Real-time multimedia streaming applications require a strict bounded end-to-end delay and are considered to be bursty as each video frame is typically transmitted as a burst of packets. In this paper we show how the distribution of video frame sizes can be used to efficiently dimension the IEEE 802.lie TXOP limit parameter to efficiently deal with this burstiness in order to enhance the transmission of real-time video streaming services. Through experimental investigation, we show that by using the mean video frame size to dimension the TXOP limit parameter, the transmission delay for the video frame is reduced by 67% under heavily loaded conditions. Other techniques investigated in this paper include applying the TXOP facility separately to each of the constituent I, P, and B video frame types.

We perform an experimental investigation of the IEEE 802.11e TXOP facility to enhance the transmission of parallel multimedia streaming sessions through efficient bandwidth reservation and explicitly consider both the audio and video streams. The delay constraints associated with the audio and video streams that comprise a multimedia session pose the greatest challenge since real-time multimedia is particularly sensitive to delay as the packets require a strict bounded end-to-end delay. We show how the TXOPLimit parameter can be efficiently dimensioned to reduce the transmission delay for the video frames. Due to its frame-based nature, video applications are considered to be bursty as each video frame is typically transmitted as a burst of packets. The size of the burst is related to the size of the video frame and the number of packets required to transmit the video frame. The TXOP facility is particularly suited to efficiently deal with this burstiness since it can be used to reserve bandwidth for the duration of the packet burst. Through experimental investigation, we show that there is a significant performance improvement for the video streams by using the TXOPLimit parameter however there is no such improvement for the audio streams. We show that over-dimensioning the TXOPLimit parameter can cause the video stream to seize too much bandwidth which results in a deterioration in performance for the other competing traffic streams. This deterioration becomes more prominent as the number of parallel multimedia streams increases. We show that there is a performance improvement to all traffic streams by providing differentiated service to the constituent I, P, and B video frame types in conjunction with the TXOP facility.

In this paper we perform an experimental investigation of using video frame differentiation in conjunction with the TXOP facility to enhance the transmission of parallel multimedia streaming sessions in IEEE 802.11e. The delay constraints associated with the audio and video streams that comprise a multimedia session pose the greatest challenge since real-time multimedia is particularly sensitive to delay as the packets require a strict bounded end-to-end delay. Video streaming applications are considered to be bursty. This burstiness is due to the frame rate of video, the intrinsic hierarchical structure of the constituent video frame types, and the different compression ratios for the different video frame types. The TXOP facility is particularly suited to efficiently deal with this burstiness since it can be used to reserve bandwidth for the duration of the packet burst associated with a packetised video frame. Through experimental investigation, we show that there is a significant performance improvement for video streaming applications under heavily loaded conditions by differentiating between the constituent video frame types. The results show that video frame differentiation reduces the mean loss rate by 12% and increases the mean PSNRby 13.1dB.

In this paper we perform an experimental investigation of using video frame differentiation in conjunction with the TXOP facility to enhance the transmission of parallel multimedia streaming sessions in IEEE 802.11e. The delay constraints associated with the audio and video streams that comprise a multimedia session pose the greatest challenge since real-time multimedia is particularly sensitive to delay as the packets require a strict bounded end-to-end delay. Video streaming applications are considered to be bursty. This burstiness is due to the frame rate of video, the intrinsic hierarchical structure of the constituent video frame types, and the different compression ratios for the different video frame types. The TXOP facility is particularly suited to efficiently deal with this burstiness since it can be used to reserve bandwidth for the duration of the packet burst associated with a packetised video frame. Through experimental investigation, we show that there is a significant performance improvement for video streaming applications under heavily loaded conditions by differentiating between the constituent video frame types. The results show that video frame differentiation reduces the mean loss rate by 12% and increases the mean PSNRby 13.1dB.

The bursty nature of video streaming applications is due to the frame-based structure of video and this has an important impact on the resource requirements of the WLAN, affecting its ability to provide quality of service (QoS) particularly under heavily loaded conditions. In this paper we analyse this bursty behaviour in depth. We show how each video frame is queued at the AP causing the packet delay to vary in a sawtooth manner that is related to the frame rate, the number of packets per video frame, and the packet size. We infer the maximum background traffic load that can be supported so that it does not negatively impact on the video streaming application. We demonstrate that there is a critical threshold load value above which the AP can no longer reliably support the video stream and compare it to the threshold load values calculated through analysis. Using this knowledge, the AP can employ resource allocation mechanisms to regulate the incoming traffic to the AP transmission queue so that QoS can be provided for streaming applications

The performance of video streaming over WLAN networks is not only influenced by the state of the network but also by the encoding configuration parameters of the video stream, such as the video content being streamed, how the video is encoded and how it is transmitted. In this paper, we analyze the unique delay characteristic of video streaming applications in a WLAN environment. We show that the "burstiness" of video is due to the frame-based nature of encoded video. We show how each video frame is transmitted as a burst of packets that is queued at the Access Point causing the delay to exhibit a sawtooth-like characteristic over time that is related to the frame rate and frame structure of the encoded video. To our knowledge, this sawtooth-like characteristic of video streaming over WLAN has not been previously reported on. In this paper, not only do we consider the end-to-end delay, but more importantly we consider the total delay required to transmit the entire video frame. We present experimental results for VBR and CBR video streams and calculate the upper bounds on video encoding parameters for streaming real-time interactive video over a WLAN

Many adaptive delivery mechanisms have been devised for streaming multimedia over best-effort IP networks, such as the Internet. Most of these adaptive schemes do not consider the user's perception of quality when making adaptations. We describe a perceptual quality adaptation algorithm (PQA) and prototype system architecture that uses knowledge of user perceived quality to make adaptation decisions using an optimum adaptation trajectory. This optimum adaptation trajectory indicates how encoding quality should be adapted (upgraded/downgraded) with respect to user perceived quality in response to rapidly fluctuating network conditions. We present simulation results that demonstrate the behavior of a perceptual quality adaptation algorithm.

Many adaptive delivery mechanisms have been devised for streaming multimedia over best-effort IP networks. Most of these adaptive schemes do not consider the user's perception of quality when making adaptations. We propose that an optimum adaptation trajectory exists which indicates how encoding quality should be adapted (upgraded/downgraded) with respect to user perceived quality in response to network conditions. This optimum adaptation trajectory can be used with any transmission adaptation policy. We describe a system architecture that uses knowledge of user perceived quality to make adaptation decisions and give an example of how this knowledge can be used to complement the sender-based adaptation algorithm, LDA.

Currently multimedia is either downloaded before viewing, or streamed over a network. However, the problem of streaming real-time or near real-time applications with a specified quality of service (QoS) over the Internet is still unsolved. The Real-time Transport Protocol (RTP) can be used to facilitate streaming, but also has the potential to support QoS. By gathering network statistics during the session and defining different QoS levels, we propose to adapt multimedia streaming to a fluctuating network load and/or client requests, thereby providing adaptive QoS. We describe the simple server and client applications we have implemented to illustrate this adaptation process.