A Model for Mapping Combined Effects of Quality of Service Parameters and Device Features on Video Streaming Quality of Experience

Maintaining quality of streaming video is challenged by network faults resulting into freezes and rebufferings on the video. On top of the network effects, device features have impacts on the image of the video frames displayed during streaming. Despite the simultaneous impacts of video quality from network and device, previous studies considered individual impact of network parameters or devices as influencing factors to propose Quality of Experience (QoE) models. This study proposed QoE model by mapping combined effects from both network and device on video streamed QoE. An experiment to study the effects of video quality from combined effects of network and device over the wireless involved 35 subjects. Combination of packet loss, packet reordering, and delay were emulated using network emulator following Design of Experiment methodology. Through analysis of variance, the study found that packet loss had the highest impact, followed by device features, reordering, and delay on the video QoE. From the combined effects, two-way interactions and three-way interactions had significant effects on the video QoE. Through additive and linearity behavior of the input factors from network and device on video streaming QoE, a multi-factor model was derived. Keywords: Design of Experiment (DOE); Mean Opinion Score (MOS); Quality of Experience (QoE); Quality of Service (QoS); Video Quality Assessmen

A Framework for Multiaccess Support for Unreliable Internet Traffic using Multipath DCCP

Mobile nodes are typically equipped with multiple radios and can connect to multiple radio access networks (e.g. WiFi, LTE and 5G). Consequently, it is important to design mechanisms that efficiently manage multiple network interfaces for aggregating the capacity, steering of traffic flows or switching flows among multiple interfaces. While such multi-access solutions have the potential to increase the overall traffic throughput and communication reliability, the variable latencies on different access links introduce packet delay variation which has negative effect on the application quality of service and user quality of experience. In this paper, we present a new IP-compatible multipath framework for heterogeneous access networks. The framework uses Multipath Datagram Congestion Control Protocol (MP-DCCP) - a set of extensions to regular DCCP - to enable a transport connection to operate across multiple access networks, simultaneously. We present the design of the new protocol framework and show simulation and experimental testbed results that (1) demonstrate the operation of the new framework, and (2) demonstrate the ability of our solution to manage significant packet delay variation caused by the asymmetry of network paths, by applying pluggable packet scheduling or reordering algorithms

Network degradation effects on different codec types and characteristics of video streaming

Nowadays, there is a quickly growing demand for the transmission of voice, video and data over an IP based network. Multimedia, whether we are talking about broadcast, audio and video transmission and others, from a global perspective is growing exponentially with time. With incoming requests from users, new technologies for data transfer are continually developing. Data must be delivered reliably and with the fewest losses at such high speed. Video quality as part of multimedia technology has a very important role nowadays. It is influenced by several factors, where each of them can have many forms and processing. Network performance is the major degradation effect that influences the quality of resulting image. Poor network performance (lack of link capacity, high network load…) causes data packet losses or different delivery time for each packet. This work focuses exactly on these network phenomena. It examines the impact of different delays and packet losses on the quality parameters of triple play services, to evaluate the results using objective methods. The aim of this work is to bring a detailed view on the performance of video streaming over IP-based networks

ChitChat: Making Video Chat Robust to Packet Loss

Video chat is increasingly popular among Internet users. Often, however, chatting sessions suffer from packet loss, which causes video outage and poor quality. Existing solutions however are unsatisfying. Retransmissions increase the delay and hence can interact negatively with the strict timing requirements of interactive video. FEC codes introduce extra overhead and hence reduce the bandwidth available for video data even in the absence of packet loss. This paper presents ChitChat, a new approach for reliable video chat that neither delays frames nor introduces bandwidth overhead. The key idea is to ensure that the information in each packet describes the whole frame. As a result, even when some packets are lost, the receiver can still use the received packets to decode a smooth version of the original frame. This reduces frame loss and the resulting video freezes and improves the perceived video quality. We have implemented ChitChat and evaluated it over multiple Internet paths. In comparison to Windows Live Messenger 2009, our method reduces the occurrences of video outage events by more than an order of magnitude

RTP MIDI : Recovery Journal Evaluation and Alternative Proposal

An RTP payload for MIDI commands is under development. As a part of this draft, a default resiliency mechanism for the transport over lossy networks defines a journalling method called recovery journal. But the theoretical size of this recovery journal can be very large and its format is complex. This report will present an empirical evaluation of the recovery journal size based on a few MidiFiles. We will also propose an alternative solution for the resiliency of RTP MIDI streams based on the combined use of redundancy and retransmissions. Our solution is simpler and might be interesting for some scenarios, typically: short grouping times, complex streams or unconventional semantics

