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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.

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.

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.

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.

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.

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.

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.