Jitter
Jitter is the measure of time between when a packet is expected to arrive to when it actually arrives. In other words, with a constant packet transmission rate of every 20 ms, every packet would be expected to arrive at the destination exactly every 20 ms. This situation is not always the case. For example, the figure below shows packet one (P1) and packet three (P3) arriving when expected, but packet two (P2) arriving 12 ms later than expected and packet four (P4) arriving 5 ms late.
The greatest culprit of jitter is queuing variations caused by dynamic changes in network traffic loads. Another cause is packets that might sometimes take a different equal-cost link that is not physically (or electrically) the same length as the other links. Media gateways have play-out buffers that buffer a packet stream so that the reconstructed voice waveform is not affected by packet jitter. Play-out buffers can minimize the effects of jitter, but cannot eliminate severe jitter. Although some amount of jitter is to be expected, severe jitter can cause voice quality issues because the media gateway might discard packets arriving out of order. In this condition, the media gateway could starve its play-out buffer and cause gaps in the reconstructed waveform.
Real-time applications based on the UDP are significantly less tolerant to packet loss. UDP does not have retransmission facilities; however, retransmissions would almost never help. In an RTP session, by the time a media gateway could receive a retransmission, it would no longer be relative to the reconstructed voice waveform; that part of the waveform in the retransmitted packet would arrive too late. It is important that bearer and signaling packets are not discarded, otherwise, voice quality or service disruptions might occur. In such instances, CoS mechanisms become very important. By configuring CoS parameters, you can give packets of greater importance a higher priority in the network, thus ensuring packet delivery for critical applications, even during times of network congestion. Although packet loss of any kind is undesirable, some loss can be tolerated. Some amount of packet loss for voice services could be acceptable as long as the loss is spread out over a large amount of users. As long as the amount of packet loss is less than five percent for the total number of calls, the quality generally is not adversely affected. It is best to drop a packet, versus increasing the latency of all delivered packets by further buffering them.
The greatest culprit of jitter is queuing variations caused by dynamic changes in network traffic loads. Another cause is packets that might sometimes take a different equal-cost link that is not physically (or electrically) the same length as the other links. Media gateways have play-out buffers that buffer a packet stream so that the reconstructed voice waveform is not affected by packet jitter. Play-out buffers can minimize the effects of jitter, but cannot eliminate severe jitter. Although some amount of jitter is to be expected, severe jitter can cause voice quality issues because the media gateway might discard packets arriving out of order. In this condition, the media gateway could starve its play-out buffer and cause gaps in the reconstructed waveform.
Packet Loss
Packet loss occurs for many reasons, and in some cases, is unavoidable. Often the amount of traffic a network is going to transport is underestimated. During network congestion, routers and switches can over flow their queue buffers and be forced to discard packets. Packet loss for non-real-time applications, such as Web browsers and file transfers, are undesirable, but not critical. The protocols used by non-real-time applications, usually TCP, are tolerant to some amount of packet loss because of their retransmission capabilities.Real-time applications based on the UDP are significantly less tolerant to packet loss. UDP does not have retransmission facilities; however, retransmissions would almost never help. In an RTP session, by the time a media gateway could receive a retransmission, it would no longer be relative to the reconstructed voice waveform; that part of the waveform in the retransmitted packet would arrive too late. It is important that bearer and signaling packets are not discarded, otherwise, voice quality or service disruptions might occur. In such instances, CoS mechanisms become very important. By configuring CoS parameters, you can give packets of greater importance a higher priority in the network, thus ensuring packet delivery for critical applications, even during times of network congestion. Although packet loss of any kind is undesirable, some loss can be tolerated. Some amount of packet loss for voice services could be acceptable as long as the loss is spread out over a large amount of users. As long as the amount of packet loss is less than five percent for the total number of calls, the quality generally is not adversely affected. It is best to drop a packet, versus increasing the latency of all delivered packets by further buffering them.
