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This work studies the jitter buffer management algorithm for Voice over IP in WebRTC. In particular, it details the core concepts of WebRTC’s jitter buffer management. Furthermore, it investigates how jitter buffer management algorithm behaves under network conditions with packet bursts. It also proposes an approach, different from the default WebRTC algorithm, to avoid distortions that occur under such network conditions. Under packet bursts, when the packet buffer becomes full, the WebRTC jitter buffer algorithm may discard all the packets in the buffer to make room for incoming packets. The proposed approach offers a novel strategy to minimize the number of packets discarded in the presence of packet bursts. Therefore, voice quality as perceived by the user is improved. ITU-T Rec. P.863, which also confirms the improvement, is employed to objectively evaluate the listening quality.

WebRTC includes a set of novel technologies and standards that provide high quality audio, video and arbitrary data exchange. By enabling rich interactivity experience, We-bRTC allows the users to directly interact with use of web and mobile applications without the need for additional plugins. The open standard Opus codec is a default audio codec employed by WebRTC for speech communication and music streaming with a wide range of bitrates supported. This range of bitrates is available for narrowband (300–3400 Hz), wideband (100–7000 Hz) up to super-wideband (50–14000 Hz) audio bandwidths. To estimate a quality experienced by the end user for voice transmission service, E-model standardized in the ITU-T Rec. G.107 (a narrowband version) and the ITU-T Rec. G.107.1 (a wideband version) can be used. With the upcoming extension towards the super-wideband E-model, it will be soon possible to estimate the quality of narrowband, wideband as well as super-wideband speech communication. In this paper, we calculate two codec specific parameters, namely the equipment impairment factor Ie and packet loss robustness Bpl for the super-wideband mode of the Opus codec using the instrumental method. The derived values make it possible to integrate the E-model in realtime communication applications including WebRTC to assess the quality experienced by the end user.

Nowadays, media content can be delivered via diverse broadband and broadcast technologies. Although these different technologies have somehow become rivals, their coordinated usage and convergence, by leveraging of their strengths and complementary characteristics, can bring many benefits to both operators and customers. For example, broadcast TV content can be augmented by on-demand broadband media content to provide enriched and personalized services, such as multi-view TV, audio language selection, and inclusion of real-time web feeds. A piece of evidence is the recent Hybrid Broadcast Broadband TV (HbbTV) standard, which aims at harmonizing the delivery and consumption of (hybrid) broadcast and broadband TV content. A key challenge in these emerging scenarios is the synchronization between the involved media streams, which can be originated by the same or different sources, and delivered via the same or different technologies. To enable synchronized (hybrid) media delivery services, some mechanisms providing timelines at the source side are necessary to accurately time align the involved media streams at the receiver-side. This paper provides a comprehensive review of how clock references (timing) and timestamps (time) are conveyed and interpreted when using the most widespread delivery technologies, such as DVB, RTP/RTCP and MPEG standards (e.g., MPEG-2, MPEG-4, MPEG-DASH, and MMT). It is particularly focused on the format, resolution, frequency, and the position within the bitstream of the fields conveying timing information, as well as on the involved components and packetization aspects. Finally, it provides a survey of proofs of concepts making use of these synchronization related mechanisms. This complete and thorough source of information can be very useful for scholars and practitioners interested in media services with synchronization demands., This work has been funded, partially, by the "Fondo Europeo de Desarrollo Regional" (FEDER) and the Spanish Ministry of Economy and Competitiveness, under its R&D&i Support Program in project with ref TEC2013-45492-R.

The Internet has evolved to a stage where the number of Internet enabled devices exceeds the global population. With much of the required connectivity over wireless, contention for bandwidth is an important issue that has to be addressed. In this context, applications that have certain quality of service QoS requirements must be protected. WiFi-enabled mobile devices such as smartphones and tablets are commonly used by both personal and business users for voice over IP VoIP communications, while also supporting conventional data applications such as email, file transfer and web access. Previous research has shown that time synchronized endpoints can provide better QoS by calculating accurate mouth-to-ear delays, and using this information to better inform buffer playout strategies. The IEEE 802.11e protocol extends 802.11 by providing different traffic priorities based on traffic type. However, it cannot distinguish between traffic streams within the same category. In this paper, we present an access point-centred mechanism for further distinguishing between streams within 802.11e categories. The mechanism predicts the VoIP quality for multiple sessions using the ITU-T E-Model. Delay values are determined from RTCP packet timing information. We provide implementation details on a real world proof-of-concept and present results that correlate well with previous NS-3 based simulations. In addition, we address scalability issues for large scale deployments. Copyright © 2015 John Wiley & Sons, Ltd.

