voice that was transmitted to not be confusing to the recipient. Any variation of a packet arrival is called delay. The total delay experienced by the end-user in the communication is referred to as latency. Packet loss causes similarly confusing voice calls because there is not only a delay of a packet of information but a complete omission of that piece of the call. Packet loss happens when a router's output buffer is too congested with information and a packet is simply dropped. Because of the real-time requirements of VoIP, retransmission of the information is useless to the quality of the call. The current Internet cannot possibly deliver all VoIP packets over the same paths and through the same routers. The delay caused by several packets having differing experiences in their transmission and the resultant latency variations over many packets is known as jitter.

While packet-switched voice communication faces a host of performance issues particular to the medium, VoIP offers an expanded range of communication possibilities that can be attained with the right implementation. Utilizing standards and protocols specific to the QoS needs of VoIP and the development of an Overlay Network to allow for routing privileges over and above standard packet transmission will be the foundation of a new viable and cost effective communication network for business.

Problem Statement

The implementation of a VoIP telephony system will offer ...