Voice Processing over Internet Protocol (VoIP): State of Art
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Voice over Internet Protocol (VoIP) allows data network and voice networks to converge. The bandwidth is saved in VoIP system by not sending the IP packets during silence periods in the conversation and this saved bandwidth can be used for other applications. VoIP system uses Internet Protocol (IP) which is a network layer protocol according to the seven-layer Open System Interconnected (OSI) model.VoIP involves digitization of voice streams and transmitting the digital voice as packets over the IP based packet network. The analog speech signal is compressed using various codecs such as G.711,...
Voice over Internet Protocol (VoIP) allows data network and voice networks to converge. The bandwidth is saved in VoIP system by not sending the IP packets during silence periods in the conversation and this saved bandwidth can be used for other applications. VoIP system uses Internet Protocol (IP) which is a network layer protocol according to the seven-layer Open System Interconnected (OSI) model.VoIP involves digitization of voice streams and transmitting the digital voice as packets over the IP based packet network. The analog speech signal is compressed using various codecs such as G.711, G.729 etc. The encoded speech is packetized and then speech packets are routed to the destination through the internet. At the receiver end, the whole process is reversed. The voice packets are depacketized and decoded subsequently. The decoding process may include de-jittering, error correction and packet loss concealment. The quality of the VoIP signal is influenced with various network factors such as packet delay, jitter and packet loss. Delay is the time taken by the voice to reach from talker's mouth to the listener's ear. The variation in the arrival time
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