We describe a new characteristic of soft handoff, call failure when a mobile moves from the handoff region to the normal region, and introduce the metric of region-transition failure probability to more accurately assess the performance of CDMA systems that support real-time video streaming services. This characteristic has not been considered in previous research. Simulations show that this failure must be considered since it significantly degrades system performance.

The audio delay is becoming an important factor in audio streaming over short-range wireless network. In this study, we propose an efficient two-level delay control method, called frame sequence adaptation and audio sampling frequency compensation, for achieving stable audio delay with a small variation. To prove the effectiveness of our scheme, we implemented and evaluated the scheme on a Bluetooth network. Experimental results show that our scheme can control audio delay robustly and remove phase shift problem in multi-channel stereophonic audio broadcasting as well.

An H.264-based selective FGS coding scheme is proposed. It selectively uses the interframe-prediction data inside the enhancement-layer only when those data can significantly reduce the temporal-redundancies. Since this minimizes the drift effects, the overall coding efficiency is improved. Simulations show that average PSNR of the proposed scheme is higher by 1-3 dB and 3-5 dB than those of the H.264-based FGS and the MPEG-4 video FGS profile, respectively.

Vertical handoff is required to achieve anywhere and anytime internet access in the fourth generation (4G) network providing interoperability between universal mobile telecommunications system (UMTS) and wireless LAN (WLAN). However, video data can be lost due to latency caused by vertical handoff. To solve this problem, in this paper, we propose a video streaming technique supporting error concealment (EC) for video on demand (VOD) services that provides seamless playout at the client in vertical handoff. In the proposed method, the streaming server first predicts the client buffer status (CBS). Using the predicted CBS and the channel rate, the streaming server selects a proper video transmission method for vertical handoff between frame selective pre (FSP)-transmission and re-transmission. Performance evaluations are presented to demonstrate the effectiveness of the proposed method.

Live streaming is delay sensitive and can tolerate some amount of loss. The QoS Multicast for Live Streaming (QMLS) Protocol, focuses on the characteristics of live streaming. It has been shown to improve the performance of live streaming multicast by reducing the end-to-end packet loss probability. However, the placement of active routers performing the QMLS function has not been discussed. This paper proposes a dynamic method to activate and deactivate routers in order to minimize the number of active routers for each QMLS-packet flow and discusses its parameters. The results of an evaluation show that the proposed method can reduce the number of active routers for each flow and adjust the active routers according to changes in the multicast tree.

IEEE 802.11 DCF is a contention-based channel access protocol, and medium access delay greatly increases as the number of contending stations in a service area increases. This severely affects on delay-sensitive video applications. In this paper, we focus on MAC-layer solutions for realizing high quality video transmission in 802.11 DCF networks. A new channel access protocol based on the timestamp (TS) of video packets is proposed for real-time video. The TS information is carried by RTP header from the video application to 802.11 MAC. For video packets with the same RTP TS, they are simultaneously transmitted in a single channel access. Additional contention and back-off processes can be avoided because the whole packets of a video frame are completely delivered. The proposed TS-based access protocol can be easily implemented by the DCF with packet bursting. In addition, it is backward compatible to the legacy DCF. Extensive simulations show that the TS-based channel access achieves lower cumulative distributions of application-level video frame delay when compared to the DCF protocol.

This letter studies the effect of node mobility on application-level QoS of audio-video multipath streams in wireless ad hoc networks. The audio-video streams are transmitted with the MultiPath streaming scheme with Media Synchronization control (MPMS), which was previously proposed by the authors. We perform computer simulation with a grid topology network of IEEE 802.11b including two mobile nodes. The simulation results show that MPMS is effective in achieving high application-level QoS in mobile networks as well.

Providing efficient Video-On-Demand (VOD) service on Broadband Cable Networks (BCNs) is still a challenging issue, although BCNs have been the most important form of broadband services in the US. To address this issue, we have proposed the optimal full-sharing scheduling approach that minimizes the bandwidth consumption of video sessions [1]. We have shown that full-sharing scheduling has remarkable advantages in minimizing the bandwidth consumption of VOD service on BCNs. Furthermore, we design two adaptation algorithms in this paper, which not only keep bandwidth consumption minimal but also significantly reduce the mean service delay.

