telecommunications industry spans over 100 years, and Asterisk integrates most,if not all of the major technologies that it has made use of over the last century. To make the most out of Asterisk, you need not be a professional in all areas, but understanding the differences between the various codecs and protocols will give you a greater appreciation and understanding of the system as a whole.
This article explains Voice over IP and what makes VoIP networks different from the traditional circuit-switched voice networks that were the topic of the last chapter. We will explore the need for VoIP protocols, outlining the history and potential future of each. We'll also look at security considerations and these protocols' abilities to work within topologies such as Network Address Translation (NAT). The following VoIP protocols will be discussed:
Codecs are the means by which analog voice can be converted to a digital signal and carried across the Internet. Bandwidth at any location is finite, and the number of simultaneous conversations any particular connection can carry is directly related to the type of codec implemented. In this chapter, we'll also explore the differences between the following codecs in regards to bandwidth requirements (compression level) and quality:
We will then conclude the chapter with a discussion of how voice traffic can be routed reliably, what causes echo and how to minimize it, and how Asterisk controls the authentication of inbound and outbound calls.
The need for VOIP
The essential reason of VoIP is the packetization of audio streams for transport over Internet Protocol-based systems. The difficulties to achieving this identify with the way where people communicate. Not exclusively should the sign land in basically a similar structure that it was transmitted in, yet it needs to do as such in under 300 milliseconds. In the event that packets are lost or deferred, there will be corruption in the quality of the communications experience.
This word hasn't exactly made it into the lexicon, yet it is a term that is turning out to be increasingly normal. It alludes to the way toward cleaving a constant flow of data into tactful pieces (or packets), appropriate for conveyance freely of each other.
The vehicle conventions that on the whole are called "the Internet" were not initially structured in light of ongoing streaming of media. Endpoints were required to determine missing packets by standing by longer for them to show up, mentioning retransmission, or, sometimes, believing the data to be away for acceptable and basically continuing without it. In an ordinary voice discussion, these systems won't serve. Our discussions don't adjust well to the loss of letters or words, nor to any apparent postponement among transmittal and receipt.
The customary PSTN was planned specifically with the end goal of voice transmission, and it is superbly fit to the undertaking from a specialized stance. From an adaptability angle, be that as it may, its blemishes are evident to try and individuals with a constrained comprehension of the innovation. VoIP holds the guarantee of consolidating voice communications into the various conventions we carry on our systems, however because of the unique requests of a voice discussion, extraordinary aptitudes are expected to configuration, assemble, and keep up these systems.
The issue with bundle based voice transmission comes from the way that the manner by which we talk is absolutely contrary with the manner by which IP transports information. Talking and listening comprise of the handing-off of a stream of audio, though the Internet conventions are intended to cleave everything up, embody the bits of data into a large number of bundles, and afterward convey each bundle in the path conceivable to the far end. Obviously, a scaffold was required.