Voice Over IP Fundamentals

VoIP carries calls over the Internet.

Voice over IP (VoIP) captures a sound wave, digitizes it and cuts it into segments to be sent over the Internet. A VoIP program uses pre-existing network technology to create an interactive telephone conversation between two or more endpoints.

Data Capture

Sound is captured in segments and converted into a digital signal. The size of each segment is called the sample size. The program that performs this task is called a codec. A codec follows one of several protocols. Each codec has its defined sample interval, which is the time period during which sound is grabbed. The sample interval determines the sample size.

Session Management

The calling process emulates standard telephones. The address of a remote device is entered into the VoIP user interface. The receiving user is notified that a connection is required. When the receiver signals acceptance, a connection is established. At the end of the conversation, the users hang up and the connection is broken. These session management processes in VoIP are defined in a standard called session initiation protocol (SIP).

Data Transfer

The application sends its digitized data segments to communication software for transmission. The programs put the data in packets and address them. These functions follow user datagram protocol (UDP), real-time transport protocol (RTP) and Internet protocol (IP) standards.

References

• Cisco Systems: Voice Over IP---Per Call Bandwidth Consumption
• Vonage: How VoIP Works