Overview of VOIP Data Handling

"Before any voice can be sent, a call must be placed. In an ordinary phone system, this process involves dialing the digits of the called number, which are then processed by the telephone company's system to ring the called number. With VOIP, the user must enter the dialed number, but after that a complex series of packet exchanges must occur. The problem is that computer systems are addressed using their IP address, but the user enters an ordinary telephone number to place the call. The telephone number must be linked with an IP address to reach the called party, much as an alphabetic web address, such as www.nist.gov must be linked to the IP address of the NIST web server. A number of protocols are involved in determining the IP address that corresponds to the called party's telephone number.

"Once the called party answers, voice must be transmitted by converting the voice into digitized form, then segmenting the voice signal into a stream of packets. The first step in this process is converting analog voice signals to digital, using an analog-digital converter. Since digitized voice requires a large number of bits, a compression algorithm must be used to reduce the volume of data to be transmitted. Next, voice packets are inserted into data packets to be carried on the Internet. The protocol for the voice packets is typically the Real Time Protocol, RTP. RTP packets have special data fields that hold data needed to correctly re-assemble the packets into a voice signal on the other end. But voice packets will be carried as payload by UDP protocols that are also used for ordinary data transmission. In other words, the RTP packets are carried as data by the UDP packets, which can then be processed by ordinary network nodes throughout the Internet. At the other end, the process is reversed: the packets are disassembled and put into the proper order, digitized voice data extracted from the packets and uncompressed, then the digitized voice is processed by an digital-to-analog converter to render it into analog signals for the called party's handset speaker."