Energy is an important factor that must be considered by multi-hop wireless mesh routing protocols because most sensors are powered by batteries with a limited capacity. We focus on the industry-standard RPL (Routing Protocol over Low-power and lossy networks) routing protocol that must find energy-efficient paths in low-power and lossy networks. However, the existing RPL objective functions route based on hop-count and ETX (expected transmission count) metrics alone, ignoring the energy cost of data transmission and reception. We address this issue in two ways. First, we design an objective function for RPL that finds paths that require, in expectation, the minimum amount of energy. Second, we design a probing mechanism which configures the transmission power of sensors to minimize energy consumption. The proposed approach is implemented and evaluated using simulations as well as on a small testbed with two Zolertial Z1 motes.

In the multiratecasting problem in wireless sensor networks, the source sensor is usually required to report to multiple destinations at dif- ferent rates for each of them. We present a MST-based rate-aware cost-efficient multiratecast routing protocol (MSTRC). The proposed MSTRC examines only one set partition of destinations at each for- warding step. A message split occurs when the locally-built minimum spanning tree (MST) over the current node and the set of destina- tions has multiple edges originated at the current node. Destinations spanned by each of these edges are grouped together, and for each of these subsets the best neighbor is selected as the next hop. We also suggested a novel face recovery mechanism to deal with void ar- eas, when no neighbor provides positive progress toward destinations. It constructs a MST of current node and destinations without the progress via neighbors, and for each set partition of destinations cor- responding to an edge e in MST, the face routing keeps going until a node that is closer to one of these destinations is found, allowing for greedy continuation, while the process repeats for the remaining desti- nations similarly. Our experimental results demonstrate that MSTRC is highly rate-efficient in all scenarios, and unlike existing solutions, it is adaptive to destination rate deviations.

We now live in an information society where increasingly large volumes of data are exchanged between networked nodes in distributed systems. Recent years have seen a multitude of dierent serialization frameworks released to efficiently handle all this information while minimizing developer effort. One such format is Google Protocol Buers, which has gained additional code generation support for a wide variety of programming languages from third-party developers. Ada is a widely used programming language in safety-critical systems today. However, it lacks support for Protocol Buffers. This limits the use of Protocol Buffers at companies like Saab, where Ada is the language of choice for many systems. To amend this situation Ada code generation support for Protocol Buffers has been developed. The developed solution supports a majority of Protocol Buffers' language constructs, extensions being a notable exception. To evaluate the developed solution, an artificial benchmark was constructed and a comparison was made with GNATColl.JSON. Although the benchmark was artificial, data used by the benchmark followed the same format as an existing radar system. The benchmark showed that if serialization performance is a limiting factor for the radar system, it could potentially receive a significant speed boost from a substitution of serialization framework. Results from the benchmark reveal that Protocol Buffers is about 6 to 8 times faster in a combined serialization/deserialization performance comparison. In addition, the change of serialization format has the added benefit of reducing size of serialized objects by approximately 45%.

Title page, table of contents and abstract only. The complete thesis in print form is available from the University of Adelaide Library. / Internet Protocol multicast has been standardised since the late 1980's, but is yet to be extensively deployed by most Internet Service Providers. Many organisations are not willing to bear the additional router CPU load and memory requirements that multicast entails, and the IP multicast suite of protocols requires deployment on every router spanned by the multicast group to operate. Additionally, these protocols are predominantly designed for the general case of multiple-source, multiple-receiver transmission and can be complex and inefficient to use in simpler scenarios. Single-source streaming of multimedia on the Internet is rapidly becoming a very popular application, and is predominantly being served by content providers using simultaneous unicast streams. A multicast transmission protocol designed for this application that can operate without requiring a widely deployed IP multicast infrastructure has the potential to save content-providers and network service providers significant amounts of bandwidth. This protocol should provide packet duplication and forwarding capabilities on routers in the network, rather than pushing this functionality to the receivers themselves, requiring them to become part of the multicast infrastructure. We describe Lorikeet, a new protocol for the multicast distribution of multimedia streams from a single source. This protocol builds its multicast tree from the source, discovering routers that support the protocol in the network and using them to provide branching in the tree. The tree itself is managed in a decentralised fashion, with joining receivers finding parent routers through a limited, recursive search of the tree. On a participating node, information about the tree's structure is limited to the addresses of that node's children and its path through the tree back to the source. Unlike most other multicast protocols, a new receiver is connected to the tree using its forward path from the source and packets are delivered through the tree via hop-by-hop delivery over unicast connections between nodes. Lorikeet also actively maintains the tree structure using a localised rearrangement algorithm triggered by a topological change in the tree structure. This rearrangement allows the tree to remain efficient in the face of changes to the receiver population, which can change the shape of the tree over time. Lorikeet is designed to operate with no further protocol support than that provided by existing Internet unicast protocols. It requires none of the standard IP multicast infrastructure, such as Class D group addressing. Its use of unicast connections between nodes allows it to be deployed incrementa.lly on the network, and its behaviour will degrade to simultaneous unicast when no routers that support the protocol are present at all. However, significant performance gains can be achieved even when there are only a few supporting routers present in the network: Lorikeet produces trees with half the cost of a unicast tree when just 10% of routers are Lorikeet-capable. Lorikeet's tree construction and rearrangement algorithms generate multicast trees of comparable total cost to those created by algorithms of considerably higher message complexity, such as those that employ exhaustive searches of the tree during joins. We develop the Lorikeet protocol from a set of requirements based on its target application and the properties of the current Internet. After describing the protocol's behaviour, we analyse its message complexity and its performance in terms of tree cost. We also analyse several other multicast protocols from the research literature, comparing their performance to that of Lorikeet in both complete deployment and incremental deployment scenarios. / http://proxy.library.adelaide.edu.au/login?url= http://library.adelaide.edu.au/cgi-bin/Pwebrecon.cgi?BBID=1283785 / Thesis (Ph.D.) -- University of Adelaide, School of Electrical and Electronic Engineering, 2007