This paper investigates the impact of frequent and small playout delay adjustments (time-shifting) of 30 ms or less introduced to silence periods by Voice over IP (VoIP) jitter buffer strategies on listening quality perceived by the end user. In particular, the quality impact is assessed using both a subjective method (quality scores obtained from subjective listening test) and an objective method based on perceptual modelling. Two different objective methods are used, PESQ (Perceptual Evaluation of Speech Quality, ITU-T Recommendation P.862) and POLQA (Perceptual Objective Listening Quality Assessment, ITU-T Recommendation P.863). Moreover, the relative accuracy of both objective models is assessed by comparing their predictions with subjective assessments. The results show that the impact of the investigated playout delay adjustments on subjective listening quality scores is negligible. On the other hand, a significant impact is reported for objective listening quality scores predicted by the PESQ model i.e. the PESQ model fails to correctly predict quality scores for this kind of degradation. Finally, the POLQA model is shown to perform significantly better than PESQ. We conclude the paper by identifying further related research that arises from this study.

In this chapter, we provide a comprehensive overview of timing. We describe the underlying concepts that comprise timing through examples and then present a range of mature, standardised and evolving techniques to improve the so-called time awareness across the full Information and Communications Technology (ICT) infrastructure over which multimedia applications operate. Although the media synchronisation community is already acutely aware of timing issues, this chapter offers some valuable insights through its holistic approach to timing.

Honeypots are deployed to capture cyber attack data for analysis of attacker behavior. This paper analyses a honeypot dataset to establish attack types and corresponding temporal patterns. It calculates the probability of each attack type occurring at a particular time of day and tests these probabilities with a random sample from the honeypot dataset. Attacks can take many forms and can come from different geographical sources. Temporal patterns in attacks are often observed due to the diurnal nature of computer usage and thus attack types captured on a honeypot will also reflect these patterns. We propose that it is possible to determine the probability of differing attack types occurring at certain times of the day. Understanding attack behavior informs the implementation of more robust security measures. The paper also proposes automating this process to create dynamic and adaptive honeypots. An adaptive honeypot that can modify its security levels, can increase the attack vector at different times of the day. This will improve data collection for analysis that ultimately will lead to better cyber defenses.

Wireless Personal Area Networks (WPAN) allow for the implementation of applications such as home automation, remote control services, near-field technologies and personal health care management. Security is a critical requirement of the standards and protocols for these environments. One suite of layered protocols within WPAN is ZigBee. ZigBee is a low bit rate protocol utilised in Wireless Sensor Networks (WSN). Attacks such as physical, crypto key interception, injection and replay are perpetrated on ZigBee networks. These attacks can be instigated and controlled within the physical ZigBee WSN location or via a gateway. This paper creates a honeypot that simulates a ZigBee gateway. It is designed to assess the presence of ZigBee attack intelligence on a SSH attack vector. It captures all attack traffic for retrospective analysis. It sandboxes attacks of interest to determine if any attempts are targeting ZigBee specifically. Finally it concludes that all captured mass attacks are mainstream DDoS and bot malware, whereas individual attackers where attracted to and interacted with the ZigBee simulated Honeypot.

A smart city system will contain diverse heterogeneous smart objects. Their complexity will range from simple reduced function devices (RFD) acting as common nodes, to full function devices (FFD) acting as coordinators and controlling actuators. As part of the Internet of Things, web facing devices can be remotely accessed for monitoring, control and data exchange. This makes them vulnerable to cyber attacks and compromise. To analyse such attacks, honeynets and honeypots are deployed to attract attackers and capture their activity for behavioural analysis. Designing honeynets is difficult due to the broad engineering scope of smart cities as a concept, and consequent diversity of smart object characteristics such as communication channels, interaction, data exchange and embedded security. This paper brings order to this diversity and scope by taking a data-centric view of smart city devices. The data-centric view assesses smart devices for their criticality, security and complexity. It presents a framework using this view, for adaptive honeynet development. It then validates the new framework by categorizing smart objects using the data centric view and applying them to the framework.

