Det huvudsakliga syftet med denna C-uppsats är att undersöka hur teleoperatörer anpassar sin verksamhet för att bemöta den förändrade användningen av mobiltelefonen. Studien har undersökt hur operatörerna hanterar kunder som vill använda mobila tjänster istället för fasta. Mobiltelefonen används idag av teleoperatörens kunder för mer än bara samtal och textmeddelanden. För att besvara frågeformulering har intervjuer med fem teleoperatörer som tillsammans äger mer än 97 % av den svenska telemarknaden. Intervjuerna visar att teleoperatörer är mycket medvetna om de förändringar som har ägt rum och att de försöker vinna tid för att anpassa sin verksamhet, produkter och tjänster för den nya användningen av mobiltelefonen. Framtida abonnemang kommer att bli dyrare för att kompensera förlusten av intäkter från SMS och röstsamtal. Operatörernas kunder kommer ha möjlighet att ta med sig sina olika abonnemang som bl.a. tv-kanaler och bredband till sommarstugan och på så sätt undvika att betala för flera olika typer av abonnemang. / The main purpose of this C-level essay is to examine how telephone operators adapt their activities to address the changing use of the mobile phone. Also to examine how the phone operators deal with customers who want to use mobile services instead of fixed line services. Today, the mobile phone is used by customers for more than just calls and text messages. In order to answer our questionnaires we have chosen to interview five telephone operators who share more than 97% of the Swedish telecommunications market. The interviews with the five operators show us that they are aware of the changes that are taking place and that they are trying to gain time in order to adapt their products and services for the new uses of the mobile phone. Future subscriptions will be more expensive to compensate for the loss of revenue from text messaging and voice calls. The customer will be able to bring their TV channels and broadband with them to such places as the summer cottage and thus avoid having several different types of subscriptions.

Traditional public switched telephone networks (PSTN) are replaced more and more by VoIP services these days.  Although it is good for saving costs, the disadvantage of this development is that VoIP networks are less secure than the traditional  way of transmitting voice. Because VoIP networks are being deployed in open environments and rely on other network  services, the VoIP service itself becomes vulnerable to potential attacks against its infrastructure or other services  it relies on. This thesis will present a discussion of security issues of the Session Initiation Protocol (SIP), the signalling protocol for  VoIP services. The main focus is on active attacks against the protocol that aim to reduce the service's availability -- so called  Denial of Service (DoS) attacks. Existing countermeasures and detection schemes do not adequately differentiate between DoS attacks. However, the differentiation  is important with respect to performance loss, as various protection schemes involve more computationally intensive processes. Based on that discussion, this thesis attempts to provide an ontological approach to describing, and eventually preventing attacks from  having their intended effects.

Steganography is a technique which hides secret information. In this work, we will hide a secret information in the packets which are produced during a Skype call. Skype is one of the best known and the most widely used VoIP applications. We will propose, describe and implement a steganography method by which we will send the secret message during the Skype call. For embedding the message into packets and extracting them, we will use steganographic method called matrix encoding. To avoid packet loss, we will increase the robustness of this method by error-correcting and self-synchronising codes. As error-correcting codes, we will use the binary Hamming (7, 4) -codes and for the self-synchronising, we will use T-codes. 1

The multimedia domain is undergoing a rapid development phase with transition in audio, image, and video systems such as VoIP, Telepresence, Live/On-Demand Internet Streaming, SecondLife, and many more. In such a situation, the analysis of multimedia systems, like retrieval, quality evaluation, enhancement, summarization, and re-targeting applications, from various context is becoming critical. Current methods for solving the above-mentioned analysis problems do not consider the existence of humans and their affective characteristics in the design methodology. This contradicts the fact that most of the digital media is consumed only by the human end-users. We believe incorporating human feedback during the design and adaptation stage is key to the building process of multimedia systems. In this regard, we observe that affect is an important indicator of human perception and experience. This can be exploited in various ways for designing effective systems that will adapt more closely to the human response. We advocate an affect-based modeling approach for solving multimedia analysis problems by exploring new directions. In this dissertation, we select two representative multimedia analysis problems, e.g. Quality-of-Experience (QoE) evaluation and Image Enhancement in order to derive solutions based on affect-based modeling techniques. We formulate specific hypothesis for them by correlating system parameters to user's affective response, and investigate their roles under varying conditions for each respective scenario. We conducted extensive user studies based on human-to-human interaction through an audio conferencing system.We also conducted user studies based on affective enhancement of images and evaluated the effectiveness of our proposed approaches. Moving forward, multimedia systems will become more media-rich, interactive, and sophisticated and therefore effective solutions for quality, retrieval, and enhancement will be more challenging. Our work thus represents an important step towards the application of affect-based modeling techniques for the future generation of multimedia systems.

