<p>With the Internet expansion, real-time transmission over Internet is becoming a new promising application. Successful real-time communication over IP networks requires reasonably reliable, low delay, low loss date transport. Since Internet is a non-synchronous packet switching network, high load and lack of guarantees on data delivery make real-time communication such as Voice and Video over IP a challenging application to become realistic on the Internet. </p><p>This thesis work is composed of two parts within real-time voice and video communication: network simulation and measurement on the real Internet. In the network simulation, I investigate the requirement for the network"overprovisioning"in order to reach certain quality-of-service. In the experiments on the real Internet, I simulate real-time transmission with UDP packets along two different traffic routes and analyze the quality-of- service I get in each case. </p><p>The overall contribution of this work is: To create scenarios to understand the concept of overprovisioning and how it affects the quality-of-service. To develop a mechanism to measure the quality-of-service for real-time traffic provided by the current best-effort network.</p>

<p>This thesis evaluates and shows an implementation of a video telephony solution for network connected set-top boxes based on the SIP protocol for managing sessions. </p><p>Unlike other video telephony implementations the set-top box does not handle both audio and video, but only video. A separate phone is used to handle audio. To maintain compatibility with other video telephony implementations, which expect a single SIP device with both audio and video capabilities, a mechanism to merge the audio (SIP-phone) and video (set-top box) into a single entity was developed using a back-to-back user agent. </p><p>Due to the set-top boxes'limited hardware it could be impossible to have video compression and decompression performed by the set-top boxes. However, numerous performance tests of compression algorithms showed that the computational power available in the set-top boxes is sufficient to have acceptable frame rate and image quality in a video telephony session. A faster CPU or dedicated hardware for video compression and decompression would however be required in order to compete with dedicated video telephony systems available today. </p><p>The implemented video telephony system is based on open standards such as SIP, RTP and H.261, which means interoperability with other video telephony implementations, such as Microsoft's Windows Messenger 4.7, is good.</p>

<p>POTS (Plain Old Telephony Service) systems have traditionally been connected via synchronous connections. When installing new nodes in the telephone network, they may sometimes be connected via packet networks such as Ethernet. Ethernet is an asynchronous network which means that nodes connected to the network don’t have access to the same clock frequency if it is not provided in some other way. If two nodes have different clock frequency, the receiver’s buffer will eventually overflow or starve. While not being a severe problem for telephony, devices used for data transmission, e.g. modems and fax will not be able to function properly. To avoid this it is necessary to synchronize the nodes.</p><p>This thesis investigates methods to synchronize nodes connected over Ethernet by simulating them in Matlab. The simulations show that under certain circumstances it is possible to produce a clock signal conforming to relevant standards.</p>

This dissertation relates and analysis the development of the Swedish mobile telephone industry from 1950-98. Based on a research of entrepreneurial theories three forms of entrepreneurship are formulated, the individual as entrepreneur, the company as entrepreneur, and the network as entrepreneur. The mobile telephone industry is seen as a technological system incorporating a broad spectrum of participants, and the account analysis how the entrepreneurs carry out different types of innovations. It starts out as a technical project, with market related innovations becoming a greater force in the final phase. The source of entrepreneurship varies over time and the individual plays a significant role early on while the network becomes of a greater importance as the system develops. / Diss. Stockholm : Handelshögsk.

With the Internet expansion, real-time transmission over Internet is becoming a new promising application. Successful real-time communication over IP networks requires reasonably reliable, low delay, low loss date transport. Since Internet is a non-synchronous packet switching network, high load and lack of guarantees on data delivery make real-time communication such as Voice and Video over IP a challenging application to become realistic on the Internet. This thesis work is composed of two parts within real-time voice and video communication: network simulation and measurement on the real Internet. In the network simulation, I investigate the requirement for the network"overprovisioning"in order to reach certain quality-of-service. In the experiments on the real Internet, I simulate real-time transmission with UDP packets along two different traffic routes and analyze the quality-of- service I get in each case. The overall contribution of this work is: To create scenarios to understand the concept of overprovisioning and how it affects the quality-of-service. To develop a mechanism to measure the quality-of-service for real-time traffic provided by the current best-effort network.

