_________________________________________________________ISSC 2002, Cork.
June 25-26
Adaptive Multicast Architecture For Streamed MPEG-4 over IP Networks Michael Searles, Nicola Cranley, Liam Murphy Department of Computer Science, University College Dublin, Belfield, Dublin4, Ireland.
[email protected],
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[email protected] __________________________________________________________________________________________ Abstract -- Currently multimedia is either downloaded before viewing, or streamed over a network. However, streaming real-time or near real-time applications with a specified Quality of Service (QoS) over best-effort IP networks is not yet a solved problem. The Real-time Transport Protocol (RTP) can be used to facilitate streaming, but also has the potential to support QoS. Multicast transmissions are currently quite static and inflexible as all users receive the same treatment and it is only the network behaviour which differentiates between their perceived QoS. By gathering network statistics during the session and defining different customer groups, we propose to adapt multicast multimedia streaming to a fluctuating network load and/or client requests, thereby providing adaptive QoS. _______________________________________________________________________________________
I. INTRODUCTION MPEG-4 is a new standard designed for video conferencing and other video applications such as interactive mobile multimedia, digital multimedia broadcasting, and interactive local multimedia. MPEG-4 flexibly decomposes a scene into objects, each with its own audio and video track that will vary over time. MPEG-4 content can be tailored (e.g. by layered encoding) so that if the network is congested or overloaded, not all objects need to be sent to the client (user), only those that make the video sequence coherent [1] and which match the client's needs. In computer and telecommunications networks, QoS refers to the network operators’ commitment to providing and maintaining acceptable values of parameters or characteristics of user applications in order to satisfy the users’ application requirements and expectations [2]. Providing QoS guarantees is difficult in networks that offer "best effort" service, such as those based on IP. The IP protocol makes no guarantees about when data will arrive, or the order of packet arrivals, or how much data it can deliver [3]. The Real-time Transport Protocol, RTP [4] and its control protocol, RTCP, are designed to support realtime applications, independent of the underlying transport and network. By itself, RTP does not ensure QoS guarantees are met, but simply facilitates QoS
support by adding extra functionality to UDP/IP such as payload identification, sequence numbers, timestamps, and delivery monitoring. RTCP is used to periodically convey statistics about the transmission to the client and server [5]. RTCP is particularly useful as it enables the integration of some QoS mechanisms by means of client feedback, allowing the server to modify its transmission in response to the clients’ requirements and network conditions.
II. SYSTEM MOTIVATION The main objective of this system is to provide adaptive QoS for users within a best effort network. In a best effort network the server cannot prevent packet loss or packet delays. The server should be able to adapt its transmission to these conditions. Typically, multicast transmissions are very inflexible; for example, users within a multicast group all receive the same treatment by the server in that a packet is broadcast to the group irrespective of their preferences or network connections. However, users should be allowed to have some influence over their connection status. If a customer is willing to pay for premium service, then the server should endeavor to satisfy this requirement. Premium customers should be provided with the premium QoS the server can provide. It should be possible for a user to
change their customer group if they are unhappy with the service they are being provided. This problem providing adaptive QoS is solved by using multicast groups representing each of the customer classes and adapting the transmission characteristics within each multicast group in response to the clients feedback of their QoS. III.
SYSTEM OVERVIEW
Our system is a basic client-server architecture. Both, client and server consist of the RTP/UDP/IP stack for packetisation and de-packetisation of the payload. An MPEG-4 codec encodes and decodes the payload to be streamed through the session. An IP network simulator is used to dynamically simulate the behaviour of an IP networks providing delay and loss characteristics. (Fig.1.)
Figure 1: System Overview. The server consists of a number of sub-servers. The request server accepts client requests and acts as an out of band messaging channel for the clients connected to the server. There are three other subservers, the economy sub-server, the standard subserver and the premium sub-server. (Fig.2.) Each of which corresponds to a particular customer group. The economy sub-server transmits a base quality encoding to customers and so on so forth. The server encodes a multimedia session into MPEG-4 with a number of quality levels. Each quality level encoding corresponds to a customer group. The system adapts each group as a result of the perceived QoS of the group. The server makes various adjustments to the transmission state of each group to reflect current network conditions and group preferences, without requiring continual user involvement. A typical user has little interest in parameters such as the number of lost packets or the round trip delay provided that the overall perceived quality of the transmission is acceptable to them [6].
Figure 2: Sub-Server Design.
IV. MPEG-4 QUALITY LEVELS MPEG-4 is a standardised method of achieving high compression ratios for video content. High compression ratios are desirable so that when transmitted MPEG-4 encoded data, its requires much less bandwidth. Codecs designed for the Internet require greater scalability, lower computational complexity, and greater resiliency to network losses and lower encode/decode latency for real-time transmission such as broadcasting. Codecs must be tightly connected to the network delivery software to achieve the greatest quality. MPEG-4 codecs are designed to be scalable, flexible providing high quality low bandwidth multimedia. Certain codecs support spatial, temporal and other scalable video solutions. MPEG-4 content can therefore be significantly tailored to suit particular network conditions and in the case of this system, customer groups. The Sorenson MPEG-4 codec is being used by the system to encode multimedia content to MPEG-4 at varying quality levels [7]. The values quoted are approximations as the true value depends upon the type of content being encoded. These quality levels have the following encoding characteristics:
Frame Size Frame Rate Data Rate
ECONOMY
STANDARD
PREMIUM
QCIF
QCIF/CIF
QCIF/CIF
4-10
10-15
15-30