In this graduation thesis wants introduce possibilities of voice transportation in IP networks like Internet. There are described behaviors of IP network, internet connectivity possibilities, voice protocols included Skype and economical outputs. There are mapped positives and negatives of technology including implementation issues. All economic comparisons are made to Telefonica O2 Czech Republic Inc., which is still telecommunication company with majority in the Czech Republic. There are compared samples of VoIP providers with preferences like company live time, lowest cost and guaranteed quality. Description starts with model situation and continues to real case studies. Case studies describe economic calculation, implementation and target analysis. There are three case studies: small company with fast migration in one time, medium company witch accelerated migration in two steps and large company with conservative and control migration in three steps.

This diploma thesis deals with the implementation of ICT projects according to the third release of internationally respected IT infrastructure library (ITIL v3). To be specific, it is about two projects executed by the Renault company -- ToIP which deals with the voice transmission using the data network and S.P.T. concerning workstations' standardization. The objective of this thesis is to describe the implementation process of both projects into the business life and to evaluate the processes and outcomes based on the ITIL v3 recommendations. This goal will be delivered by studying available theoretical materials, authors personal work experience from the position of IT support in Renault and by confrontation praxis with theoretical knowledge. The main benefits of this work are the evaluation of the implementation process correctness according to the ITIL best practices, eventual deficiencies revealing and possible improvement propositions. This work is divided into theoretical and practical part. First, theoretical part is then divided into two captures -- the opening section describes the ITIL v3 best practices with the main focus on Service Transition and Service Operation. Second section is dedicated to the VoIP (voice over internet protocol) technology, part of which is also theoretical knowledge of voice transformation into the digital form, components of VoIP network and the most frequently used architectures. Second, practical part focuses on the projects` implementation. In conclusion the implementation process will be confronted with the ITIL v3 and based on detected failures possible improvements will be proposed.

The goal with this thesis work has been to identify what a telecommunications company should monitor and to find a network monitoring system that can monitor these identified objects on two different platforms: Windows and Linux. The network monitoring system has been implemented in a telecommunications company’s environment and this thesis presents how the system monitors their environment. The subject for this thesis work is within network monitoring. The problem formulation has been answered by conducting a literature study and by testing network monitoring systems’ features in a lab environment. The sources used in the literature study consists of scientific articles and other articles found on the web. The lab environment consisted of virtual machines that runs Linux or Windows as an operating system.   The purpose of the work was to enlighten Cellip in what objects they should monitor and to help the company to monitor them by implementing a network monitoring system. Cellip is a telecommunications company that provides IP-telephony services through Session Initiation Protocol. The limits of this thesis work are based on what their environment supports in terms of monitoring. Cellip’s environment consists of Linux and Windows servers, Cisco switches and firewalls, and Sonus Session Border Controllers.   In summary, the result of this thesis gives the reader information about what a telecommunications company with a similar environment to Cellip should monitor, what three systems that can monitor these objects, which of the three systems that has most automatized features and finally how the chosen system Datadog monitors and presents the objects. Some of the objects that are important to monitor is: memory, disk storage, latency, packet loss. In conclusion, this thesis presents a monitoring baseline for telecommunication companies with a similar environment to Cellip.

