Multimedia networking is perhaps the most exciting development in the Internet today. People throughout the world are spending less time in front their radios and televisions and are instead turning to the Internet to receive audio and video emissions, both live and prerecorded. As high-speed access penetrates more residences, this trend will continue -- couch potatoes throughout the world will access their favorite video programs through the Internet rather then through the traditional microwave and satellite channels. In addition to audio and video distribution, the Internet is also being used to transport phone calls. In fact, over the next ten years the Internet mayl render the traditional circuit-switched telephone system obsolete in many countries. The Internet will not only provide phone service for less money, but will also provide numerous value-added services, such as video conferencing, online directory services, and voice messaging services.
In Section 6.1 we classified multimedia applications into three categories: streaming stored audio and video; one-to-many transmission of real-time audio and video; and real-time interactive audio and video. We emphasized that multimedia applications are delay sensitive and loss tolerant, which is very different from static-content applications, which are delay tolerant and loss intolerant. We also discussed some of the hurdles that today's best-effort Internet places before multimedia applications. We surveyed several proposals to overcome these hurdles, including simply improving the existing networking infrastructure (by adding more bandwidth, more network caches, and deploying multicast), adding functionality to the Internet so that applications can reserve end-to-end resources (and so that the network can honor these reservations), and finally introducing service classes to provide service differentiation.
In Sections 6.2-6.4 we examined architectures and mechanisms for multimedia networking in a best-effort network. In Section 6.2 we surveyed several architectures for streaming stored audio and video. We discussed user interaction -- such as pause/resume, repositioning, and visual fast forward -- and provided an introduction to RTSP, a protocol that provides client-server interaction to streaming applications. In Section 6.3 we examined how interactive real-time applications can be designed to run over a best effort network. We saw how a combination of client buffers, packet sequence numbers and timestamps can greatly alleviate the effects of network induced jitter. We also studied how forward error correction and packet interleaving can improve user perceived performance when a fraction of the packets are lost or are significantly delayed. In Section 6.4 we explored media chunk encapsulation, and we investigated in some detail one of the more important standards for media encapsulation, namely, RTP. We also looked at how RTP fits into the emerging H.323 architecture for interactive real-time conferencing
Sections 6.5-6.9 looked at how the Internet can evolve to provide guaranteed QoS to its applications. In Section 6.5 we identified several principles for providing QoS to multimedia applications. These principles include packet marking and classification, isolation of packet flows, efficient use of resources, and call admission. In Section 6.6 we surveyed a variety scheduling policies and policing mechanisms that can provide the foundation of a QoS networking architecture. The scheduling policies include priority scheduling, round-robin scheduling, and weighted-fair queuing. We then explored the leaky bucket as a policing mechanism, and showed how the leaky bucket and weighted-fair queuing can be combined to bound the maximum delay a packet experiences at the output queue of a router.
In Sections 6.7-6.9 we showed how these principles and mechanisms have led to the definitions of new standards for providing QoS in the Internet. The first class of these standards is the so-called intserv standard, which includes two services -- the guaranteed QoS service and the controlled load service. The guaranteed QoS service provides hard, mathematical provable guarantees on the delay of each of the individual packets in a flow. The control-load service does not provide any hard guarantees, but instead ensures that most of an application's packets will pass through a seemingly uncongested Internet. The intserv architecture requires a signaling protocol for reserving bandwidth and buffer resources within the network. In Section 6.8 we examined in some detail an Internet signaling protocol for reservations, namely, RSVP. We indicated that one of the drawbacks of RSVP (and hence the Intserv architecture) is the need for routers to maintain per-flow state, which may not scale. We concluded the chapter in Section 6.9 by outlining a recent and promising proposal for providing QoS in the Internet, namely, the diffserv architecture. The diffserv architecture does not require routers to maintain per-flow state; it instead classifies packets into a small number of aggregate classes, to which routers provide per-hop behavior. The diffserv architecture is still in its infancy, but because the architecture requires relatively minor changes to the existing Internet protocols and infrastructure, it could be deployed relatively quickly.
Now that we have finished our study of multimedia networking, it is time to move on to another exciting topic in networking, namely, network security. Recent advances in multimedia networking may displace the distribution of audio and video information to the Internet; as we shall see in the next chapter, recent advances in network security may displace the majority of economic transactions to the Internet.
Copyright 1996–2000. James F. Kurose and Keith W. Ross. All Rights Reserved.