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Voice Over IP/VoIP

Voice Over IP/VoIP

Companies have been using traditional PBX systems requiring separate networks for voice and data communications for the past 5 decades. With the new VoIP technology’s disruptive revolution businesses are rapidly moving on to VoIP PBX systems that manage both data and voice thereby delivering a huge advantage by converging voice and data.

Voice over Internet Protocol (VoIP) is a methodology and group of technologies that are used to deliver voice communications and media sessions over Internet Protocol (IP) networks, the biggest of which is the Internet.

The steps and principles involved in originating VoIP telephone calls are similar to traditional digital telephony and involve signaling, channel setup, digitization of the analog voice signals, and encoding. The difference is when it comes to transport, instead of being transmitted over a circuit-switched network, however, the digital information is packetized, and transmission occurs as Internet Protocol (IP) packets over a packet-switched network. With VoIP, voice data is digitally encoded using µ-law or A-law Pulse Code Modulation (PCM). The voice data can then be compressed if necessary and sent over the network in User Datagram Protocol (UDP) packets. Standard TDM telephony sends voice data at a low constant data rate. With VoIP, relatively small packets are sent at a constant rate. The total overall rate of sending data is the same for each kind of telephony. The advantage of VoIP is that one high-speed network can carry the packets for many voice channels and possibly share with other types of data at the same time (for example, FTP, HTTP, and data sockets). A single high-speed network is much easier to set up and maintain than a large number of circuit switched connections (for example, T1 circuits). The User Datagram Protocol (UDP) is used to transmit voice data over a VoIP network. UDP is a ‘send and forget’ protocol with no requirement for the transmitter to retain sent packets should there be a transmission or reception error. Real Time Transport (RTP) is used with VoIP to provide a method of handling disordered and missing packets and makes the best possible attempt to recreate the original voice data stream, comfort noise is intelligently substituted for missing packets.