Time synchronisation plays a critical role in time-sensitive distributed applications. While a variety of such applications exist across many domains, one particular set of applications where improved time synchronisation can lead to significant benefits, particularly with respect to QoE (Quality of Experience), is multimedia applications. While time synchronisation is not a new challenge, advances in wireless technologies have drastically transformed network infrastructures. 802.11 wireless networks increasingly represent the last hop within the ever expanding Internet and whilst users expect the same levels of multimedia QoE as exist over wired networks, the reality of moving back to contention based access leaves many disappointed. This transformation of networks has also proven problematic for time synchronisation protocols that were designed for wired infrastructures. Wireless networks, particularly contention based networks, can be the source of very significant non-deterministic packet latencies. In certain scenarios, such latencies can greatly degrade the performance of time synchronisation. This work details and validates a technique that can be used to determine the latency of time messages in real-time as they traverse an 802.11 wireless link. Knowledge of these latencies can be used to greatly reduce the error in a dataset employed by time synchronisation protocols such as NTP and, thus, improve their performance. Experimental results confirm error reductions of up to 90% in a dataset and prove that the use of this technique can deliver time accuracies akin to those achievable over wired networks. This in turn can greatly benefit users by enabling multimedia applications to benefit from the continued use of time synchronisation for QoE management. We outline two such scenarios, one where time synchronisation is used to prioritise VoIP traffic within an Access Point and a second where the aim is to use time synchronisation to optimise jitter buffer strategies for WebRTC.

Over the last decade or so, significant research has focused on defining Quality of Experience (QoE) of Multimedia Systems and identifying the key factors that collectively determine it. Some consensus thus exists as to the role of System Factors, Human Factors and Context Factors. In this paper, the notion of context is broadened to include information gleaned from simultaneous out-of-band channels, such as social network trend analytics, that can be used if interpreted in a timely manner, to help further optimise QoE. A case study involving simulation of HTTP adaptive streaming (HAS) and load balancing in a content distribution network (CDN) in a flash crowd scenario is presented with encouraging results.

Time is central to predicting, measuring and controlling properties of the physical world, and is one of the most important constraints distinguishing Cyber-Physical Systems (CPS) from distributed computing in general. However, mixing the cyber and the physical presents a fundamental challenge, since computers and communications systems have abstracted away the physical layer and timing is fundamentally a physical signal. While such abstractions have yielded significant benefits, time has been a casualty. CPS used in industry today achieve time-awareness by making use of time-aware field-buses and devices with specialized proprietary software. However, this approach has proved restrictive in both the topologies achievable and the scalability of networks beyond a certain size. The new era of the Internet-of-Things and the Industrial Internet is paving the way for convergence, where time needs to be an integral part of the cyber, making integration of cyber and physical seamless. However, this requires successful research in a number of different areas. The National Institute of Standards and Technology (NIST) has formed a CPS Public Working Group (PWG), with members from global industry, academia and government. This CPS PWG is tasked with creating a set of frameworks and reference architectures for CPS, to promote proper function and interoperability. Public documents from this effort will soon be available. We discuss the timing section of the CPS PWG document and focus on the status of challenges and efforts to integrate time-sensitive with best-effort processes in CPS nodes and the networks that connect them.

Various approaches seek to optimize the quality of service of VoIP applications. We propose a system that uses synchronized time to combine the useful characteristics of both fixed and adaptive buffer strategies, thereby improving VoIP quality of service. Using a combination of global positioning system (GPS) technologies and the network time protocol (NTP), hosts can learn the precise end-to-end delay for each packet. This information can benefit both domestic and business Internet telephony users. We outline our proposed system and discuss issues arising from the use of synchronized time.