This paper proposes a subjective model for estimating the quality of video streaming services with dynamic bit-rate control. In a subjective quality assessment test, we clarify users' perceptions of distributed video signals whose quality is time-variant due to dynamic bit-rate control. Using this result, we constructed an estimation model considering the following three characteristics: 1) the influence of the video section where quality degradation is large will strongly affect the overall quality, 2) the impression of a past quality weakens with the passage of time, and 3) the range of evaluation scores becomes wider when the time duration of an evaluation is longer. We found that the proposed model enables the accuracy of estimating overall subjective quality to be dramatically improved compared with that of a model that averages segmental quality. The estimation error of the proposed model is less than the statistical reliability of the subjective score even for verification data. We also show that our findings are applicable to QoS design/management issues for video streaming services with dynamic bit-rate control.

In this paper, we propose a novel dynamic bandwidth allocation scheme for the downlink real-time video streaming in the wireless cellular networks. Our scheme is able to maximize the bandwidth utilization, while satisfying the required packet loss probability, a QoS constraint, by dynamically determining the amount of bandwidth to be allocated at each unit time interval by measuring the queue length and the packet loss probability. The simulation results show that, without the need of a priori knowledge about the traffic traces, our scheme is able to achieve the same level of performance as what can be accomplished with the pre-calculated effective bandwidth in terms of the bandwidth utilization and the packet loss rate.

Regarding IP-based video applications over wireless networks, the multi-layer header overhead may significantly affect the estimation of target video encoding bit rate and the effective throughput of wireless network. Based on the existing header structure of video packets, this study intends to deal with the header overhead problem from the end-to-end viewpoint. This paper first proposes a simple yet robust closed-form that can determine accurately and timely the optimal video payload length at the video sender based on the current wireless channel condition. The contribution can effectively improve the WLAN throughput and enhance the error resilience effect of scalable video data simultaneously. This study further explores the impact of multi-layer header overhead to the video coding work and proposes a Dynamic Header Overhead Accommodation (DHOA) scheme, which is executed in the video compression layer, to adjust dynamically the available video encoding bits for accommodating the header overhead in advance. The contributions of this paper are robust for various IP implementations such as IPSec (IP Security) over different 802.11 standards. Analytical and simulation results verify the accuracy and effectiveness of the proposed closed-form and header accommodation method. Using DHOA, the bandwidth mismatch between the actual bandwidth demand of packetized video data and the available network bandwidth is no more than 1.1% regardless of the packet sizes used in this paper.

This paper proposes the MultiPath streaming scheme with Media Synchronization control (MPMS) for audio-video transmission in wireless ad hoc networks. In many audio-video streaming applications, media compensate each other from a perceptual point of view. On the basis of this property, we treat the two streams as separate transport streams, and then the source transmits them into two different routes if multiple routes to the destination are available. The multipath transmission disturbs the temporal structure of the streams; in MPMS, the disturbance is remedied by media synchronization control. In order to implement MPMS in this paper, we enhance the existing Dynamic Source Routing (DSR) protocol. We compare the application-level QoS of MPMS and three other schemes for audio-video transmission by simulation with ns-2. In the simulation, we also assess the influence of the multipath transmission on other traffic. The simulation result shows that MPMS is effective in achieving high QoS at the application-level.

One of the important problems in overlay multicast is how to deal with node failures and ungraceful leavings. When a non-leaf end host fails or leaves the multicast session, all downstream nodes will be affected. In this paper, we adopt the proactive approach, which pre-calculates a candidate node (called parent-to-be) for each node to connect to in case its current parent dies. The goal is to recover the overlay multicast tree quickly so that the disruption of service to those affected nodes is minimized. We combine the local parent-to-be locating and global parent-to-be locating schemes together, in order to take advantage of less interference in the local scheme and the flexibility of the global scheme. The quality of the recovered tree is improved while the responsiveness of the proactive approach is maintained.

Live streaming media are delay sensitive and have limited allowable delays. Current conventional multicast protocols do not have a loss retransmission mechanism. Even though several reliable multicast protocols with retransmission mechanisms have been proposed, the long delay and high packet loss rate make them inefficient for live streaming. This paper proposes a multicast protocol focusing on the allowable delay called the QoS Multicast for Live Streaming (QMLS) protocol. QMLS routers are placed along the multicast tree to detect and retransmit lost packets. We propose a method that enables data recovery to be done immediately after lost packets are detected by the QMLS router and a method that reduces the unnecessary packets sent to end receivers. This paper discusses the mathematical analysis of the proposed protocol and compares it with other multicast protocols. The results reveal that our protocol is more effective in live streaming. Finally, we do a simulation to evaluate its performance and study the effect of consecutive losses. The simulation reveals that consecutive losses can slightly increase losses with our protocol.

