Bundling frames to save energy while streaming video from LTE mobile device

REDUP: a packet loss recovery scheme for real-time audio streaming over wireless IP networks

Multi-hop wireless networks (MWNs) provide an important infrastructure for ubiquitous multimedia content access. However, the accumulated packet losses greatly decrease the quality of multimedia services, particularly video streaming services. In order to overcome packet losses in MWNs, in this paper we propose a proportional integral derivative control-based forward error correction (PID-FEC) mechanism to improve the quality of video streaming services. Our proposed method adopts the cross-layer approach, and leverages the functionalities of different network layers. The automatic repeat request (ARQ) on the media access control (MAC) layer is used as an indicator of packet losses. With the packet loss information, the redundancy rates are adaptively regulated based on the PID control algorithm. Experimental results, presented herein, show that the PID-FEC achieves a better quality of video streaming, as well as a higher FEC efficiency, as compared with conventional FEC schemes, over a variety of network conditions.

Until now, multi-hop device-to-device (MD2D) routing in mobile wireless networks has been designed to operate over a single-protocol routing framework. Consequently, to the best of our knowledge, no framework has been developed to enable the dynamic deployment and switching of multiple MD2D routing protocols under one framework. This paper proposes a multi-protocol framework to introduce the idea of clustering a cell and deploying different MD2D routing protocols for each cluster based on the cluster requirements. To this end, four clusters are developed with varying network conditions (i.e., node density, mobility rate, and the number of flows). Then, the performance of our two previously designed multi-hop routing protocols, namely hybrid SDN architecture for wireless distributed networks (HSAW) and source-based virtual ad hoc routing protocol (VARP-S), are investigated in each cluster in terms of energy consumption, end-to-end (E2E) delay, packet loss, and cellular-band overhead. Based on the achieved simulation results, a multiple-criteria decision-making (MCDM) approach based on the analytic hierarchy process (AHP) is developed to choose the most suitable protocol for each cluster to provide the best performance. The simulation results indicate that our proposed multi-protocol framework provides better performance compared to traditional single-protocol architectures.

In this paper, we analyse the results of a seven-month real-time streaming experiment, which was conducted between a number of unicast dialup clients, connecting to the Internet through access points in more than 600 major U.S. cities, and a backbone video server. During the experiment, the clients streamed low-bitrate MPEG-4 video sequences from the server over paths with more than 5,000 distinct Internet routers. We describe the methodology of the experiment, the architecture of our NACK-based streaming application, study end-to-end dynamics of 16 thousand ten-minute sessions (85 million packets), and analyze the behavior of the following network parameters: packet loss, round-trip delay, one-way delay jitter, packet reordering, and path asymmetry. We also study the impact of these parameters on the quality of real-time streaming.

Real-time HTTP streaming has gained global popularity for delivering video content over Internet. In particular, the recent MPEG-DASH (Dynamic Adaptive Streaming over HTTP) standard enables on-demand, live, and adaptive Internet streaming in response to network bandwidth fluctuations. Meanwhile, emerging is the new-generation video coding standard, H.265/HEVC (High Efficiency Video Coding) promises to reduce the bandwidth requirement by 50% at the same video quality when compared with the current H.264/AVC standard. However, little existing work has addressed the integration of the DASH and HEVC standards, let alone empirical performance evaluation of such systems. This paper presents an experimental HEVC-DASH system, which is a pull-based adaptive streaming solution that delivers HEVC-coded video content through conventional HTTP servers where the client switches to its desired quality, resolution or bitrate based on the available network bandwidth. Previous studies in DASH have focused on H.264/AVC, whereas we present an empirical evaluation of the HEVC-DASH system by implementing a real-world test bed, which consists of an Apache HTTP Server with GPAC, an MP4Client (GPAC) with open HEVC-based DASH client and a NETEM box in the middle emulating different network conditions. We investigate and analyze the performance of HEVC-DASH by exploring the impact of various network conditions such as packet loss, bandwidth and delay on video quality. Furthermore, we compare the Intra and Random Access profiles of HEVC coding with the Intra profile of H.264/AVC when the correspondingly encoded video is streamed with DASH. Finally, we explore the correlation among the quality metrics and network conditions, and empirically establish under which conditions the different codecs can provide satisfactory performance.

