The voice quality of VOIP or Voice Over Internet Protocol highly depends on the quality of the Internet service of the parties on both ends of the VOIP call. The common scenario for VOIP calls is that the VOIP service provider acts as a registrar allowing the caller and the called party to recognize each other and connect the call. Once the call is connected, the voice signal is converted into data by the VOIP devices or software and then sent to the other party as real-time packets or RTP usually through the UDP or user datagram protocol. The caller and the called party both send and receive voice data from each other and keep in contact with the VOIP service provider to monitor if the call should still continue. The VOIP service provider basically controls the call but the call itself has become peer to peer or P2P.

UDP is a one-way protocol in a sense that the sending party doesn’t verify if the receiving party actually received the data. This is in contrast to the TCP or transport control protocol which includes a method to verify if the receiving party actually received the data and would retransmit if the data wasn’t received. VOIP uses UDP in a P2P transaction because voice data is tolerant to some data transmission loss. This means that in an actual call, the voice data may not be completely received but the voice signal is still understandable. Why not use TCP for VOIP traffic to ensure no data loss? While TCP ensures complete data transmission, it doesn’t ensure real-time transmission which would mean excessive delays in the VOIP traffic leading to echo and delayed voice signal. So it is a compromise between data completeness and real-time transmission and for VOIP, real-time transmission is a higher priority to make the VOIP call understandable. For other Internet activity like browsing, email and data download, data completeness is a must but real-time transmission is not, so TCP is used.

So now we know that VOIP is data loss tolerant which of course means, the lesser the data loss, the better the voice quality. What causes data loss? There are two most important factors that contribute to data loss in an UDP data transmission. First, of course, is the Internet connection bandwidth which is usually not guaranteed in residential type connections. During peak hours, actual bandwidth that becomes available for a VOIP call would be limited resulting to some of the voice data to be discarded: there is simply not enough bandwidth for it to get through. If the network is extremely busy, sometimes VOIP calls would not be possible at all. The second factor is the latency which is the time it takes for a data packet to travel from one end of the VOIP call to the other end. This should be around 300ms and can be checked using the ping command. If the latency is much higher than 300ms, there will be extensive delay before the listener can hear the speaker. Also, very high latencies cause many of the packets to be discarded significantly deterioriating the VOIP call.

In summary, available bandwidth and low and consistent latency will help ensure a clear VOIP call and these factors are highly dependent on the quality of service that the Internet service provider can deliver.