This paper introduces the water ring scan method especially designed for the scalable video coding schemes such as fine granularity scalabilities (FGS) on the basis of MPEG-4 part-2 and the H.264. The proposed scanning method can improve the subjective quality of the decoded video by most-preferentially encoding, transmitting and decoding the image information of the region of interest. From the various simulation results of FGS coding schemes with MPEG-4 part 2 and H.264, the proposed scanning method can improve the subjective picture quality about 0.5 dB 3.5 dB better than the widely used raster scan order, especially on the region of interest, without significant loss of the quality in the left-over region.

CDN (Content Delivery Networks) improves end-user performance by replicating web contents on a group of geographically distributed servers. However, repeatedly keeping the entire replica of the original objects into many content servers consumes too much server resource. This problem becomes more serious for the large-sized objects such as streaming media, e.g. high quality video. In this paper, we therefore propose an efficient replication method for layered video streams in CDN, which can reduce user response delays and storage costs simultaneously. Based on an analytical formulation of the cooperative replication of layers and segments of each video stream, we derive a replication algorithm which solves next three problems quantitatively. (1) How many servers should be selected to replicate a given video stream? (2) For a single video stream, how many layers and segments should be stored in a given server? (3) After selecting a group of servers for each video stream, how do we allocate the replication priority (i.e. order) to each server? Simulation results verify that the proposed algorithm efficiently resolves the above problems and provides much better performance than conventional methods.

This innovative traffic-monitoring-system makes it possible to observe data-communication traffic on an oscilloscope-style display. It provides an efficient way of evaluating streaming-data quality. The monitoring system has a high time-resolution traffic value sampling function and a real-time data representation/recording mechanism that operate in synchrony. The user can directly evaluate the traffic shape with the monitoring system. In this paper, after describing the concept of the traffic monitoring system, we will describe a prototype built with programmable network equipment called A-BOX. We will then review a performance evaluation and other experimental results to prove that our monitoring system is suitable for video streaming.

In this paper, we propose an autonomously distributed QoS control method for MPEG video streaming in a wide area network. The capacity of the links and the characteristics of video streams change dynamically. However, managing the condition of all the links and streams in the network is difficult. In the proposed method, the routers in the network monitor the conditions of the links and streams locally and control the transmission rate of the stream server. Picture-quality oriented fairness is achieved by reducing the transmission rate of the streams with the higher PSNR in the bottleneck link. The computer simulation results show that the proposed method can be applied to a wide area network.

In this paper, we propose an MPEG-2 TS encryption and decryption scheme which can support MPEG-2 TS streaming, and a key management scheme which can provide secure delivery of the key used to encrypt MPEG-2 TS from the package server to DRM clients in a secure manner. The key management scheme protects the key not only from purchasers but also among the other principals who manage the distribution servers and license servers. The proposed scheme can protect from attacks over a network since a pre-encrypted MPEG-2 TS is sent by a streaming server and only DRM clients can decrypt the streamed MPEG-2 TS. An MPEG-2 TS streaming server can send the encrypted stream without modifying it and degrading its performance because MPEG-2 TS is encrypted but compliant to the MPEG-2 TS format. The processing time to decrypt a streamed MPEG-2 TS in client systems is minimized to guarantee the quality of streaming.

In packet networks including the Internet and commercial 3G wireless bearers, the network states that a streaming media application experiences are not known a priori and exhibit time-varying characteristics. For such dynamic environments, network-adaptive techniques are essential to efficiently deliver video data. In this paper, we propose a frame-based optimal scheduling algorithm which incorporates a MAP (Maximum A Posteriori) framework for adapting to varying network loss rate. The optimal transmission schedule is determined such that effective frame-rate is maximized at playback. Also, for multiple packets per frame, frame-based selection of delivery order greatly reduces computational complexity for a server scheduler when compared with packet-based scheduling techniques. In addition, by dynamically estimating instantaneous packet loss probability, the proposed scheduler performs network-adaptive transmission for streaming video over time-varying packet networks. Simulation results for test video sequence show that the proposed scheduling algorithm outperforms conventional ARQ-based schemes from a view point of reconstructed video quality as well as playable frame-rate. In particular, the proposed scheduling algorithm exhibits significant improvements of frame-rate over highly lossy channels.