Telecommunications is converging on the use of IP based networks. Due to the low cost of VoIP applications, they are being increasingly used instead of conventional telephony services. IEEE802.11 WLANs are already widely used both commercially and domestically. VoIP applications will also expand from usage over wired networks to voice communications over IEEE802.11 WLANs. This is known as VoWLAN. The use of VoWLAN may reach the maximum capacity of a wireless channel if there are many simultaneous VoIP calls operating close to each other. There is much published research based on a single IEEE802.11 infrastructure WLAN concluding that packet loss, transmission efficiency and latency issues are the major challenges limiting the VoWLAN capacity. The VoIP service quality will drop sharply when the demands exceed the WLAN’s capacity. This thesis demonstrates that these challenges also apply to distributed WLANs. To extend these findings from the existing research, the analysis in this thesis indicates that the capacity of a single IEEE802.11 WLAN channel is 12 VoIP calls. When the number of simultaneous VoIP calls is within the capacity, the WLAN can deliver more than 90% of the voice packets to the receiver within 150 ms (the lowest network performance for supporting acceptable VoIP service). However, as soon as the traffic loads are beyond the wireless channel capacity e.g. the number of simultaneous VoIP calls is greater than 13, the VoIP service quality catastrophically collapses. When the capacity is exceeded there are almost no voice packets that can be delivered to the receiver within 150 ms. Our research results indicate that the delay accumulation for voice packets in the transmitter’s outgoing buffer causes this problem. Our research also found that dropping ‘stale’ voice packets that are already late for delivery to the receiver can give more transmission opportunities to those voice packets that may still be delivered in time. This thesis presents a new strategy called Active Cleaning Queue (ACQ) which actively drops ‘stale’ voice packets from the outgoing buffer and prevents the accumulation of delay in congested conditions. When ACQ is applied in a saturated wireless channel the network performance for supporting VoIP traffic was found to gradually decrease proportional to the numbers of simultaneous VoIP calls rather than catastrophically collapse. There is also published research suggesting that the aggregation of packets can improve the efficiency of WLAN transmissions. An algorithm called Small Packet Aggregation for Wireless Networks (SPAWN) is also presented in this thesis to improve transmission efficiency of small voice packets in WLANs without introducing further delay to VoIP traffic. The evaluation result shows that after applying the SPAWN algorithm, the VoIP capacity of a single wireless channel can be extended up to 24 simultaneous calls.

Voice over IP (VoIP) is a key enabling technology for the migration of circuit-switched PSTN architectures to packet-based IP networks. However, this migration is successful only if the present problems in IP networks are addressed before deploying VoIP infrastructure on a large scale. One of the important issues that the present VoIP networks face is the problem of unwanted calls commonly referred to as SPIT (spam over Internet telephony). Mostly, these SPIT calls are from unknown callers who broadcast unwanted calls. There may be unwanted calls from legitimate and known people too. In this case, the unwantedness depends on social proximity of the communicating parties. For detecting these unwanted calls, I propose a framework that analyzes incoming calls for unwanted behavior. The framework includes a VoIP spam detector (VSD) that analyzes incoming VoIP calls for spam behavior using trust and reputation techniques. The framework also includes a nuisance detector (ND) that proactively infers the nuisance (or reluctance of the end user) to receive incoming calls. This inference is based on past mutual behavior between the calling and the called party (i.e., caller and callee), the callee's presence (mood or state of mind) and tolerance in receiving voice calls from the caller, and the social closeness between the caller and the callee. The VSD and ND learn the behavior of callers over time and estimate the possibility of the call to be unwanted based on predetermined thresholds configured by the callee (or the filter administrators). These threshold values have to be automatically updated for integrating dynamic behavioral changes of the communicating parties. For updating these threshold values, I propose an automatic calibration mechanism using receiver operating characteristics curves (ROC). The VSD and ND use this mechanism for dynamically updating thresholds for optimizing their accuracy of detection. In addition to unwanted calls to the callees in a VoIP network, there can be unwanted traffic coming into a VoIP network that attempts to compromise VoIP network devices. Intelligent hackers can create malicious VoIP traffic for disrupting network activities. Hence, there is a need to frequently monitor the risk levels of critical network infrastructure. Towards realizing this objective, I describe a network level risk management mechanism that prioritizes resources in a VoIP network. The prioritization scheme involves an adaptive re-computation model of risk levels using attack graphs and Bayesian inference techniques. All the above techniques collectively account for a domain-level VoIP security solution.

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