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The following is quoted from www.stikeman.com/newslett/TelJun04.htm.

Two key differences between VoIP and traditional telephone technology are the network over which a call is transmitted and the transmission format. In traditional telephony, a call over the public switched telephone network (PSTN) generally results in the creation of an end-to-end circuit that establishes a physical connection between the caller and the called party through the wires, cables and switches of the telephone network. The circuit exists for the duration of the call until hang-up, and is typically dedicated to the voice signals flowing between the two parties participating in the call. In contrast, in a VoIP call there is no dedicated circuit. The contents of a VoIP call flow between the caller and the called party over various networks that comprise the Internet. This happens through a router or switch operated by the VoIP provider, which matches a telephone number with an IP address. Different portions of a VoIP call may be routed over different transmission paths, with the contents of the call moving over the Internet amidst other traffic, including other VoIP calls, e-mails, miscellaneous data and video traffic.

The call format is the second feature distinguishing VoIP from traditional telephony. Voice communications on a PSTN call begin as analog signals, or electronic sound waves, which are converted into digital format (i.e., a series of ones and zeroes). The digits are transmitted in sequence over the circuit and converted back into analog format to be heard by the other party on the call. The digits move across the circuit in sequence, corresponding to the voice signals on the call. Spoken words on a VoIP call are converted from analog into digitized Internet Protocol (IP) packets. A packet is analogous to an envelope with data on the inside and sequence coding, as well as originating and destination addresses, on the outside. The VoIP users telephone is connected to a router. These packets enter the Internet at the router and are sent individually, though not necessarily in sequence, over the Internet. At the router closest to their destination (i.e. the other party on the call), they are reassembled in original sequence and converted back into analog format for reception by the recipient. A key point associated with VoIP is the significantly lower transmission costs of this technology.



Feedback received on this FAQ entry:
  • Try out Ozeki VoIP SIP SDK. You can download and use trial version free of charge. Here is the link: http://voip-sip-sdk.com/p_21-download-ozeki-voip-sip-sdk-voip.html I would like to mention another page as well. It is useful to get more information on IVR systems: http://www.voip-sip-sdk.com/p_90-how-to-handle-the-speaker-and-the-microphone-in-c-net-when-building-a-softphone-voip.html I think it is worth reading those valuable webpages. Have a nice day!

    2012-09-12 05:32:32 (38934683 See Profile)



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by Styvas See Profile edited by canoe See Profile
last modified: 2005-06-16 22:38:12