Maintaining the Quality-of-Service (QoS) of Voice-over-Internet Protocol (VoIP) is a major problem due to the instability of network conditions. One factor that influences VoIP call quality is the employed codec. In this research, the influence of codec on the QoS of VoIP calls is evaluated under Wi-Fi, 2G, 3G and 4G networks. VoIP calls are established using PCMA, PCMU, GSM and G.722 codecs and QoS parameters including jitter, round trip time and packet loss are measured. QoS measurements are compared against considered networks and codecs. Results illustrate that there exists a codec that better suits each network with considerable advantages over other codecs and minimum drawbacks.

One of TCP's critical tasks is to determine which packets are lost in the network, as a basis for control actions (flow control and packet retransmission). Modern TCP implementations use two mechanisms: timeout, and fast retransmit. Detection via timeout is necessarily a time-consuming operation; fast retransmit, while much quicker, is only effective for a small fraction of packet losses. In this paper we consider the problem of packet loss detection in TCP more generally. We concentrate on the fact that TCP's control actions are necessarily triggered by inference of packet loss, rather than conclusive knowledge. This suggests that one might analyze TCP's packet loss detection in a standard inferencing framework based on probability of detection and probability of false alarm. This paper makes two contributions to that end: first, we study an example of more general packet loss inference, namely optimal Bayesian packet loss detection based on round trip time. We show that for long-lived flows, it is frequently possible to achieve high detection probability and low false alarm probability based on measured round trip time. Second, we construct an analytic performance model that incorporates general packet loss inference into TCP. We show that for realistic detection and false alarm probabilities (as are achievable via our Bayesian detector) and for moderate packet loss rates, the use of more general packet loss inference in TCP can improve throughput by as much as 25%.

Cloud gaming is a class of services that promises to revolutionize the videogame market. It allows the user to play a videogame with essential equipment while using a remote server for the actual execution. The multimedia content is streamed through the network from the server to the user. Hence, this service requires low latency and a large bandwidth to work properly with low response time and high-definition video. Three of the leading tech companies (Google, Sony, and NVIDIA) entered this market with their products, and others, like Microsoft and Amazon, are also launching their platforms. However, these companies have released little information about their cloud gaming operation and how they utilize the network. In this work, we study cloud gaming services from the network point of view. We collect more than 200 packet traces under different application settings and network conditions from a broadband network to poor mobile network conditions, for 3 cloud gaming services, namely Stadia from Google, GeForce Now from NVIDIA and PS Now from Sony. We analyze the employed protocols and the workload that they impose on the network. We find that GeForce Now and Stadia use the RTP protocol to stream the multimedia content, with the latter relying on the standard WebRTC APIs. Depending on the network and video quality, they result in bandwidth-hungry services consuming up to 45 Mbit/s. PS Now instead uses only undocumented protocols and never exceeds 13 Mbit/s. 4G mobile networks can often sustain these loads, while traditional 3G connections struggle. The systems quickly react to deteriorated network conditions, and packet losses up to 5% do not cause a reduction in resolution.

In the current Internet video streaming is dominating fixed and mobile consumer traffic. However, wireless and mobile networks may lack bandwidth or suffer from severe bandwidth fluctuations. The question arises: What is the impact of bandwidth fluctuations on the Quality of Service (QoS) and the user perceived quality? The key contribution of this paper is the analysis of QoS in terms of application-level parameters (stalling duration and frequency) for reliable video-on-demand streaming. The QoS is mapped to Quality of Experience (QoE) based on an existing QoS-QoE model for HTTP streaming. Different network conditions, in particular average and variances of network bandwidths, and video bit rates are taken into account in the analysis. We approach this problem by modeling the video player as a queueing system and evaluate the transient phase of the non-stationary system by means of discrete-event simulations. The results of our study show that the characteristics of the network have a high impact on the QoS and a surprisingly low impact on the QoE under certain conditions.

Roaming across multiple basic service sets (BSS) using access points (AP) in wireless LAN (WLAN) always poses considerable challenges, as it will cause heavy packet loss error. Especially for real-time video streaming, it will damage receiving video quality. In this paper, we combine the benefits of multiple description coding (MDC) and multipath routing to maintain a best constant video quality despite packet loss when roaming in WLAN. We incorporate a channel status detection mechanism to decide which channel will be selected or multiple channels will be used to take advantage of path diversity to deliver the streaming video content. The loss-rate and round-trip time are used to determine the channel status by using active probing. Simulation results indicate that the proposed method can significantly improve the quality of streamed video when a mobile client roams in WLAN.

