he concept of an integrated services network with both real-time and data services is not new. In fact, two alternate schemes are currently contending to provide all the services seen in Fig. 1 . From one point of view, a new backbone from the telephony world -interexchange carrier (IXC)/asynchronous transfer mode (ATM) or frame relay (FR) -would provide all the required quality of service (QoS) levels for integrated services. From the other point of view, the existing Internet and corporate intranets would carry real-time voice traffic in addition to data. Traditionally, the networking world has been divided along such lines. There has long been a telecommunications network that is circuit-switched and designed for point-to-point communication of real-time audio. Subsequently it has been adapted for the growing needs of data communication via modem technologies, ISDN, digital carriers, and, most recently, integrated services ATM and FR backbones. In contrast, there also exists a data networking world of store-and-forward packet technologies created primarily for data transport over local and wide areas. These networks are the vast collection of small and large IP networks that are intertwined in the form of the Internet and many partitioned intranets. Data networks comprise links, routers, bridges, and switches in the form of local and wide area networks. Although these two networking worlds are coming together in a model of shared data, voice, and video, proponents of each view are looking at the future as an extension of their own technology. The telecommunications world has envisioned an integrated network via a large-scale ATM backbone that supports many levels of QoS, including traditional n x 64 kb/s voice. Since the telecommunications world has always been very QoS-focused, ATM is provisioned with mechanisms to provide different QoS levels. From the IP community, the long-term view is that real-time voice and video services can multiplex with existing data traffic. However, QoS has not been considered with the same intensity -the current Internet service model is flat, offering a classless, best-effort delivery service. As such, QoS is an ad hoc extension to the IP infrastructure. The next generation of IP, version 6, includes support for "flows" of packets between one or more hosts [1]. In conjunction with a hop-by-hop resource reservation protocol such as RSVP [2], end-to-end capacity can be set aside for real-time traffic. Much can be said of the trade-offs between IP and ATM solutions for providing integrated services. However, in this article we focus on the technical issues involved in supporting audio in the current Internet. The dominant standard for transmitting multimedia in packet-switched networks is International Telecommunication Union (ITU) Recommendation H.323 [3, 4] , which uses IP/UDP/RTP encapsulation for audio. RTP, the Real-Time Protocol [5] , is a generic mechanism for supporting the integration of voice, video, and data. RTP headers provide the sequence number and timestamp information needed to reassemble a real-time stream from packets. H.323 does not  We discuss the architecture and technical viability of transporting real-time voice over packet-switched networks such as the Internet. The value of integrating voice and data networks onto a common platform is well known. The telephony industry has proposed the ATM standard as a means of upgrading the Internet to provide both real-time and data services. In contrast, voice services may be added to traditional IP networks that were originally designed for data transmission alone. Here, we consider the feasibility and expected quality of service of audio applications over IP networks such as the Internet. In particular, we examine possible architectures for voice over IP and discuss measured Internet delay and loss characteristics. 12
For example, Kostas REF studied the feasibility of real-time voice over the Internet and discussed measured delay and loss characteristics.
Real-Time Voice Over Packet-Switched Networks
ICC (1)
Computer Science