Buffer-driven adaptive video streaming with TCP-friendliness
Introduction
The Internet has recently been experiencing an explosive growth in the use of audio and video streaming applications. Such applications are delay-sensitive, semi-reliable, and rate-based. Loss-tolerant video streaming applications, such as video conferencing, prefer UDP (User Datagram Protocol) as a transport protocol that has no congestion control algorithm. The emergence of non-congestion-controlled streaming applications threatens unfairness to competing TCP (transmission control protocol) traffic and possible congestion collapse. UDP is considered selfish and ill-behaving because TCP throttles its transmission rate against the network congestion, whereas UDP does not have such a congestion control algorithm. As a result, the available bandwidth to TCP connections is oppressed and their performance sharply deteriorates because the current Internet does not attempt to guarantee an upper limit on end-to-end delay and a lower limit on available bandwidth [1].
Studies on the congestion-controlled streaming mechanism has been increasingly done since the 1990s [2], [3], [4], [5], [6], [7], [8], [9], [10], [11], [12], [13], [14], [15], [16]. These works attempt to guarantee the network stability and fairness with competing TCP connections. In [1], non-TCP connections are defined as TCP-friendly when “their long-term throughput does not exceed the throughput of a conformant TCP connection under the same network conditions”. TCP-friendly rate control mechanisms regulate its data sending rate according to the network condition, typically expressed in terms of the RTT (round trip time) and the packet loss probability. Therefore, they improve the network stability and fairness with competing TCP connections. However, by considering only the network stability, these works ignore the quality of video stream which is delivered to the user. Moreover, most of existing streaming mechanisms have no consideration for the characteristics of video stream which affects significantly the quality of streaming services.
In this paper, we propose a new adaptive video streaming mechanism. The proposed mechanism adjusts the sending rate of video stream-based on the current network state and controls the quality of video stream-based on the receiver buffer occupancy. It also considers the characteristics of video stream which is requested from the user. Therefore, our mechanism maintains the network stability accomplished by previous works and achieves the smooth playback by preventing the buffer underflow or overflow.
The rest of this paper is organized as follows. In Section 2, we review and discuss some of the previous works and in Section 3, we present the concept and algorithms introduced in our mechanism. Detailed description of our simulation results are presented in Section 4. Finally, Section 5 concludes the paper and discusses some of our future work.
Section snippets
Related work
While data applications such as Web and FTP (File Transfer Protocol) are mostly based on TCP, video streaming applications will be based on UDP due to its real-time characteristics. However, UDP does not support a congestion control algorithm. For this reason, UDP-based streaming mechanisms cause the unfairness with competing TCP traffic and the starvation of congestion-controlled TCP traffic which reduces its bandwidth share during overload situations. To overcome this limitation, several
Adaptive video streaming mechanism
This section briefly introduces our adaptive video streaming mechanism to adjust the sending rate of the video stream to the desired TCP-friendly rate. First, we describe the end-to-end architecture of the proposed mechanism. After that, the detailed algorithms are described.
Simulation environment
In this section, we present our simulation results. Using the ns-2 simulator, the performance of the proposed adaptive video streaming mechanism has been measured and compared with the TFRC and the buffer-driven scheme [20]. To emulate the competing network conditions, background TCP traffics are introduced.
Fig. 6 shows the network topology for our simulations. All of our experiments use a single bottleneck topology and the bottleneck queue is managed with the drop-tail mechanism. The RTCP
Conclusion
Most of streaming mechanisms are based on UDP with no end-to-end congestion control. For this reason, wide usage of multimedia applications in Internet might lead to congested networks. To avoid such a situation, several congestion controlled streaming mechanisms were proposed recently. However, by considering only the network stability, most of previous works ignore the requirements of streaming services and the characteristics of the video stream.
In this paper, in order to overcome
Acknowledgement
This research has been conducted by the Research Grant of Kwangwoon University in 2007.
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