Transmission of Voice Using IP Networks

Here is how a VoIP transmission is completed:

Step 1:  Because all transmissions must be digital, the caller’s voice is digitized. This can be done by the telephone company (which is how carriers use IP in their networks), by an Internet service provider (ISP), or by a PC on your desk.

Step 2: Next using complex algorithms the digital voice is compressed and then separated into packets; and using the Internet protocol, the packets are addressed and sent across the network to be reassembled in the proper order at the destination.  Again, this reassembly can be done by a carrier, and ISP, or by one’s PC.

Step 3: During transmission on the Internet, packets may be lost or delayed, or errors may damage the packets. Conventional error correction techniques would request retransmission of unusable or lost packets, but if the transmission is a real-time voice communication that technique obviously would not work, so sophisticated error detection and correction systems are used to create sound to fill in the gaps.  (This process stores a portion of the incoming speaker’s voice, and uses a complex algorithm to “guess” the contents of the missing packets and create new sound information to enhance the communication.)

Step 4: After the packets are transmitted and arrive at the destination, the transmission is assembled and decompressed to restore the data to an approximation of the original form.

As this explanation suggests, technology that works fine for sending data may be less than perfect for voice transmissions.  The technology is improving, but still the quality of a voice transmission using packet technology is inferior to a circuit-switched connection, and that difference in quality would normally be obvious to any listener.  As IP technology improves, the quality advantage for voice communication enjoyed by the circuit-switched will decrease, but most experts see parity in quality as still a distant prospect.