The point of this project is to cover a gap in today’s supply of provisioning systems for VoIP endpoint devices. Today the norm is that every manufacturer of endpoints builds their own system to provision their endpoints, but most operators use a vast verity of products to meet their customers’ needs. This system is based on open source software to allow adaptations to different environments and demands from customers and to require no license fee thus being affordable to the small size and midsized operators. This thesis is mainly an engineering type of thesis and the main bulk of the work has been put in developing the solution.

Voice over Internet Protocol (VoIP) has become very popular in recent days andbecome the first choice of small to medium companies for voice and data integration inorder to cut down the cost and use the IT resources in much more efficient way. Anotherpopular technology that is ruling the world after the year 2000 is 802.11 wirelessnetworks. The Organization wants to implement the VoIP on the wireless network. Thewireless medium has different nature and requirement than the 802.3 (Ethernet) andspecial consideration take into account while implementing the VoIP over wirelessnetwork.One of the major differences between 802.11 and 802.3 is the bandwidthavailability. When we implement the VoIP over 802.11, we must use the availablebandwidth is an efficient way that the VoIP application use as less bandwidth as possiblewhile retaining the good voice quality. In our project, we evaluated the differentcompression and decompression (CODEC) schemes over the wireless network for VoIP.To conduct this test we used two computers for comparing and evaluatingperformance between different CODEC. One dedicated system is used as Asterisk server,which is open source PBX software that is ready to use for main stream VoIPimplementation. Our main focus was on the end-to-end delay, jitter and packet loss forVoIP transmission for different CODECs under the different circumstances in thewireless network. The study also analyzed the VoIP codec selection based on the MeanOpinion Score (MOS) delivered by the softphone. In the end, we made a comparisonbetween all the proposed CODECs based on all the results and suggested the one Codecthat performs well in wireless network.

Denna studie tacklar ett av de problem som modern IP-telefoni brottas med, som gör att det är svårare att konkurrera med traditionell telefoni: fördröjningar på grund av otillräckliga Session Initiation Protocol-servrar (SIP). Genom tester av de viktigaste faktorerna jämförs fyra högaktuella Open Source SIP-servrar: OpenSIPS, Kamailio, FreeSWITCH och Yate. Avsikten är att underlätta valet av SIP-server för nya applikationer inom IP-telefoni, öka prestandan samt snabba på utvecklingen. Studien behandlar de intressantaste faktorerna vid val SIP, såsom användarvänlighet, hastighet, lagring av användardata samt ljudkvalité. Slutsatsen blev att Kamailio stod som klar segrare, med överlägsna resultat i jämförelse med övriga servrar, på de parametrar som valts ut. Skillnaderna var förhållandevis små prestandamässigt – det som verkligen avgjorde var främst hur avancerade servrarna var att installera, använda samt konfigurera.

Internet telephony has had a tremendous impact on how people communicate. Many now maintain contact using some form of Internet telephony. Therefore the motivation for this work has been to address the quality aspects of real-world Internet telephony for both fixed and wireless telecommunication. The focus has been on the quality aspects of voice communication, since poor quality leads often to user dissatisfaction. The scope of the work has been broad in order to address the main factors within IP-based voice communication. The first four chapters of this dissertation constitute the background material. The first chapter outlines where Internet telephony is deployed today. It also motivates the topics and techniques used in this research. The second chapter provides the background on Internet telephony including signalling, speech coding and voice Internetworking. The third chapter focuses solely on quality measures for packetised voice systems and finally the fourth chapter is devoted to the history of voice research. The appendix of this dissertation constitutes the research contributions. It includes an examination of the access network, focusing on how calls are multiplexed in wired and wireless systems. Subsequently in the wireless case, we consider how to handover calls from 802.11 networks to the cellular infrastructure. We then consider the Internet backbone where most of our work is devoted to measurements specifically for Internet telephony. The applications of these measurements have been estimating telephony arrival processes, measuring call quality, and quantifying the trend in Internet telephony quality over several years. We also consider the end systems, since they are responsible for reconstructing a voice stream given loss and delay constraints. Finally we estimate voice quality using the ITU proposal PESQ and the packet loss process. The main contribution of this work is a systematic examination of Internet telephony. We describe several methods to enable adaptable solutions for maintaining consistent voice quality. We have also found that relatively small technical changes can lead to substantial user quality improvements. A second contribution of this work is a suite of software tools designed to ascertain voice quality in IP networks. Some of these tools are in use within commercial systems today. / <p>Also identified as KTH publication TRITA-EE 2009:025</p> / Real-time router / Vinnova SIBED

Internet Protocol Telephony (IPT), also known as Voice over Internet Protocol (VoIP), has evolved from a niche technology to one that is adopted fairly well in developed countries. The aim of this research report was to determine whether IPT will also be a success in the Republic of South Africa, which is one of many developing countries. The technology was analysed and it was found that cost reduction, increased productivity and enhanced applications were the most valuable benefits the technology could offer. Particular interesting impediments of the technology were discovered and it was found that there existed both hard issues like security and quality problems, as well as softer issues like internal politics, that could hinder the global success of the technology. The adoption rate of South Africa was compared to that of industrial countries and it was found that South African organisations overall posed a wait-and-see attitude towards IPT. Various implementation models were discussed and it was found that a hybrid approach would be the most viable option for local organisations. The South African environment were analysed and it was discovered that the biggest obstacle for success in South Africa was the regulatory environment. But it was also found that the environment would soon change and that competitors, including Black Economic Empowerment companies, should use the opportunities available. Social factors like HIV/AIDS and theft as well as economic factors like the exchange rate could hamper the competitiveness of local companies using IPT. IPT technology can only be a success in South Africa if it is intensely supported by Government, implemented in the correct manner and adopted aggressively by the local market. / Dissertation (MBA)--University of Pretoria, 2006. / Graduate School of Management / unrestricted

