In recent years there has been a rapid increase in the deployment of multimedia streaming applications such as audio and video broadcasting. However, since the best-effort Internet is an unreliable network with a high packet loss rate and nonuniform packet arrival, it does not provide any QoS control. This is crucial to sustain real-time multimedia traffic....[ Read more ]

In recent years there has been a rapid increase in the deployment of multimedia streaming applications such as audio and video broadcasting. However, since the best-effort Internet is an unreliable network with a high packet loss rate and nonuniform packet arrival, it does not provide any QoS control. This is crucial to sustain real-time multimedia traffic.

To resolve the problem, a multimedia bitrate adaptation flow control for streaming multimedia data over the Internet is proposed. It constantly maintains the buffer at a prescribed capacity, even with bursty network loss, by adapting the multimedia bitrate at the streaming encoder. Simulation results showed that the proposed system with multimedia bitrate adaptation can maintain a higher buffer fill-up rate and larger amount of stored playtime, even in bursty loss period, compared to systems without such an adaptation scheme. A loss packet recovery mechanism and a nonuniform packet arrival mechanism are also proposed to provide error recovery for the system and to deal with the out-of-sequence packet arrival problems. Moreover, a client-based congestion control algorithm able to resolve network congestion problems by adapting the sending rate of the server is presented. Simulation results showed that the proposed client-based congestion control maintains a degree of TCP-friendliness compared to that of the TFRC scheme used in the TCP-friendly congestion control for multimedia traffic. It also provides better resource allocation among different multimedia traffic using a simple weight factor scheme.

Next, a novel gateway-assisted congestion control mechanism called "Jitter Detection" (JD) is described. This improves the QoS in multimedia networking by detecting and discarding useless packets that have accumulated a large enough delay jitter. The JD scheme helps to maintain a high bandwidth for packets within the delay jitter tolerance of the multimedia traffic. The JD can further be used to stream layered multimedia multicast traffic over the Internet in order to preserve the base layer traffic with the best-effort when passing through the gateways. Simulation results have shown that the proposed JD scheme can effectively lower the average received packet delay jitter and increase the goodput of the received packets while maintaining the same TCP-friendliness compared to those using RED and DropTail schemes. The results have also shown that the modified JD scheme can provide better quality in terms of PSNR than that of using RED for layered multimedia multicast traffic.

Lastly, a "Minimum Redundancy Tree" (MRT) is presented for key distribution in secure multimedia multicast, in order to reduce the update communication overhead for re-keying. The MRT is optimal in terms of minimum re-keying costs in that it keeps the minimum average number of keys needed to be updated for each member and maintains the minimum average tree height for each member. The tree update procedure maintains the optimality of the key tree after the re-keying. Analytical analysis of the proposed algorithms is presented and the key tree update interval under constrained network resources is computed. By combining MRT and subgrouping, multiple MRTs can be generated such that the member storage, GC storage and update communication overhead can further be minimized compared to those of other key management schemes.