Voice over IP and video applications continue to increase the amount of traffic over the Internet. These applications utilize the UDP protocol because TCP is not suitable for streaming applications. The flow and congestion control mechanisms of TCP can change the connection transmission rate too drastically, affecting the user-perceived quality of the transmission. Also, the TCP protocol provides a level of reliability that may waste network resources, retransmitting packets that have no value. On the other hand, the use of end-to-end flow and congestion control mechanisms for streaming applications has been acknowledged as an important measure to ease or eliminate the unfairness problem that exist when TCP and UDP share the same congested bottleneck link. Actually, router-based and end-to-end solutions have been proposed to solve this problem. This thesis introduces a new end-to-end protocol based on TCP SACK called SF-SACK that promises to be smooth enough for streaming applications while implementing the known flow and congestion control mechanisms available in TCP. Through simulations, it is shown that in terms of smoothness the SF-SACK protocol is considerably better than TCP SACK and only slightly worse than TFRC. Regarding friendliness, SF-SACK is not completely fair to TCP but considerably fairer than UDP. Furthermore, if SF-SACK is used by both streaming and data-oriented applications, complete fairness is achieved. In addition, SF-SACK only needs sender side modifcations and it is simpler than TFRC.

The imminent inception of third-generation (3G) mobile communication networks offers an unprecedented opportunity for the development of video streaming applications through wireless Internet access. Different design challenges exist in implementing video streaming connections spanning both wired and wireless domains. A split-domain TCP-friendly streaming video transmission protocol is presented based on adaptive rate encoding in the MPEG-4 video format. Network simulations are conducted to demonstrate the benefits and viability of such a video streaming scheme over existing options. Further feature enhancements and refinements are necessary for the proposed protocol to achieve its full potential.

The imminent inception of third-generation (3G) mobile communication networks offers an unprecedented opportunity for the development of video streaming applications through wireless Internet access. Different design challenges exist in implementing video streaming connections spanning both wired and wireless domains. A split-domain TCP-friendly streaming video transmission protocol is presented based on adaptive rate encoding in the MPEG-4 video format. Network simulations are conducted to demonstrate the benefits and viability of such a video streaming scheme over existing options. Further feature enhancements and refinements are necessary for the proposed protocol to achieve its full potential.

In this study, we study the dynamics of network adaptive video streaming and propose novel algorithms for rate distortion control in video streaming. While doing so, we maintain inter-protocol fairness with TCP (Transmission Control Protocol) that is the dominant transport protocol in the current Internet. The proposed algorithms are retransmission-based and necessitate the use of playback buffers in order to tolerate the extra latency introduced by retransmissions. In the first part, we propose a practical network-adaptive streaming scheme based on TCP transport and the idea of Selective Frame Discarding (SFD) that makes use of two-layer temporally scalable video. The efficacy of the SFD scheme is validated for playout buffer times in the order of seconds and therefore makes it suitable more for delay tolerant streaming applications. In the second part of the thesis, we propose an application layer rate-distortion control algorithm which provides Optimal Scheduling and Rate Control (OSRC) policies in the average reward sense in order to achieve efficient streaming of video. The Optimal Scheduling (OS) we propose maximizes the probability of successfully on time delivery according to a prespecified set of rate constraints, and different channel conditions by using Markov Decision Process (MDP) models. On the other hand optimal rate control (RC) is achieved by calculating the optimal rate constraint which minimizes the average distortion of a video streaming session by making use of the video distortion model derived for lossy channels and achievable success probabilities provided by the set of optimal schedules. For numerical examples, we focus on an equation-based TCP friendly rate control (TFRC) protocol where transport layer retransmissions are disabled and Fine Granular Scalable (FGS) coded video is used for improved rate adaptation capabilities but with an additional rate distortion penalty. The efficacy of the proposed OSRC algorithm is demonstrated by means of both analytical results and ns-2 simulations.

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