The handoff in Mobile IP networks causes packet sequence disruption during a packet forwarding procedure and may result in performance degradation in higher layer protocols. We investigate the impact of handoff in the Mobile IPv6 networks, where an optimized routing with the smooth handoff is adopted. The impact on the packet sequence is measured by an 'unstable time period (UTP)' and a 'silence time period (STP).' The UTP explains the time duration of out-of-sequence packets while the STP reflects the blackout duration of a mobile node after the initiation of handoff procedure. With the analysis on the UTP and STP, the total transient time period (denoted as handoff time period or HTP) after the handoff initiation can be estimated. In our previous work, focusing on the UTP, the packet flow sequence under the smooth handoff is analyzed for the Mobile IPv4 networks. The proposed queuing-based analysis is extended in this work for the Mobile IPv6 networks. That is, several modifications are made to conform to Mobile IPv6 and at the same time the queuing analysis itself is improved to better model the handoff procedure. The numerical results show that the queuing delay for the handoff packets (affected by background traffic) and the involved link (or route) capacities affect the estimated UTP, STP, and HTP. In addition, two schemes such as priority queuing and buffered packet forwarding are introduced to reduce the transient period and the improvements are analyzed for comparison.

We present a simple bit allocation scheme based on adaptive coding for MIMO-OFDM (Multiple Input Multiple Output - Orthogonal Frequency Division Multiplexing) systems with V-BLAST (Vertical-Bell laboratories LAyered Space-Time) detector. The proposed scheme controls the code rate of the channel coding and assigns the same modulation and coding to the set of selected sub-channels, which greatly reduces the feedback burden while achieving good performance. Simulation results show that the proposed scheme with minimal feedback provides significant performance improvement over other systems.

Overlay networking makes it easy for users add new network functionalities while keeping existing Internet connectivity intact. This paper introduces SCONE (Service-COmposable InterNEt) as a networking service to facilitate the management of service overlay networking. By looking into the structure of programmable overlay nodes, SCONE provides programmable IP service gateways (PSGs) that ensure high-speed per-flow packet processing for overlay networking. In order to meet the data-rate requirements of various host applications, each PSG is accelerated by hardware packet processing for its data plane. It also leverages the space-efficient pattern matching of entity cloning and provides localized (i.e., de-centralized) services to assist the scalable support for software-defined networking (SDN). An experiment result shows that the proposed PSGs can support high-fidelity overlay networking from both performance and scalability perspectives.

The recent growth in available network bandwidth envisions the wide-spread use of broadband applications such as uncompressed HD-SDI (High-definition serial digital interface) over IP. These cutting-edge applications are also driving the development of a media-oriented infrastructure for networked collaboration. This paper introduces imCast, a high-quality digital media platform dealing with uncompressed HD-SDI over IP, and discusses its internal architecture in depth. imCast mainly provides cost-effective hardware-based approaches for high-quality media acquisition and presentation; flexible software-based approaches for presentation; and allows for economical network transmission. Experimental results (taken over best-effort IP networks) will demonstrate the functional feasibility and performance of imCast.

A new inter-client synchronization framework employing a server-client coordinated adaptive playout and error control toward one-to-many (i.e., multicast) media streaming is discussed in this paper. The proposed adaptive playout mechanism controls the playout speed of audio and video by adopting the time-scale modification of audio. Based on the overall synchronization status as well as the buffer occupancy level, the playout speed of each client is manipulated within a perceptually tolerable range. By coordinating the playout speed of each client, the inter-client synchronization with respect to the target presentation time is smoothly achieved. Furthermore, RTCP-compatible signaling between the server and group-clients is performed to achieve the inter-client synchronization and error recovery, where the exchange of controlling message is restricted. Simulation results show the performance of the proposed multicast media streaming framework.

In this letter, we propose a network-adaptive video streaming scheme based on cross-layered hop-by-hop video rate control in wireless multi-hop networks. We argue that existing end-to-end network-adaptive video rate control schemes, which utilize end-to-end statistics, exhibit serious performance degradation in severely interfered wireless network condition. To cope with this problem, in the proposed scheme, intermediate wireless nodes adjust video sending rate depending upon wireless channel condition measured at MAC (Medium Access Control) layer. Extensive experimental results from an IEEE 802.11a-based testbed show that the proposed scheme improves the perceptual video quality compared to an end-to-end scheme.

In IEEE 802.11 networks, the access point (AP) selection based on the strongest signal strength often results in the extremely unfair bandwidth allocation among mobile users (MUs). In this paper, we propose a distributed AP selection algorithm to achieve a fair bandwidth allocation for MUs. The proposed algorithm gradually balances the AP loads based on max-min fairness for the available multiple bit rate choices in a distributed manner. We analyze the stability and overhead of the proposed algorithm, and show the improvement of the fairness via computer simulation.

This paper introduces a network-aware video delivery framework where the quality-of-service (QoS) interaction between prioritized packet video and relative differentiated service (DiffServ) network is taken into account. With this framework, we propose a dynamic class mapping (DCM) scheme to allow video applications to cope with service degradation and class-based resource constraint in a time-varying network environment. In the proposed scheme, an explicit congestion notification (ECN)-based feedback mechanism is utilized to notify the status of network classes and the received service quality assessment to the end-host applications urgently. Based on the feedback information, DCM agent at ingress point can dynamically re-map each packet onto a network class in order to satisfy the desired QoS requirement. Simulation results verify the enhanced QoS performance of the streaming video application by comparing the static class-mapping and the class re-mapping based on loss-driven feedback.

To guarantee the timely provisioning of QoS to real-time oriented multiparty and distributed computing applications (e.g., video conferencing and grid computing) that require the utilization of time and quantity-related resources, the resource should be reserved in advance. However, this new type of reservation (i.e., advance reservation) may collide with legacy (i.e., immediate) reservations that do not specify their session duration. In this paper, to suggest a balanced solution between sharing pools of resources and the managing of collisions between reservation calls, a revenue-based resource sharing scheme (focusing on the bandwidth) is proposed. The proposed scheme attempts to avoid possible collisions by employing a virtual profile of resources reserved in advance in an effort to achieve enhanced resource utilization. It is shown through NS-2 [22] -based network simulations that the proposed scheme can achieve balanced performance when compared with other schemes, including static resource partitioning and complete resource sharing.