We expect that WebRTC will experience high adoption rate as a peer-to-peer realtime communication standard for browsers. WebRTC allows direct media and data transport between browsers without having to go through a web server. In this study, we use a black-box testing technique to evaluate, via PESQ, the voice quality of WebRTC sessions under varying network delay and jitter. Network emulators are employed to implement the delay and jitter variations. Our results highlight the dangers of black-box testing, whereby test-bed issues can result in very misleading results. This is especially the case when executed on a single machine. This paper also provides an extendable baseline methodology for WebRTC centric research.

WiFi-enabled mobile devices such as smartphones and tablets are being increasingly used by both personal and business users for Voice over IP (VoIP) communications, while also supporting conventional data applications such as email, file transfer and web access. Previous research has shown that time synchronized endpoints can provide better QoS by utilizing the Real-time Transport Control Protocol (RTCP) Sender and Receiver Reports (SR/RR) to calculate accurate Mouth to Ear (M2E) delays. The IEEE 802.11e protocol extends 802.11 by providing different traffic priorities based on traffic type. However it cannot distinguish between traffic streams within the same category. In this paper we propose an Access Point (AP)-centred mechanism for further distinguishing between streams within 802.11 categories. It is based on calculating E-Model R-factors for multiple VoIP sessions from delay values that are calculated using RTCP packet information. We detail a real world test-bed and present initial results that correlate well with a previous NS-3 based implementation.

A system based on Raman spectroscopy has been developed for closed-loop control during the manufacture of titanium dioxide (TiO2). Spectra were obtained in the 50 to 700 cm−1 range using excitation at ∼532 nm and a fixed transmission-grating spectrograph coupled to a CCD detector. Quantitation has been achieved by linear interpolation based on the integrated intensity of phonons at 142 and 610 cm−1 due to anatase and rutile phases, respectively. The spectroscopic instrumentation has been integrated with a powder sampling system and installed on four full-scale manufacturing plants. Fiber coupling (up to 100 m) has been employed, and each spectrometer system is multiplexed to three sampling heads. Spectra are corrected for changes in sample temperature, and the systems are designed to be fully automatic and require no user intervention.

In the existing IEEE 802.11e standard, all VoIP sessions contend within the same prioritization Access Category (AC), despite potentially having very different, and varying one-way (M2E - Mouth to Ear) delays. In this paper we show how VoIP endpoints that are time synchronized can help optimize 802.11e EDCA in order to prioritize VoIP sessions that have relatively large M2E delays and thus distinguish between VoIP sessions. Using the NS-3 Network Simulator, we quantify the benefits achievable through synchronization of an 802.11e network handling multiple VoIP calls in the presence of other TCP traffic.

WiFi-enabled mobile handheld devices are being increasingly used for Voice over IP (VoIP) as well as supporting conventional data applications such as email, file transfer and web access. Wireless LANs (WLANs) are thus increasingly required to support Quality of Service (QoS)-centric applications, which are delay sensitive and require a certain level of throughput. While 802.11e goes some way towards meeting this need, severe congestion leading to unacceptable delays and packet loss can still occur. In the existing 802.11e standard, all VoIP sessions contend within the same prioritization Access Category (AC), despite potentially having very different, and varying one-way (M2E - Mouth to Ear) delays. In this paper we provide a detailed analysis on simulations that demonstrate how VoIP endpoints that are time synchronized can help optimize 802.11e EDCA in order to prioritize VoIP sessions that have relatively large M2E delays and thus distinguish between VoIP sessions. Using the NS-3 Network Simulator, we quantify the benefits achievable through synchronization of an 802.11e network handling multiple VoIP calls in the presence of other TCP traffic. We present a heuristic EDCA tuning algorithm which uses the ITU-T E-Model R-Factor QoS planning tool as the basis of control.

This paper describes and validates an extension to the widely used Flooding Time Synchronisation Protocol (FTSP) namely Dynamic Flooding Time Synchronisation Protocol for Wireless Sensor Networks. The extension enhances FTSP by dynamically optimising communication exchange with the dual aim of conserving energy whilst improving accuracy. This optimisation is realised by altering the transmission rate of motes based on real-time monitoring and the WSN application accuracy requirements. The proposed mechanism, thus, eliminates the need with FTSP to determine and pre-configure a suitable transmission interval for a particular environment. Our results show that the proposed protocol extension can perform as well as FTSP in a stable temperature environment and can achieve similar accuracies with a much reduced (up to 75 percent) number of transmissions in an environment whereby motes are exposed to temperature fluctuations.

