That covers VoIP basics, but what about the more advanced options, and why is VoIP able to offer more advanced features where a regular phone can/t? Again, the secret is software. A VoIP system, whether home or business, can access a much richer software layer than a standard line from the plain old telephone service (POTS). On the business side, this flexibility has extended to integrating VoIP with other forms of communication to such a degree they all become a single platform, generally called Unified Communications as a Service (UCaaS). You won't anything that sophisticated when you're shopping for residential service, however.
The RTCP extended report VoIP metrics block specified by RFC 3611 is generated by an IP phone or gateway during a live call and contains information on packet loss rate, packet discard rate (because of jitter), packet loss/discard burst metrics (burst length/density, gap length/density), network delay, end system delay, signal/noise/echo level, mean opinion scores (MOS) and R factors and configuration information related to the jitter buffer. VoIP metrics reports are exchanged between IP endpoints on an occasional basis during a call, and an end of call message sent via SIP RTCP summary report or one of the other signaling protocol extensions. VoIP metrics reports are intended to support real-time feedback related to QoS problems, the exchange of information between the endpoints for improved call quality calculation and a variety of other applications.
On the early ARPANET, real-time voice communication was not possible with uncompressed pulse-code modulation (PCM) digital speech packets, which had a bit rate of 64 kbps, much greater than the 2.4 kbps bandwidth of early modems. The solution to this problem was linear predictive coding (LPC), a speech coding data compression algorithm that was first proposed by Fumitada Itakura of Nagoya University and Shuzo Saito of Nippon Telegraph and Telephone (NTT) in 1966. LPC was capable of speech compression down to 2.4 kbps, leading to the first successful real-time conversation over ARPANET in 1974, between Culler-Harrison Incorporated in Goleta, California, and MIT Lincoln Laboratory in Lexington, Massachusetts.[71] LPC has since been the most widely used speech coding method.[72] Code-excited linear prediction (CELP), a type of LPC algorithm, was developed by Manfred R. Schroeder and Bishnu S. Atal in 1985.[73] LPC algorithms remain an audio coding standard in modern VoIP technology.[71]
The only additional piece of equipment that you need is an Analog Telephone Adapter (also referred to as an ATA) that allows you to connect your existing telephone to your home Internet. This equipment is typically provided on a free lease basis from the home VoIP provider that you choose, or you can use you own device if you prefer. You can also use IP phone(s) instead of using the ATA with your existing analog phones. The sound quality is better but there is more up front cost as IP phones are more expensive than the ATA devices.
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