Seamless access to video services in Next Generation Mobile Networks (NGMNs) introduces new challenges in the video quality provisioning. Terminal mobility and network handover affect the Quality of Service (QoS) and the perception of the mobile users. To improve the Quality of Experience (QoE) in NGMNs, cross-layer adaptation of the video streaming is required. A technique to replace the applied video codec during ongoing real-time video transmissions, such as video call, is described in this paper. The presented solution enables to seamlessly switch between the video codecs independently from the applied coding schemes, allows to maintain ongoing sessions, and helps to reduce the changeover-related artifacts to a large degree. The proposed algorithm has been implemented and evaluated in a NGMN testbed. The technique enabled to significantly improve user perception during wireless transmission affected by packet loss, and performed best as compared to network handover and video bit rate reduction that are considered as alternative quality adaptation measures. This proves the relevance of the presented codec changeover solution for video quality provisioning in mobile networks.

The use of traditional TCP, in its present form, for reliable transport over Ad hoc Wireless Networks (AWNs) leads to a significant degradation in the network performance. This is primarily due to the congestion window (cwnd) updation and congestion control mechanisms employed by TCP and its inability to distinguish congestion losses from wireless losses. In order to provide an efficient reliable transport over AWNs, we propose Learning-TCP, a novel learning automata based reliable transport protocol, which efficiently adjusts the cwnd size and thus reduces the packet losses. The key idea behind Learning-TCP is that, it dynamically adapts to the changing network conditions and appropriately updates the cwnd size by observing the arrival of acknowledgment (ACK) and duplicate ACK (DUPACK) packets. Learning-TCP, unlike other existing proposals for reliable transport over AWNs, does not require any explicit feedback, such as congestion and link failure notifications, from the network. We provide extensive simulation studies of Learning-TCP under varying network conditions, that show increased throughput (9-18%) and reduced packet loss (42-55%) compared to that of TCP.

With the development of Internet computing techniques, continuous data streams from remote sites are commonly used in scientific and commercial applications. Correspondingly, there is increasing demand of assuring the integrity and authenticity of received data streams. Existing strategies of assuring data integrity and authenticity mainly use message authentication codes (MAC) generated on data blocks and transfer the MAC to the receiver for authentication through either out of band communication or in band communication. Transferring the MAC via out of band communication inevitably introduces communication overhead and additional complexity to synchronize the out of band communication with the data communication. Transferring the MAC via in band channel can be achieved by either appending the MAC to the original data or embedding the MAC into the original data, which would either incur communication overhead or change the original data. It would be desirable to be able to authenticate the stream data without any communication overhead and changing the original data at the same time. To deal with data packet or block loss, many of existing stream data authentication schemes rely on hash chaining, the current usage of which results in uncertainty in authenticating the subsequent data blocks once the first data packet or block loss is detected. In this paper, we propose a novel application layer authentication strategy called DaTA. This authentication scheme requires no change to the original data and causes no additional communication overhead. In addition, it can continue authenticating the rest of data stream even if some data loss has been detected. Our analysis shows that our authentication scheme is robust against packet loss and network jitter. We have implemented a prototype system to evaluate its performance. Our empirical results show that our proposed scheme is efficient and practical under various network conditions

In this paper, we present the implementation and analysis of our implemented testbed considering the Optimized Link State Routing (OLSR) protocol. We investigate the effect of mobility and topology changing in the throughput of a Mobile Ad hoc Network (MANET). The mobile nodes move toward the destination at a regular speed. When the mobile nodes arrive at the corner, they stop for about three seconds. Experimental time is 150 seconds. In this work, we consider 6 models and we assess the performance of our testbed in terms of throughput, round trip time and packet loss. From our experiments, we found that the OLSR protocol has a good performance when the source node is moving. However, the performance is not good when the relay nodes are moving.

An overview is presented of compression and packet loss effects in speech biometrics. These new problems appear particularly in recent applications of biometrics over mobile or Internet networks. The influence of speech compression on speaker recognition performance in mobile networks is investigated. In a first experiment, it is found that the use of GSM coding degrades the performance. In a second experiment, the features for the speaker recognition system are calculated directly from the information available in the encoded bit stream. It is found that a low LPC order in GSM coding is responsible for most performance degradations. A speaker recognition system was obtained which is equivalent in performance to the original one which decodes and reanalyses speech before performing recognition. The joint packet loss and compression effects over IP networks are also studied. It is experimentally demonstrated that the adverse effects of packet loss alone are negligible, while the encoding of speech, particularly at a low bit rate, coupled with packet loss, can reduce the verification accuracy considerably.