Audio networking is a rapidly increasing field which introduces new exiting possibilities for the professional audio industry. When well established, it will drastically change the way live sound systems will be designed, built and used. Today's networks have enough bandwidth that enables them to transfer hundreds of high quality audio channels, replacing analogue cables and intricate installations of conventional analogue audio systems. Currently there are many systems in the market that distribute audio over networks for live music and studio applications, but this technology is not yet widespread. The reasons that audio networks are not as popular as it was expected are mainly the lack of interoperability between different vendors and still, the need of a wired network infrastructure. Therefore, the development of a wireless digital audio networking system based on the existing widespread wireless technology is a major research challenge. However, the ΙΕΕΕ 802.11 standard, which is the primary wireless networking technology today, appears to be unable to handle this type of application despite the large bandwidth available. Apart from the well-known drawbacks of interference and security, encountered in all wireless data transmission systems, the way that ΙΕΕΕ 802.11 arbitrates the wireless channel access causes significantly high collision rate, low throughput and long overall delay. The aim of this research was to identify the causes that impede this technology to support real time wireless audio networks and to propose possible solutions. Initially the standard was tested thoroughly using a data traffic model which emulates a multi-channel real time audio environment. Broadcasting was found to be the optimal communication method, in order to satisfy the intolerance of live audio, when it comes to delay. The results were analysed and the drawback was identified in the hereditary weakness of the IEEE 802.11 standard to manage broadcasting, from multiple sources in the same network. To resolve this, a series of modifications was proposed for the Medium Access Control algorithm of the standard. First, the extended use of the "CTS-to-Self" control message was introduced in order to act as a protection mechanism in broadcasting, similar to the RTC/CTS protection mechanism, already used in unicast transmission. Then, an alternative "random backoff" method was proposed taking into account the characteristics of live audio wireless networks. For this method a novel "Exclusive Backoff Number Allocation" (EBNA) algorithm was designed aiming to minimize collisions. The results showed that significant improvement in throughput can be achieved using the above modifications but further improvement was needed, when it comes to delay, in order to reach the internationally accepted standards for real time audio delivery. Thus, a traffic adaptive version of the EBNA algorithm was designed. This algorithm monitors the traffic in the network, calculates the probability of collision and accordingly switches between classic IEEE 802.11 MAC and EBNA which is applied only between active stations, rather than to all stations in the network. All amendments were designed to operate as an alternative mode of the existing technology rather as an independent proprietary system. For this reason interoperability with classic IEEE 802.11 was also tested and analysed at the last part of this research. The results showed that the IEEE 802.11 standard, suitably modified, is able to support multiple broadcasting transmission and therefore it can be the platform upon which, the future wireless audio networks will be developed.

This study investigates Audio over IP. A stress test was used to see what impact asymmetric latency had on the audio signal in a closed network. The study was constructed into two parts. The first part is the stress test where two AoIP solutions were tested. The two solutions where exposed in two forms of asymmetric latency. First a fixed value was used, next, a custom script was used to simulate changing values of asymmetric latency. The second part of this study involved interviews that where conducted with representatives from the audio industry that are working with audio over IP on a dayto-day usage. The goal for these interviews was to figure out what knowledge the audio industry had about asymmetric latency, if the industry had experienced problems related to latency and what general knowledge the industry has about networks. It was found in the interviews that the limitation in AoIP isn’t the technology in itself but rather missing knowledge with the people that are using the systems.