The success of the Internet has brought significant changes to the telecommunication industry. One of the remarkable outcomes of this evolution is Voice over IP (VoIP), which enables realtime voice communications over packet switched networks for a lower cost than traditional public switched telephone networks (PSTN). Nevertheless, security and privacy vulnerabilities pose a significant challenge to hindering VoIP from being widely deployed. The main object of this thesis is to define and elaborate unexplored security and privacy risks on standardized VoIP protocols and their implementations as well as to develop suitable countermeasures. Three research questions are addressed to achieve this objective: Question 1:  What are potential unexplored threats in a SIP VoIP network with regard to availability, confidentiality and privacy by means of unwanted traffic and information disclosure? Question 2:  How far are existing security and privacy mechanisms sufficient to counteract these threats and what are their shortcomings? Question 3:  How can new countermeasures be designed for minimizing or preventing the consequences caused by these threats efficiently in practice? Part I of the thesis concentrates on the threats caused by "unwanted traffic", which includes Denial of Service (DoS) attacks and voice spam. They generate unwanted traffic to consume the resources and annoy users. Part II of this thesis explores unauthorized information disclosure in VoIP traffic. Confidential user data such as calling records, identity information, PIN code and data revealing a user's social networks might be disclosed or partially disclosed from VoIP traffic. We studied both threats and countermeasures by conducting experiments or using theoretical assessment. Part II also presents a survey research related to threats and countermeasures for anonymous VoIP communication.

In recent years, a number of power control concepts have been studied and implementedeither in simulation or in practice for different communication systems. It is still the case that a great deal of research is being conducted within the area of energyefficient power control mechanisms for future wireless communication networksystems. However, only a limited amount of practical work has been implemented onreal test beds environment. The main goal of this thesis is to propose and develop newprototype Transmit Power Control Mechanisms (TPCM) on WARP (Wireless Open-Access Research Platform) boards for point-to-point communications, which are to bedeveloped and tested in an indoor environment. This work mainly focuses on the automaticpower control nodes, transmission and reception over-the-air. In this thesis, wehave designed and developed TPCM to adjust the power levels on a transmitter nodeby following the feedback (ACK) approach. In this case, the destination (receiver)node always sends the feedback (ACK) to transmitter node during every successfultransmission of message signal and the main focus is on a reduction in the packetloss rate (PLR), an increase in the packet reception rate (PRR) and the capacity ofthe nodes. In this real work, we have developed and measured the results based ontwo functions namely, with and without packet window function power control mechanisms. According to the measurements section, both with and without function powercontrol mechanisms proved to have better performances for different tunable parameters.If both functions are compared, then the with window function power controlmechanism was shown to produce better performances than the without windowpower control mechanism and it also converged faster than the without window function.If consideration was given to controlling a reduction in packet loss rate, thenthe with widnow function offered higher performances than those without the windowfunction. In this regard, it was found that the with window function has acheived amaximum packet reception rate than that for the without window function for differenttunable parameters. In relation to the power consumption scenario, it was determinedthat the without window fuction proved to produce energy saving performances thanthe with window function. There are several interesting aspects of the transmit powercontrol mechanisms highlighted in the results and discussion chapter.

Factum Electronics AB develops and manufactures equipment for Digital Audio Broadcasting (DAB). The increasing availability of IP networks has made the possibility to distribute the DAB signal, which is traditionally distributed over dedicated telecom links, over an IP backbone an interesting issue. This thesis investigates the possibilities to do this. Since distribution over IP is an unreliable service, it discusses techniques that can be used to eliminate the lack of reliability in IP. It also examines the advantages and drawbacks of using IP multicast, which would make it possible for many receivers to take part of the same signal. During the work behind this thesis, a protocol was developed including features to make the distribution more reliable. A prototype application including a client and a server was constructed to test and evaluate the protocol. Implementation issues of the protocol and test results drawn from the application are described in this thesis.

The purpose of this thesis is to examine the possibilities of creating a MPEG-4-compatible set-top box for IP-networks based on open standards. Existing alternatives for transporting MPEG-4 over IP are evaluated and ISMA is found to be an important actor within the area. ISMA is a non-profit corporation formed to provide a forum for the creation of specifications that define an interoperable implementation for streaming rich media over IP-networks. Two different designs based on ISMA's recommendation are constructed and evaluated. The designs have different levels of complexity and the more complex design is found to be better due to its extended functionality. During the design process a number of problems related to this kind of set-top box are discovered. It is believed, however, that many of these problems will be solved within the near future.

The interest for various multimedia services accessed via the Internet has been growing immensely along with the bandwidth available. A similar development has emerged in the 3G mobile network. The focus of this master thesis is on the speech/audio part of a 3G multimedia application. The purpose has been to implement a traffic generating tool that can measure QoS (Quality of Service) in 3G networks. The application is compliant to the 3G standards, i.e. it uses AMR (Adaptive Multi Rate), SIP (Session Initiation Protocol) and RTP (Real Time Transport Protocol). AMR is a speech compression algorithm with the special feature that it can compress speech into several different bitrates. SIP signalling is used so that different applications can agree on how to communicate. RTP carries the speech frames over the network, in order to provide features that are necessary for media/multimedia applications. Issues like perception of audio and QoS related parameters is also discussed, from the perspective of users and developers.