Title page, table of contents and abstract only. The complete thesis in print form is available from the University of Adelaide Library. / Current best-effort packet-switched Internet is not a perfect environment for real-time applications such as transmitting voice-over the network (Voice Over Internet Protocol or VOIP). Due to the unlimited concurrent access to the Internet by users, the packet loss problem cannot be avoided. Therefore, the VOIP based applications encompass problems such as "voice quality degradation caused by lost packets". The effects of lost packets are fundamental issues in real-time voice transmission over the current unreliable Internet. The dropped packets have a negative impact on voice quality and concealing their effects at the receiver does not deal with all of the drop consequences. It has been observed that in a very lossy network, the receiver cannot cope with all the effects of lost packets and thereby the voice will have poor quality. At this point the Active Networks, a relatively new concept in networking, which allows users to execute a program on the packets in active nodes, can help VOIP regenerate the lost packets, and improve the quality of the received voice. Therefore, VOIP needs special voice-packing methods. Based on the measured packet loss rates, many new methods are introduced that can pack voice packets in such a way that the lost packets can be regenerated both within the network and at the receiver. The proposed voice-packing methods could help regenerate lost packets in the active nodes within the network to improve the perceptual quality of the received sound. The packing methods include schemes for packing samples from low and medium compressed sample-based codecs (PCM, ADPCM) and also include schemes for packing samples from high compressed frame-based codecs (G.729). Using these packing schemes, the received voice has good quality even under very high loss rates. Simulating a very lossy network using NS-2 and testing the regenerated voice quality by an audience showed that significant voice quality improvement is achievable by employing these packing schemes. / http://proxy.library.adelaide.edu.au/login?url= http://library.adelaide.edu.au/cgi-bin/Pwebrecon.cgi?BBID=1147315 / Thesis (Ph.D.) -- University of Adelaide, School of Electrical and Electronic Engineering, 2004

The field of chip design is characterized by contradictory pressures to reduce time-to-market and maintain a high level of reliability. As a result, module reuse has become common practice in chip design. To save time on both design and verification, Systems-on-Chips (SoCs) are composed using pre-designed and pre-verified modules. The integrated modules are often designed by different groups and for different purposes, and are later integrated into a single chip. In the absence of a single interface standard for such modules, "plug-n-play" style integration is not likely, as the subject modules are often designed to comply with different interface protocols. For such modules to communicate correctly there is a need for some glue logic, also called a protocol converter that mediates between them. Though much research has been dedicated to the protocol converter synthesis problem of SoC communication, converter synthesis is still performed manually, consuming development and verification time and risking human error. Current approaches to automatic synthesis of protocol converters mostly lack formal foundations and either employ abstractions far removed from the Hardware Description Language (HDL) implementation level or grossly simplify the structure of the protocols considered. This thesis develops and presents techniques for automatic synthesis of provably correct on-chip protocol converters. Basing the solution on a formal approach, a novel state-machine based formalism is presented for modelling bus-based protocols and formalizing the notions of protocol compatibility and correct protocol conversion. Algorithms for automatic compatibility checking and provably-correct converter synthesis are derived from the formalism, including a systematic exploration of the design space of the protocol converter, the first in the field, which enables generation of various alternative deterministic converters. The work presented is unique in its combination of a completely formal approach and the use of a low abstraction level that enables precise modelling of protocol characteristics and automatic translation of the constructed converter to HDL.

