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In the beginning, when the Internet Protocol was first designed, no one was thinking about the possibilities of sending audio and video. Real time communication was not an issue.
Perhaps the greatest single problem is that the Internet Protocol (the IP part of VoIP - Voice over Internet Protocol) wasn't designed to ensure that the packets are delivered in the correct order. When information is transmitted using IP, the data is broken up into information packets, each of which is sent separately. The correct sequence of packets is part of the information in each packet, but nothing specifically exists to make sure that the packets are delivered, and, therefore, received in the proper order.
Now this isn't usually a significant issue for web pages, email, etc. Why? Because these aren't real-time applications. Audio and video however, especially live audio and live video are definitely real-time applications. For a real time conversation to work, the packets have to arrive - pretty much in order and also within certain time limits.
The first, and one of the major challenges then, is to restructure incoming packets into the correct order and to somehow cope with lost and/or trashed packets. Face it, the internet does not provide a quality of service guarantee. If enough packets are lost, an audio or video stream rapidly turns into a useless mess. While packets can be resent - the standard way lost/trashed packets are dealt with - real time communication means that you just can't wait around forever. After a certain time, it's simply too late to maintain a coherent stream.
What we've seen over the last few years is a gradual and now nearly explosive growth in the use of VoIP, and streaming audio and video. The reason is decline of dial-up and the growth of ISDN, DSL, ADSL, cable and other high speed, high bandwidth access modes. Bandwidth is the answer to most of the problems posed by IP. End-to-end high speed links can ensure high quality sound. The sole remaining problem is latency.
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