The transmission of Voice over Internet Protocol (VoIP) requires signal processing for reliable and efficient performance over the computer and the communication platform. Speech quality, as perceived by the users of VoIP telephony, is critically important to update the service. Signal quality is degraded by network layer problems, ,including delay, jitter and packet loss. This paper presents a simulation model for VoIP and a simulative analysis of VoIP. The digital filtering algorithm is implemented on degraded VoIP speech signal.

The traditional phone systems that are based on circuit switching have been used for over 100 years. In this system, there is a connection between the caller and callee, and this characteristic gives the system namethe circuit. After the era of circuit switching, analog communication has moved towards digital communication. In digital communication, the analog signal is communicated by digital means. At the transmitting end, the signal enters the network via a traditional phone set, is then encoded into digital form and then transmitted over the network. At the receiving end, the process is reversed. This transmission is ISDN (Integrated Services Data Network) (Danny, 1977; Fred, 2001; Joe, 2004; Global IP Sound, 2004; Ajay, 2006; and dissertations.port.ac.uk/124/01/WheelerM.pdf). But this technique has some inefficiencies; for example, during a call sometimes there are unneeded silences, which waste almost half of the space provided. To deal with these inefficiencies, the idea of packet switching, which is the base for Voice over Internet Protocol (VoIP), has been introduced. VoIP refers to technologies that allow telephone-like voice communications carried within Internet Protocol (IP) data packets, rather than dedicated voice signal lines. Audio is digitized, compressed and filled into multiple IP packets, and then these packets are transmitted over User Datagram Protocol (UDP). At the receiver end, these packets are received by UDP receive and decompressed, unpacketized and converted back into analog voice signal, and replayed using suitable audio device (Fred, 2001; Joe, 2004; and dissertations.port.ac.uk/124/01/WheelerM.pdf).

The transmission of VoIP requires signal processing for reliable and efficient performance over the computer and the communication platform. Communication between two computers in the same room is called point-to-point communication. However, if the computers are located in different parts of the country, public carrier facilities must be used. Normally, this involves the Public Switched Telephone Network (PSTN), which requires a device called modem for transmitting data.

Electrical and Electronics Engineering Journal, VoIP Signal Processing, Digital Filtering Algorithm, Traditional Phone Systems, Digital Communication, Integrated Services Data Network, Internet Protocol Data Packets, User Datagram Protocol, Public Switched Telephone Network, Digital Signal Processing Algorithms.