The field of chip design is characterized by contradictory pressures to reduce time-to-market and maintain a high level of reliability. As a result, module reuse has become common practice in chip design. To save time on both design and verification, Systems-on-Chips (SoCs) are composed using pre-designed and pre-verified modules. The integrated modules are often designed by different groups and for different purposes, and are later integrated into a single chip. In the absence of a single interface standard for such modules, "plug-n-play" style integration is not likely, as the subject modules are often designed to comply with different interface protocols. For such modules to communicate correctly there is a need for some glue logic, also called a protocol converter that mediates between them. Though much research has been dedicated to the protocol converter synthesis problem of SoC communication, converter synthesis is still performed manually, consuming development and verification time and risking human error. Current approaches to automatic synthesis of protocol converters mostly lack formal foundations and either employ abstractions far removed from the Hardware Description Language (HDL) implementation level or grossly simplify the structure of the protocols considered. This thesis develops and presents techniques for automatic synthesis of provably correct on-chip protocol converters. Basing the solution on a formal approach, a novel state-machine based formalism is presented for modelling bus-based protocols and formalizing the notions of protocol compatibility and correct protocol conversion. Algorithms for automatic compatibility checking and provably-correct converter synthesis are derived from the formalism, including a systematic exploration of the design space of the protocol converter, the first in the field, which enables generation of various alternative deterministic converters. The work presented is unique in its combination of a completely formal approach and the use of a low abstraction level that enables precise modelling of protocol characteristics and automatic translation of the constructed converter to HDL.

Doctor of Philosophy (PhD) / Up till now, architects collaborating with other colleagues did so mostly face-to-face (FTF). They had to be in the same space (co-located) at the same time. Communi-cation was ‘spontaneous’ and ideas were represented, whether verbal or non-verbal, by talking and using ‘traditional drawing tools’. If they were geographically displaced, the interaction was then space affected as well as the probability of being time affected. In this case communication was usually mediated through the tele-phone, and graphically represented ideas were sent by Fax or posted documents. Recently, some architectural firms started using modems and Internet connections to exchange information, by transferring CAD drawings as well as design informa-tion, through e-mail and file transfer protocol (FTP). Discussing ideas in architecture, as a more abstract notion, is different from discuss-ing other more concrete arguments using video conferencing. It is more important to ‘see’ what is being discussed at hand than ‘watch’ the other person(s) involved in the discussion. In other words the data being conveyed might be of more impor-tance than the mode of communication. Taking into consideration recent developments in computer and communication technologies this thesis investigates different communication channels utilised in architectural collaboration through Computer Mediated Collaborative Design (CMCD) sessions as opposed to FTF sessions. This thesis investigates the possi-ble effects these different channels have on collaborative design in general and col-laborative design communication in particular. We argue that successful CMCD does not necessarily mean emulating close prox-imity environments. Excluding certain communication channels in a CMCD envi-ronment might affect the flow and quantity of synchronous collaborative communica-tion, but not necessarily the quality and content of mutually communicated and rep-resented design ideas. Therefore different communication channels might affect the type of communication and not necessarily the content of the communication. We propose that audio and video are not essential communication channels in CMCD environments. We posit that architects will collaborate and communicate design representations effectively although with some differences, since those two chan-nels might cause interruptions and successful collaborative sessions can take place without them. For this purpose we conducted twenty-four one-hour experiments involving final year architecture students all working to the same design brief. The experiments were divided into three categories, FTF, full computer mediated collaborative design sessions (CMCD-a; audio-video conferencing plus whiteboard as a shared drawing space) and limited computer mediated collaborative design sessions (CMCD-b; with Lambda MOO used as a chat medium plus whiteboard as a shared drawing space). The experiments were video and audio taped, transcribed and coded into a custom developed coding scheme. The results of the analysed coded data and observations of the videotapes provided evidence that there were noticeable differences between the three categories. There was more design communication and less communication control in the CMCD-b category compared to the FTF and CMCD-a categories. Verbal communi-cation became shorter and straight to the point in CMCD-b as opposed to spontane-ous non-stop chat in the other two categories. Moreover in CMCD-b the subjects were observed to be more reflective as well as choosing and re-examining their words to explain ideas to their partners. At times they were seen scrolling back through the text of the conversation in order to re-analyse or interpret the design ideas at hand. This was impossible in FTF and CMCD-a sessions, since the sub-jects were more spontaneous and audio representations were lost as soon as they were uttered. Also the video channel in the CMCD-a category was ignored and hardly used except for the first few minutes of the experiments, for a brief exchange of light humour on the appearance of each subject. The results obtained from analysing the experiments helped us conclude that differ-ent communication channels produce different collaborative environments. The three categories of communication for architectural collaboration explored in our ex-periments are indicative of the alternatives available to architects now. What is not clear to architects is why they would choose one category over another. We pro-pose that each category has its own strengths and difficulties for architectural col-laboration, and therefore should be selected on the basis of the type of communica-tion considered to be most effective for the stage and tasks of the design project.

Thesis (Ph.D.)--Auburn University, 2005. / Abstract. Vita. Includes bibliographic references (ℓ. 98-99)

Thesis (M.App.Sc.) - Carleton University, 2003. / Includes bibliographical references (p. 90-92). Also available in electronic format on the Internet.