April 25, 2005
Please note that contributed articles, blog entries, and comments posted on EDACafe.com are the views and opinion of the author and do not necessarily represent the views and opinions of the management and staff of Internet Business Systems and its subsidiary web-sites.
Jack Horgan - Contributing Editor

by Jack Horgan - Contributing Editor
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Voice over Internet Protocol (VoIP) also referred to as Internet Voice and as IP Telephony is a technology that allows users to make telephone calls using a broadband Internet connection (high speed cable or DSL) instead of a regular (or analog) phone line. Some services using VoIP may only allow users to call other people using the same service, but others may allow users to call anyone who has a telephone number - including local, long distance, mobile, and international numbers. Also, while some services only work over a computer or a special VoIP phone, other services allow you to use a traditional phone through an adaptor.

Advantages claimed by VoIP advocates over POTS (Plain Old Telephone Service) include:
Lower Cost

Convergence of data and voice applications

Support for telecommuting

Multimedia conferencing

Unified Messaging - email, voicemail, faxes, pages

Better support for call-centers

Facilitates Relocation
By reading snippets from on-line and print media one gets the impression that VoIP provides telephone calls for free. This was the major the major benefit touted by vendors and the impetus for early adoption when VoIP introduced in the 1990s. Like the proverbial free lunch, this is not exactly true. The calculation of ROI is more complicated. In order to determine cost savings one must first determine the amount of and cost of current telephony service usage by type (local and long distant, domestic and international, incoming and outgoing, faxes…) and then identify any new equipment and services and their associated cost required to support the desired level of VoIP. Some of the
potential savings may have been reduced because the telephone companies have dramatically reduced the cost of long distance and international calling with a myriad of plans over the last several years. Investment in existing equipment and contractual commitments to service providers are factors to be considered. Broadband Internet connectivity is a system requirement for VoIP.

For businesses possible operational cost reductions come from reduced technical and management staff for one converged network rather than for two networks, reduced number of external vendors, reduced add, move and change costs, single help desk, and simplified chargeback. Firms with a number of geographically disperse offices make excellent candidates for cost savings. The choice for VoIP is easier when opening new offices where there has been no prior investment in traditional telephony.

Vendors contend that VoIP not only saves money on long distance telephone calls but enhances productivity. With VoIP multimedia conferencing, web-enabled call centers, real-time collaboration and the like become economical. VoIP offers centralized provisioning, management, technical support, billing and chargeback for all voice and data systems.

More advanced features are offered through IP Centrex (Central EXchange) and IP PBX (Private Branch Exchange). IP Centrex allows customer to have and use features that are typically associated with a PBX without the purchase of PBX switching systems. These features include 3 or 4 digit dialing, intercom features, distinctive line ringing for inside and outside lines, voice mail waiting indication and others. IP PBX is a private local telephone system that uses a IP telephone server to provide for call processing functions and to control gateways access that allows the IPBX to communicate with the public switched telephone network and other IPBX's that are part of its network. IPBX
systems can provide advanced call processing features such as speed dialing, call transfer, and voice mail along with integrating computer telephony applications.

Given the fact that telephone networks were developed to support voice and Internet networks were developed principally to support text, we are not surprised that there are some aspects of Internet networks that make them less than ideally suited for voice communication.

The PSTN (Public Switched Telephone Network) is a “circuit-switched” network that operates by establishing dedicated connections, or “circuits,” between end points that remain open for the duration of each call and then shut down when the call ends. The charge for the call is generally a function of the duration of the call and the distance between the participants.

The basic process for VoIP is
Convert analog voice to digital format by a codec (coder/decoder) according to various encoding standards
Compress the digital data
Packetize the data
Signal or setup the call to locate and connect to destination
Reverse the process on the receiving end
The Internet is a packet-switching network. A packet is a block of user data together with necessary address and administration information attached, to allow the network to deliver the data to the correct destination. There is no persistent connection between the end points of a conversation. Each packet may take its own route through the web. Not all packets travelling between the same two end points, even those from a single message, will necessarily follow the same route. Consequently, packets may arrive at their destination out of sequence. They must be reassembled at the destination.

An Analog Telephone Adapter (ATA) is a device that adapts or converts standard analog telephone equipment for use with VoIP. An ATA will have one or more telephone jacks to connect to the analog phone stuff and an Ethernet jack for connection to a LAN or broadband modem. Alternatively soft phones require only a computer to run on and a suitable headset or integrated microphone and speaker.

Issues surrounding VoIP

There are some known problem areas associated with VoIP which are described briefly below. The quality of service has been an area of concern that has been improving.

Latency is the amount of time delay between the initiation of a service request for data transmission or when data is initially received for retransmission to the time when the data transmission service request is granted or when the retransmission of data begins. Large latency values do not necessarily degrade the sound quality of a phone call, but the result can be a lack of synchronization between the speakers, such that there are hesitations in the speaker' interactions. Two problems due to delay are echo and talkover. If you watch CNN during interviews with overseas correspondents you see the annoying impact of time delay.

Jitter is the variance of interpacket arrival times as introduced by the variable transmission delay over the network. Removing jitter requires collecting packets (play-out buffers) and holding them long enough to allow the slowest packets to arrive in time to be played in the correct sequence, which causes additional delay. This technique can not eliminate server jitter.

Packet Loss occurs when packets are dropped under peak loads and during periods of congestion (caused, for example, by link failures or inadequate capacity). Packet loss causes voice clipping and skips, often resulting in choppy and sometimes unintelligible speech. Packet loss for non-real-time applications, such as Web browsers and file transfers, is undesirable, but not critical because of their retransmission capabilities. Approaches used to compensate for packet loss include interpolation of speech by replaying the last packet and sending of redundant information. Minor loss is tolerable and may possibly pass unnoticed. The listener can always request the speaker to repeat but
this quickly becomes unacceptable if the loss is persistent.

Bandwidth Contention occurs when there are multiple users and/or multiple uses of the communication links. Administrators can assign a higher priority to voice communications which unlike text communication do not have loss detection and retransmission capabilities.

Voice quality is calculated by a measurement called MOS, or mean opinion score. This measurement is determined by an actual group of human listeners who rate the quality of audio on samples played to them. The MOS rating system is judged on a five-point scale where a score of 1 implies poor quality, and a score of 5 is excellent. Toll quality telephone service is generally granted a MOS score of 4.0.

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-- Jack Horgan, EDACafe.com Contributing Editor.

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