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Voice Over IP (VoIP) Explained


Voice over IP, or VoIP had become a buzzword in the past few years because it represents a more cost effective model for transmitting voice conversations than the old circuit switched networks. The existing telephone infrastructure consists of physical wires connecting circuit switches in which one telephone caller is connected directly to another through a switched network. This of the old switchboard operators in days of old, automated on a large scale.

The existing Internet infrastructure is far different than the circuit switched networks that carry most voice calls. The Internet carries packets of digital information data. These packets are switched and routed through the Internet from one destination to another.

The protocol that governs the Internet is called TCP/IP. It was born out of UNIX and became the de facto standard of Internet communications. Because of the ubiquitous nature of TCP/IP, it represents the obvious choice for use in digital voice communications. Since it using IP – the Internet Protocol, voice over IP is generally referred to as VoIP.

In the Internet world, pieces of data called IP packets are passed around. A good analogy for this is the post office. Each packet contains its destination, and the routers and switchers in the network forward the packets like sorters in the post office. A package at the post office will typically go from one postal sorting center to another, before arriving at the destination post office to be put on the appropriate mail trucks. Packets move around the Internet in the same way.

In VoIP, special receivers known as codecs compress and decompress digital data into the audio we here through a telephone handset. When you speak into a VoIP phone, the phone compresses your voice into digital data, which is then sent out over an IP network as a series of packets. The receiving end receives those packets, and reforms them into audio through the handset of the person you are speaking to.

In order for VoIP to work successfully, standards are necessary so that one phone can talk to another. The standard protocol used in VoIP today is SIP, or Session Initiation Protocol. This protocol contains a number of compression and communications standards and algorithms that VoIP phones must support. For years, SIP was in a battle with proprietary protocols like Cisco Skinny, and other standards like H.323 which is the dominant standard in IP videoconferencing. But ultimately SIP has prevailed.

Because the nature of VoIP is different than circuit switched networks, VoIP comes with a new set of issues. The most serious concern is latency. Latency is the amount of time it takes between when you say something, and when it is heard on the other end. If the network is too slow or busy, and the packets don't arrive on time or in order, the conversation will fall apart. Studies show that people find latencies exceeding .25 seconds to be too frustrating to use. Because of this, quality of service (QoS) is an essential portion of a VoIP network, as it guarantees that packets will be delivered with minimal interruption.

Because of the cost advantages of VoIP, it will be commonplace before too long. Don't be surprised when old phones go the way of vinyl records.



About the Author:

Rex Ryan is a telecommunications engineer and runs a website on voip technologies.

Source: www.isnare.com