Packet networks break voice, fax, and data into small samples or packets of information. Each packet has a header that identifies where the packet is going and provides information on reconstruction when the packet arrives. Packets travel independently and they can travel by different routes during a single call. Because of congestion on the packet network or failure of network processing nodes in the packet network, packets can be lost. That is, during periods of congestion, queues in network routers begin to overflow and routers are forced to drop packets. Quality of Service (QoS) allows network routers to decide which packets to drop when the queues fill up.
QoS refers to the capability of a network to provide better service to selected network traffic over various technologies, including Frame Relay, Asynchronous Transfer Mode (ATM), Ethernet and 802.1 networks, and IP-routed networks that may use any or all of these underlying technologies. The primary goal of QoS is to provide priority including dedicated bandwidth, controlled jitter and latency, and improved loss characteristics. Thus, QoS enables networks to provide better service to certain flows by either raising the priority of a flow or limiting the priority of another flow. A flow may refer to a combination of source and destination addresses, source and destination ports, and protocol. A flow may be defined more broadly as any packet from a certain application or from an incoming interface.
Compared to voice traffic, certain types of data are very sensitive to packet loss and congestion within the network. These include fax tones and modem tones used for signaling between fax machines or modems, Dual Tone Multi-Frequency (DTMF) tones, and text relay tones. In low bandwidth networks, these tones can get lost or corrupted and lead to failed modem and fax calls or missing or corrupted DTMF digits. Although much more sensitive to packet loss, packets containing these signaling tones are conventionally put in the same category as voice packets and get the same quality of service. Redundant packets can be sent to improve reliability when the probability of packet loss is high, enabling the receiving side to reconstruct the missing packets. Currently, a redundancy factor can be set in voice gateways and redundant packets are retransmitted the number of times specified by the redundancy factor. However, since the probability of packet loss is high during periods of congestion, transmitting redundant packets during periods of congestion may actually contribute to more congestion. Thus, there is a need for reducing the value of the redundancy factor while continuing to mitigate problems due to packet loss.