The Voice over Internet Protocol (VoIP) is on its way to surpassing toll quality. Although VoIP shares its transmission channel with other communication traffic, today internet has a wider bandwidth than the legacy Digital Loop Carrier and voice could be digitized higher than traditional 8 kbps, to say 16 kbps. Thus, VoIP should not be limited by the toll quality. However, VoIP quality could go down, as a result of unpredictable traffic congestion and network imperfections. These two situations cause delay jitter and packet loss of VoIP. To overcome these challenges, there are ongoing works for service providers including but not limited to optimizing routing and adding more bandwidth. There are also works by developers at the user’s end, which includes compressing voice packet size and processing playout delay adapted to the network condition. While VoIP planning or off-line quality monitoring and control use overall quality measurements such as mean opinion score (MOS) or R-factor, the real-time quality supervision typically uses the network condition factors only. The control mechanism that is based on network quality could adjust the channel parameter by changing Codec and its parameters, and changing playout delay, etc. to minimize the loss of voice quality. As bandwidth plays a prominent role in IP traffic congestion, compressing the packet header is a possible solution to minimize congestion. Replacing a completed packet header with a smaller header will significantly reduce the packet header size. For instance, with a context, a compressed header will not consist of RTP header and, thus, could reduce 16 bytes from each packet. However, the primary question is how to deal with delay jitter calculation without time stamping. In this research, a delay jitter calculation for VoIP packet without timestamp has been provided. Compressing payload or using high compressing Codecs, is another major solution for preventing quality downgrade with limited bandwidth. The challenge with many Codec and the tradeoff between Codec quality and packet loss due to limited bandwidth has been addressed in this research with a summary of Codec quality evaluation and a bandwidth planning calculation. Although the E-model and its R-factor has been proposed by the International Telecommunication Union (ITU) for VoIP quality measurement, with many network and Codec parameters, it could only be used for offline quality control. Since accessing a live traffic for monitoring live quality is somewhat impossible, at the client side, only packet loss and delay jitter matters. In this research, more in-depth investigation of adaptive playout delay based on jitter prediction has been carried out and recommended as the end user solution for quality improvement. An adaptive playout delay based on Markov model also has been developed in detail and tested with real VoIP network. This development has closed the gap between research and engineering. Therefore, the Markov model could be evaluated and implemented.

Komunikace je v dnešní době základním klíčem k úspěchu. Rychlost, dostupnost a kvalita předávaných informací je schopna ovlivnit nejen budoucnost firem, ale také jedinců. Cílem práce je poskytnout ucelený přehled o dostupných IT a SW prostředcích pro firemní komunikaci. E-mail, hlasová komunikace, videokonference, portály, ale také virtuální prostředí jsou postaveny proti klasickým formám komunikace jako je meeting a korespondence. V práci je vytvořena metodika umožňující srovnání jednotlivých forem komunikace z pohledu jejich vlastností, vhodností jejich použití pro různé modelové typy komunikace a možnosti jejich nasazení v interní a externí firemní komunikaci. Poslední kapitola práce demonstruje možnost aplikace zjištěných závěrů a získaných poznatků na středně velkou stavební firmu.

The diploma thesis is focused on exploring the possibility of VoIP service in a wireless network of University of Economics, Prague. This thesis describes the basic principles of VoIP and related wireless technologies necessary for its quality and stable operation. Subsequently, different configurations of wireless network and end clients are tested and compared, including its impact on ordinary users, in a laboratory environment with idle and fully utilized frequency band. Finally, a roaming operation with the use of several advanced 802.11 standards is tested in the real environment of the Old building in Žižkov. In conclusion, the ideal settings for all telecommunication devices are recommended in order to maximize the quality of VoIP operation and to minimize the negative impact on ordinary users.

The following thesis aims to describe the transformation of telco operators' business strategy evoked by the growing popularity and usage of Internet services like WhatsApp, Skype and Netflix, also known as over-the-top services. The thesis provides comprehensive overview of transformations in the telecom industry that have been taking place globally, explaining the drivers that led to inception of free or cheaper Internet-based services, their business models. It then narrows down to the thorough analysis of the Czech market, capturing the implications that over-the-top services have on business strategy of the major Czech telco players. The analysis is strongly supported by industry reports, statistics and quantitative research. The primary research adds in telco customers' perspectives, usage and value perception of over-the-top services, as well as their satisfaction with respective telco services and their pricing. Derived findings identify major developments and trends in the Czech telco market; they also suggest compelling observations for related businesses.

More and more sensitive information is communicated digitally and with thatcomes the demand for security and privacy on the services being used. An accurateQoS metric for these services are of interest both for the customer and theservice provider. This thesis has investigated the impact of different parameterson the perceived voice quality for encrypted VoIP using a PESQ score as referencevalue. Based on this investigation a parametric prediction model has been developedwhich outputs a R-value, comparable to that of the widely used E-modelfrom ITU. This thesis can further be seen as a template for how to construct modelsof other equipments or codecs than those evaluated here since they effect theresult but are hard to parametrise. The results of the investigation are consistent with previous studies regarding theimpact of packet loss, the impact of jitter is shown to be significant over 40 ms.The results from three different packetizers are presented which illustrates theneed to take such aspects into consideration when constructing a model to predictvoice quality. The model derived from the investigation performs well withno mean error and a standard deviation of the error of a mere 1:45 R-value unitswhen validated in conditions to be expected in GSM networks. When validatedagainst an emulated 3G network the standard deviation is even lower.v