While a traditional circuit-switched voice call has a dedicated circuit and bandwidth allocated to it, packet-switched voice calls are subject to two main issues that impact the perceived quality of a call, namely delay and jitter. Delay comes in various forms, impacted by everything from the speed at which a voice packet is created using various codecs to the amount of time that it takes to propagate a signal along a path between a sending and destination node. Of course, a variety of other factors, including congestion, can add to the overall delay of a packet. Recall that in order for a voice call to proceed smoothly, the end-to-end delay should not exceed 150 ms. Later in this section you’ll learn more about the two main types of delay that can impact packet-switched voice connections, namely constant delays and variable delays. Both need to be considered in order to understand how voice traffic is impacted when traversing a packet-switched network.
Although the overall delay impacts the quality of a voice call, another key consideration is the difference between when packets are expected to arrive and when they actually arrive – a concept known as “jitter”. While it may not make a big difference if traditional data packets are received with timing variations between packets, it can serious impact the quality of a voice conversation, where timing is everything. In order to compensate for the fact that voice packets can be received with variable rather than constant timing, VoIP endpoints implement what is known as a “dejitter buffer” in order to change the variable delay back to the expected constant delay expected.