Improving the Quality of Real Time Media Applications through Sending the Best Packet Next

Real time media applications such as video conferencing are increasing in usage. These bandwidth intensive applications put high demands on a network and often the quality experienced by the user is sub-optimal. In a traditional network stack, data from an application is transmitted in the order that it is received. This thesis proposes a scheme called "Send the Best Packet Next (SBPN)" where the most important data is transmitted first and data that will not reach the receiver before an expiry time is not transmitted. In SBPN the packet priority and expiry time are added to a packet and used in conjunction with the Round Trip Time (RTT) to determine whether packets are sent, and in which order that they are sent. For example, it has been shown that audio is more important to users than video in video conferencing. SBPN could be considered to be Quality of Service (QoS) that is within an application data stream. This is in comparison to network routers that provide QoS to whole streams such as Voice over IP (VoIP), but do not differentiate between data items within the stream or which data gets transmitted by the end nodes. Implementation of SBPN can be done on the server only, so that much of the benefit for one way transmission (e.g. live television) can be gained without requiring existing clients to be changed. SBPN was implemented in a Linux kernel on top of Datagram Congestion Control Protocol (DCCP) and compared to existing solutions. This showed real improvement in the measured quality of audio with a maximum improvement of 15% in selected test scenarios

Packet loss visibility across SD, HD, 3D, and UHD video streams

The trend towards video streaming with increased spatial resolutions and dimensions, SD, HD, 3D, and 4kUHD, even for portable devices has important implications for displayed video quality. There is an interplay between packetization, packet loss visibility, choice of codec, and viewing conditions, which implies that prior studies at lower resolutions may not be as relevant. This paper presents two sets of experiments, the one at a Variable BitRate (VBR) and the other at a Constant BitRate (CBR), which highlight different aspects of the interpretation. The latter experiments also compare and contrast encoding with either an H.264 or an High Efficiency Video Coding (HEVC) codec, with all results recorded as objective Mean Opinion Score (MOS). The video quality assessments will be of interest to those considering: the bitrates and expected quality in error-prone environments; or, in fact, whether to use a reliable transport protocol to prevent all errors, at a cost in jitter and latency, rather than tolerate low levels of packet errors

Performance Evaluation of a Wireless Network using a VoIP Traffic Generator on a Mobile Device

The problem of generating different patterns of traffic to emulate real user behaviour is receiving considerable attention with the construction of new and more complex network architectures. The theoretical modelling of waveforms or signals that flow through networks is valuable in a variety of scenarios including performance analysis and the design of communication systems. In the literature, many computer-based performance evaluation tools have been discussed. However, these tools lack the ability to run on affordable technologies such as mobile phones. The fundamental contribution of this work is the design of a traffic generating tool called MTGawn which is able to run on a mobile device. Design Science Research was the research methodology used for the design and deployment of a prototype of the proposed system. VoIP traffic was emulated using an implementation of well-known real time transport protocols such as RTP and cRTP, and parameterization was defined by using three codecs namely: G.711, G.723, and G.729. An evaluation was performed in a laboratory wireless network testbed and preliminary results were collected and analysed. The results of the experiments show that such a measuring instrument can be deployed on a mobile phone. More experiments are being done to ensure the accuracy of the data and also to compare the results with that of computer-based systems. Furthermore additional functionalities, similar to the functionality found on the computer-based open source tools, are being added to the mobile tool.Telkom, Cisco, Aria Technologies, THRIPDepartment of HE and Training approved lis

Assessment of Recovery Journal-Based Packet Loss Concealment Techniques for Low-Latency MIDI Streaming

In networked music performances, real-time Packet Loss Concealment (PLC) is a task of pivotal importance to compensate the detrimental impact of loss or late delivery of audio portions that often occur in low-latency audio-streaming scenarios.\\ This paper proposes an open-loop PLC method tailored for MIDI data and compares it to a closed-loop state-of-the-art benchmark in terms of effectiveness of audio recovery and communication overhead. Moreover, implementations aimed at reducing the computational overhead are proposed and compared for both approaches. Results show that the proposed open-loop policy achieves performances similar to those of the closed-loop one, while reducing the number of operations executed at the transmitter side