This thesis presents an intelligent early packet discard (I-EPD) for real-time video streaming over a multi-hop ad hoc network. In a multi-hop ad hoc network, the quality of transferring real-time video streams could be seriously degraded, since every intermediate node (IN) functionally like forwarding device does not possess large buffer and sufficient bandwidth. Even worse, a selected forwarding node could leave or power off unexpectedly which breaks the route to destination. Thus, a video packet temporarily buffered in intermediate nodes may exceed its time constraint when either a congested or failed link occurs; a stale video packet is useless even if it can reach destination after network traffic becomes smooth or failed route is reconfigured. In the proposed I-EPD, an IN can intelligently determine whether a buffered video packet should be discarded based on an estimated time constraint which is calculated from the RTP timestamps and the round trip time (RTT) measured by RTCP. For the purpose of validation, we implement the I-EPD scheme on a Linux-based embedded system. We compare the quality of video streams under different bit rates and different route repair time. In addition, we use PSNR to validate the quality of pictures from the aspect of application layer. The experimental results demonstrate that with I-EPD buffer utilization on IN can be more effectively used and unnecessary bandwidth wastage can be avoided.

<p>Examensarbetet bakom denna rapport utgör en undersökande studie om hur transmission av mediadata i nätverk kan göras effektivare. Det kan åstadkommas genom att viss tilläggsinformation avsedd för att jämna ut datatakten adderas i det realtidsprotokoll, Real Time Protocol, som används för strömmande media. Genom att försöka skicka lika mycket data under alla konsekutiva tidsintervall i sessionen kommer datatakten vid en godtycklig tidpunkt med större sannolikhet att vara densamma som vid tidigare klockslag. En streamingserver kan tolka, hantera och skicka data vidare enligt instruktionerna i protokollets sidhuvud. Datatakten jämnas ut genom att i förtid, under tidsintervall som innehåller mindre data, skicka även senare data i strömmen. Resultatet av detta är en utjämnad datataktskurva som i sin tur leder till en jämnare användning av nätverkskapaciteten.</p><p>Arbetet inkluderar en översiktlig analys av beteendet hos strömmande media, bakgrundsteori om filkonstruktion och nätverksteknologier samt ett förslag på hur mediafiler kan modifieras för att uppfylla syftet med examensarbetet. Resultat och diskussion kan förhoppningsvis användas som underlag för en framtida implementation av en applikation ämnad att förbättra trafikflöden över nätverk.</p>

<p>Factum Electronics AB develops and manufactures equipment for Digital Audio Broadcasting (DAB). The increasing availability of IP networks has made the possibility to distribute the DAB signal, which is traditionally distributed over dedicated telecom links, over an IP backbone an interesting issue. This thesis investigates the possibilities to do this. Since distribution over IP is an unreliable service, it discusses techniques that can be used to eliminate the lack of reliability in IP. It also examines the advantages and drawbacks of using IP multicast, which would make it possible for many receivers to take part of the same signal. </p><p>During the work behind this thesis, a protocol was developed including features to make the distribution more reliable. A prototype application including a client and a server was constructed to test and evaluate the protocol. Implementation issues of the protocol and test results drawn from the application are described in this thesis.</p>

In dieser Arbeit wird das Real-Time Streaming Protocol (RTSP) spwie das Real-Time Transport Protocol (RTP) analysiert. Die Syntax und Semantik von Präsentations-Beschreibungen im Format des Session Description Protocol (SDP) werden vorgestellt. Es wird eine kurze Einführung in das Format von MPEG-1-Videosequenzen gegeben. In der Analysephase wird der aktuelle Stand des Inline-MPEG-1-Players und des Multicast MPEG-Servers untersucht. Ausgehend davon werden die Ziele der Implementierung eines neuen Java-Applets sowie eines RTSP-Servers festgelegt und beschrieben, in welchen Schritten diese erreicht werden sollen. In der Implementierungsphase werden die Erstellung eines RTSP- und eines RTP-Klienten in Java beschrieben. Es wird dargelegt, wie unter Verwendung dieser Klienten auf Basis des Inline-MPEG-1-Players ein Java-Applet erzeugt wurde, daß MPEG-1-Videosequenzen von einem Medienserver empfangen und abspielen kann. Es wird der Entwurf eines RTSP-Medienservers und dessen Implementierung beschrieben. Mit Abschluß dieser Arbeit steht ein einsatzbereites System aus RTSP-Server und RTSP-Klient bereit, daß ohne weitere Klientsoftware auf einer Weboberfläche eingesetzt werden kann.

Eine kurze Einführung zu "Voice over IP" (dem Telefonieren über Datennetze). Es wird ein Überblick über technische Anforderungen und Lösungen geben. Behandelte Gebiete sind Audio-Codecs, das Transportprotokoll RTP sowie die Signalisierungsdienste SIP und H.323.