Voice over IP – A Simple Explanation
We’ll use the example of a VoIP business phone system. With a VoIP phone system, the phones are essentially computing devices that are plugged into your local network. Some companies plug these “IP Phones” into a separate network specifically used for VoIP; others connect the IP phones directly to the network that the rest of their computers are connected to.
When a user talks into the microphone of an IP phone, their voice gets convert to electrical signals called sine waves. These sine waves then get sampled. The sampling process essentially converts the sine wave to a binary form (1’s and 0’s) that represents the characteristics of the original voice (the highs, lows, frequency, etc). Then these 1’s and 0’s get “encapsulated” or put into Internet Protocol (IP) packets. IP is the industry standard for sending ALL forms of data across networks (web, email, video, voice, etc). These IP packets then traverse the network to the end destination. The end destination could be an IP phone sitting right next to the original phone, an IP phones on the other side of the world, or perhaps a phone line that connects the phone system to the outside world. Once the IP packets carrying this voice information reaches its destination, the 1’s and 0’s get converted back to the analog sine wave, the analog sine wave get’s amplified and then ran through a speaker in the receiving phone that recreates the audio speech of the original speaker.
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