An architecture framework for measuring and evaluating packet-switched voice

Until a recent date all telephony connections are set up via circuit switching. Advances in networking technology have made it possible for the Internet evolves into a Broadband convergence Network (BcN) and provides various services including Internet Protocol (IP) telephony over high-speed IP networks. Voice-over-IP (VoIP) uses packetized transmission of speech over the Internet. In order for the Internet to realize a profit as traditional Public Switched Telephone Network (PSTN), it must provide high quality VoIP services. The VoIP metrics report block of Real-Time Transport Protocol Control Protocol Extended Reports (RTCP XR) can be applied to any one-to-one or one-to-many voice application for which the use of RTP and RTCP is specified. However, RTCP XR only defines packet type to convey information that supplements the six statistics that are contained in the report blocks used by RTCP's Sender Report (SR) and Receiver Report (RR) packets. Our objective in this paper is to describes a practical measuring framework for end-to-end QoS of packet switched voice in an IP environment including Packet Loss Concealment (PLC) techniques. It includes concepts as well as step-by-step procedures for setting up components, creating session, measuring packetized voice streams over IP networks.