Hybrid broadcast and broadband TV (HbbTV) provides a unique system to unify various TV delivery systems, from Digital Video Broadcasting (DVB) to Internet Protocol TV (IPTV) and WebTV.

Government, business, and the general public increasingly agree that the polluter should pay. Carbon dioxide and environmental damage are considered viable chargeable commodities. The net effect of this for data center and cloud computing operators is that they should look to "chargeback" the environmental impacts of their services to the consuming end-users. An environmental chargeback model can have a positive effect on environmental impacts by linking consumers to the indirect impacts of their usage, facilitating clearer understanding of the impact of their actions. In this paper we motivate the need for environmental chargeback mechanisms. The environmental chargeback model is described including requirements, methodology for definition, and environmental impact allocation strategies. The paper details a proof-of-concept within an operational data center together with discussion on experiences gained and future research directions.

Watching a sport event via an IPTV channel and choosing a broadcast radio station as a play-out audio stream is the scenario discussed in this paper. Hybrid Broadcast and Broadband TV (hbbTV) provides an excellent platform to combine multimedia delivered via both systems. By following the recommendations in this paper, a synchronized play-out of multiple media streams delivered via the hbbTV system can be achieved.

User interactivity and customisation are key features that must be promoted to help distinguish IPTV from satellite/aerial/cable TV alternatives. Furthermore, IPTV providers also have to compete with Internet TV by providing extra features to justify the cost. In this paper, we present a number of scenarios whereby IPTV providers can achieve this through the use of synchronised time. Integrating and tightly synchronising multiple user-selected media streams in Real-Time that are logically and temporally related potentially adds significant extra value to any IPTV platform. It also raises significant research challenges. We outline the prototype developed for a TV/Text Feed scenario and present some details of ongoing and future work.

Delivering PSTN-like quality over current best-effort Internet infrastructure presents many technical challenges. Much research has focused on receiver-based approaches, which adapt to varying network conditions in order to optimize playout quality. In this paper, we propose and evaluate a receiver-based approach that implements a hybrid adaptive-fixed playout regime by integrating synchronized time into the playout algorithm. Such as approach can deliver significantly better quality than existing adaptive techniques particularly when the underlying network is not heavily congested and end-to-end delays are not excessive. We present some initial results form our testbed system using the ITU-T E-model to quantify improvement.

Previous work by the authors confirm the feasibility of implementing a delay-aware receiver buffering strategy based on synchronised time within VoIP applications. Our objective in this paper is to extensively test this delay-aware approach, termed the hybrid playout strategy over diverse networks using a combination of simulation and delay emulation. We utilise both measured delay data and derived delay models and quantify the significant performance gains through use of the ITU-T E-model. We also analyse recent Internet delay studies to further assess the hybrid’s wider applicability.

Packet delay, delay variation (jitter) and packet loss affect the performance of all Internet applications but particularly that of real-time multimedia. Developers of such applications or of multimedia delivery protocols need to be able to test them extensively before they are deployed on the Internet. We describe an architecture that combines both network analysis and network emulation. The network analysis module captures, stores and analyses network characteristics between two Internet hosts for the purpose of modeling. Models representing the network condition to be emulated can be constructed from this data or other sources by the network emulation module. This enables various Internet conditions to be recreated in a local controlled environment facilitating extensive testing of new applications and protocols.

The circuit switched POTS (plain old telephone system) preserves the timing relationship between media samples from sender to receiver through use of a common clock. For PC-based Internet multimedia, the existence of separate audio and system clocks on either end-host can introduce significant complications. Much work has taken place in recent years that addresses the issue of system clock skew and its effect on precise delay measurement. In a voice over IP (VoIP) environment, where adaptive buffering techniques are employed, system and audio clock skew can distort both delay measurement and playout control as well as lead to poor buffer performance. This paper presents a high level mechanism to measure and compensate for the skew relationships between system and audio clocks at each end of a multimedia session. The mechanism utilises both the network time protocol (NTP) and the RTP (realtime transport protocol) control protocol or RTCP. Preliminary and positive results are presented from a testbed system and plans for further work are outlined.