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Although this is a Cisco networks dedicated blog, I decided to start a series of tutorial posts about a general technology which is not directly related to Cisco but it is a field in which Cisco is again a major player. This is IP Telephony and Voice over IP (VoIP). The two terms, IP Telephony and VoIP, are related around the same concept but in my opinion they are not exactly the same thing. Many people refer to these two terms interchangeably but they are not exactly the same. So, before moving on lets clarify the difference between IP Telephony and VoIP. The technology of speech encoders / decoders in the past few years has made considerable progress thanks to advances in the field of telephony hardware architecture which is built on specialized digital signal processing chips (digital signal processors, DSP), as well as research on human speech. The new codecs are not just doing analog to digital conversion - they use sophisticated predictive models to analyze the input speech signal and subsequent transmission of speech using minimal bandwidth. A simple pulse-code modulation of speech PCM (standard ITU-T G.711) enables the transformation of speech at 64 kbit / sec of the mu-law and A-law. In both of these methods we can achieve 12-13 bits PCM quality in 8 bits using a logarithmic compression. Another commonly used method of compression is adaptive differential pulse-code modulation (ADPCM). VoIP can also be used by legacy TDM based PBX systems to transport voice calls over an IP WAN network or even over the Internet. Special voice gateways are used to connect to the legacy PBX telephone system on one end and to the IP network on the other end in order to translate the TDM voice stream into IP voice packets. So to summarize, IP Telephony is the overall concept of the modern form of voice communication which harnesses the power and features of VoIP technology in order to offer the overall experience of communicating effectively and with lots of extra features. The term VoIP or Voice over IP refers to the transfer of voice packets over networks based on Internet technology and, more specifically, the IP Protocol. The IP protocol on which the whole Internet is based on was created to implement the transmission of data in the form of data packets. This means that when a data document is transferred over the Internet is cut into small IP packets and sent over the network. When the document reaches its destination, the packets are joined again thus recreating the original document. The same logic applies if the data transferred corresponds to a voice conversation. The voice is digitized, chopped into packets of data transferred over the network via the IP protocol. At the destination the packets are rejoined to recreate the voice stream. Here we should make clear that VoIP refers to the transfer of voice over any IP network. Such a network is the Internet of course, but when considering VoIP it does not necessarily mean that we carry voice over the Internet only. It can be any IP-based network (such as a private corporate WAN network). Requirements for quality in voice networks determine the maximum latency of 150-200 ms. A greater delay value usually reduces the quality of conversation significantly. The greatest amount of delay (30 milliseconds) is introduced by the G.723 codec, and the lowest delay (0.75 ms) is found in G.711 codec. It should be noted that the smallest propagation delay time is introduced by channel switching networks, and the greatest propagation delay is found in packet-switched networks (IP networks) due to buffering. In connection with this fact, the VoIP technology is less attractive for voice transmission over the Internet, than VoATM and VoFR. Nevertheless, VoIP quality over the Internet is quite acceptable for a corporate network that needs maximum of 4 -6 concurrent voice channels.
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