POTS (Plain Old Telephony Service) systems have traditionally been connected via synchronous connections. When installing new nodes in the telephone network, they may sometimes be connected via packet networks such as Ethernet. Ethernet is an asynchronous network which means that nodes connected to the network don’t have access to the same clock frequency if it is not provided in some other way. If two nodes have different clock frequency, the receiver’s buffer will eventually overflow or starve. While not being a severe problem for telephony, devices used for data transmission, e.g. modems and fax will not be able to function properly. To avoid this it is necessary to synchronize the nodes. This thesis investigates methods to synchronize nodes connected over Ethernet by simulating them in Matlab. The simulations show that under certain circumstances it is possible to produce a clock signal conforming to relevant standards.

Telecommunication systems have evolved significantly and the recent convergence of telephony allows users to communicate through landlines, mobile phones and Voice over IP (VoIP) phones. Unfortunately this convergence has resulted in easy manipulation of caller identity, resulting in both VoIP spam and Caller ID spoofing. In this dissertation, we introduce the notion of effective identity which is a combination of mechanisms to (1) establish identity of the caller that is harder to manipulate, and (2) provide additional information about the caller. We first use effective identities to address the VoIP spam problem by proposing CallRank, a novel mechanism built around call duration and social network linkages to differentiate between a legitimate user and a spammer. To ensure that this mechanism is privacy preserving, we create a token framework that allows a user to prove the existence of a social network path between him and the user he is trying to initiate contact with, without actually revealing the path. We then look at the broader issue of determining identity across the entire telecommunication landscape to address Caller ID spoofing. Towards this, we develop PinDr0p, a technique to determine the provenance of a call - the source and the path taken by a call. In the absence of any verifiable metadata, provenance offers a means of uniquely identifying a call source. Finally, we use anomalies in timbre to develop London Calling, a mechanism to identify geography of a caller. Together, the contributions made in this dissertation create effective identities that can help address the new threats in a converged telecommunication infrastructure.

This collection of published scholarly work concerns the growth and development of community radio and mobile telephony during the period 2000-2009, with reference to specific examples and case studies. The associated report examines the extent that the public are using these media to the benefit of individuals and communities and also suggests that the public have regained use of the electromagnetic spectrum through the use of community radio and mobile phones. The over arching hypothesis is that both community radio and mobile telephony are: • providing benefits to society, rangmg from improving daily life and avoiding inconveniences, to ameliorating critical or life-threatening situations and resisting oppression, • re-establishing rights of usage in the electromagnetic spectrum for ordinary members of the public which were surrendered to governmental authorities in the early years of spectrum experimentation, • using the electromagnetic spectrum as a tool for activism, political discussion, social engagement and exchange of information about matters of common interest, • worthy of research and examination to investigate how these two forms of communications media are impacting on the lives of individuals and society as a whole.

Network Address Translation (NAT) devices are common, particularly in the small-offiee/home-office environment. Typically, a NAT device does not disrupt the oper-ation of a client-server application. even if the application is unaware of the device's presence. Unfortunately, the same does not hold for peer-to-peer applications. such as Voice over IP (VoIP). This thesis investigates the compatibility of VoIP and NAT by making VoIP calls across various NAT device types. Three different test methods are used. differing in the VoIP phone that is used and the degree of automation that is employed. Moreover, the VoIP phones are tested in different configurations, which results in significantly altered traffic and vastly different test results. All test results are presented in detail. along with explanations as to why specific phone/NAT-type combinations succeed or fail.

Location-based service (LBS) is a geographical location-related service that provides highly personalized services for users. It is a platform for network operators to provide new and innovative ways of increasing profits from new services. With the rapidly growing trend toward LBS, there is a need for standard LBS protocols. This thesis started with introducing the Internet Engineering Task Force GEOPRIV working group, which endeavors to provide standard LBS protocols capable of transferring geographic location information for diverse location-aware applications. Through careful observation, it was found that Session Initiation Protocol (SIP) is well suited to the GEOPRIV requirements. The aim of this research was therefore to explore the possibility of the integration of LBS and the SIP protocol and, to some extent fulfill the GEOPRIV requirements.