A. Field of the Invention
The present invention relates to an adaptation mechanism which can be used to monitor, maintain and control quality of voice-grade for communications among end-systems in a distributed PBX topology, thereby providing an enhanced Quality of Service (QoS) for the network.
B. Description of the Related Art
Recently, many efforts have been devoted by the Internet community to investigate transport mechanisms capable of guaranteeing Quality of Service (QoS) requirements for datagram networks (such as, for example, Internet Protocol (IP) networks). The objectives include alternatives to the transport of voice, video and multimedia by classic Telephone/ISDN and ATM networks. The basic problem is how to guarantee bandwidth, latency (delay) and packet loss, required by voice and video, in datagram network architecture.
Communication links between PBXs require a fairly large bandwidth for operation on data networks. Maintaining this amount of bandwidth is expensive and often results in degradation in the overall quality of service among all applications running on the network. Known solutions to the problem include attempting to reserve bandwidth using approaches such as Resource reSerVation Protocols (RSVP) or policy management systems.
In addition, approaches to maintaining voice quality, such as setting up a fixed Packet Delay Variation (PDV) buffer size is not ideally suited to Internet Protocol (IP) Networks in which messages flow in a bursty, rather than uniform, manner. If the buffer size is too small, the most recently arriving data will overflow and the preceding data is lost. If the buffer is too large, there will be gaps, resulting in breaks between message packets. Attempts to set the buffer size at periodic intervals is a tedious process due to the redundancy of the traffic flow, and is usually based on trial and error.
Until now, the main obstacle to implementation of an adaptable QoS monitoring mechanism from an application prospect has been the manner in which measurements are obtained. Several algorithm-based solutions have been proposed, including using the Internet Control Message Protocol (ICMP). This, however, has been shown to produce not very accurate measurements, has real-time impact on performance of the system, and adds more traffic to the network. The added traffic overhead is directly proportional to the level of accuracy being sought for the measurements.
Moreover, many real-time operating systems (RTOS) do not support multiple applications which use raw sockets (used mainly for ICMP). This usually results in interference between applications attempting to use the sockets.
Nor is using the Transmission Control Protocol (TCP) or the User Datagram Protocol (UDP) always acceptable, since these protocols add overhead onto both end systems, represented by the need to create at least one process on each end machine to simulate the functionality of the ICMP. Furthermore, processing time is added on because raw sockets interact directly with the network layer and other types of sockets interact with the upper layers of the IP stacks. This in turn results in inaccuracy in the sending and arrival time of messages. Furthermore, as in the case of ICMP, traffic is added to the network.
USPTO Backgrounds
USPTO Backgrounds