This thesis deals with the issues of detecting the codec used for the encoding of voice data carried by the RTP protocol without having access to the information carried by signalisation protocols in VoIP applications. Its main goal is to create and implement a fast algorithm for detecting the codec used for voice transfer via the RTP protocol. This algorithm should be fast enough to be used for offline analysis of captured data as well as for real-time online analysis. For research of possibilities were compared two approaches of detection. Detection by the characteristics of the codecs was chosen to solve the problem itself. Within the solution was performed data analysis and implementation of the application followed by testing on data.

Diplomová práce popisuje VoLTE službu, vývoj a nasazení LTE (zaváděcí fázi, skutečný LTE stav a výhledy do budoucna atd.), EPC-IMS architekturu (popis funkce uzlu, rozhraní atd.) Komunikace mezi uzly a funkce, rozhraní a protokoly jsou používány v průběhu signalizace (SIP SDP) a datový tok (RTCP RTP). Práce stručně popisuje základní toky hovorů, typy nosičů (GBR and N-GBR), a to vytvoření / mazaní nosičů během komunikace. Další část diplomové práce o implementaci volte, instalace a konfigurace IMS. Závěrečná část diplomové práce popisuje zkoušky sítě a, analýzu protokolu.

Diplomová práce popisuje VoLTE službu, vývoj a nasazení LTE (zaváděcí fázi, skutečný LTE stav a výhledy do budoucna atd.), EPC-IMS architekturu (popis funkce uzlu, rozhraní atd.) Komunikace mezi uzly a funkce, rozhraní a protokoly jsou používány v průběhu signalizace (SIP SDP) a datový tok (RTCP RTP). Práce stručně popisuje základní toky hovorů, typy nosičů (GBR and N-GBR), a to vytvoření / mazaní nosičů během komunikace. Další část diplomové práce o implementaci volte, instalace a konfigurace IMS. Závěrečná část diplomové práce popisuje zkoušky sítě a, analýzu protokolu.

During the last decade, the data rates provided by mobile networks have improved to the point that it is now feasible to provide richer services, such as streaming multimedia, to mobile users. However, due to factors such as radio interference and cell load, the throughput available to a client varies over time. If the throughput available to a client decreases below the media’s bit rate, the client’s buffer will eventually become empty. This causes the client to enter a period of rebuffering, which degrades user experience. In order to avoid this, a streaming server may provide the media at different bit rates, thereby allowing the media’s bit rate (and quality) to be modified to fit the client’s bandwidth. This is referred to as adaptive streaming. The aim of this thesis is to devise an algorithm to find the media quality most suitable for a specific client, focusing on how to detect that the user is able to receive content at a higher rate. The goal for such an algorithm is to avoid depleting the client buffer, while utilizing as much of the bandwidth available as possible without overflowing the buffers in the network. In particular, this thesis looks into the difficult problem of how to do adaptation for live content and how to switch to a content version with higher bitrate and quality in an optimal way. This thesis examines if existing adaptation mechanisms can be improved by considering the characteristics of different mobile networks. In order to achieve this, a study of mobile networks currently in use has been conducted, as well as experiments with streaming over live networks. The experiments and study indicate that the increased available throughput can not be detected by passive monitoring of client feedback. Furthermore, a higher data rate carrier will not be allocated to a client in 3G networks, unless the client is sufficiently utilizing the current carrier. This means that a streaming server must modify its sending rate in order to find its maximum throughput and to force allocation of a higher data rate carrier. Different methods for achieving this are examined and discussed and an algorithm based upon these ideas was implemented and evaluated. It is shown that increasing the transmission rate by introducing stuffed packets in the media stream allows the server to find the optimal bit rate for live video streams without switching up to a bit rate which the network can not support. This thesis was carried out during the summer and autumn of 2008 at Ericsson Research, Multimedia Technologies in Kista, Sweden. / Under det senaste decenniet har överföringshastigheterna i mobilnätet ökat så pass mycket att detnu är möjligt att erbjuda användarna mer avancerade tjänster, som till exempel strömmandemultimedia. I mobilnäten varierar dock klientens bandbredd med avseende på tiden på grund avfaktorer som störningar på radiolänken och lasten i cellen. Om en klients överföringshastighetsjunker till mindre än mediets bithastighet, kommer klientens buffert till slut att bli tom. Dettaleder till att klienten inleder en period av ombuffring, vilket försämrar användarupplevelsen. Föratt undvika detta kan en strömmande server erbjuda mediet i flera olika bithastigheter, vilket gördet möjligt för servern att anpassa bithastigheten (och därmed kvalitén) till klientens bandbredd.Denna metod kallas för adaptive strömning. Syftet för detta examensarbete är att utveckla en algoritm, som hittar den bithastighet som är bästlämpad för en specifik användare med fokus på att upptäcka att en klient kan ta emot media avhögre kvalité. Målet för en sådan algoritm är att undvika att klientens buffert blir tom ochsamtidigt utnyttja så mycket av bandbredden som möjligt utan att fylla nätverksbuffertarna. Merspecifikt undersöker denna rapport det svåra problemet med hur adaptering för direktsänd mediakan utföras. Examensarbetet undersöker om existerande adapteringsmekanismer kan förbättras genom attbeakta de olika radioteknologiers egenskaper. I detta arbete ingår både en studie avradioteknologier, som för tillfället används kommersiellt, samt experiment med strömmandemedia över dessa. Resultaten från studien och experimenten tyder på att ökad bandbredd inte kanupptäckas genom att passivt övervaka ”feedback” från klienten. Vidare kommer inte användarenatt allokeras en radiobärare med högre överföringshastighet i 3G-nätverk, om inte den nuvarandebäraren utnyttjas maximalt. Detta innebär att en strömmande server måste variera sinsändningshastighet både för att upptäcka om mer bandbredd är tillgänglig och för att framtvingaallokering av en bärare med högre hastighet. Olika metoder för att utföra detta undersöks ochdiskuteras och en algoritm baserad på dessa idéer utvecklas. Detta examensarbete utfördes under sommaren och hösten 2008 vid Ericsson Research,Multimedia Technologies i Kista, Sverige.

With its enormous potential Voice over Internet Protocol is one of the latest buzzwords in information technology. Despite the numerous advantages of Voice over IP, it is a major technical challenge to achieve a similar call quality as experienced in the ordinary Public Switched Telephone Network. This thesis introduces standardized Internet protocols for Voice over IP, such as Session Initiation Protocol (SIP), Real-time Transport Protocol (RTP), in its background chapter. In order to provide better Quality of Service (QoS) Voice over IP applications should support a feedback mechanism, such as the Real-time Control Protocol (RTCP), and use accurate timing information, provided by the Network Time Protocol (NTP). Additionally this thesis considers synchronization issues in calls with two and more peers. After a rather academic overview of Voice over IP, the open source real-time application “minisip”, a SIP user agent, and its operation and structure for handling audio streams will be introduced. Minisip was extended by an implementation of NTP and RTCP to provide a test platform for this thesis. A clear conclusion is that the addition of global time helps facilitate synchronization of multiple streams from clients located any where in the network and in addition the ability to make one-way delay measurements helps SIP user agents to provide better quality audio to their users. / Röst över IP, eller Internettelefoni baserad på “Voice over Internet Protocol” (VoIP), har med sin stora potential blivit ett av de senaste modeorden inom informationsteknologin. Vid sedan av ett antal fördelar med VoIP så innebär det en stor teknisk utmaning att uppnå en likadan samtalskvalitet som i det vanliga, fasta, telenätet. I den här uppsatsen beskrivs hur tjänstevalitet för VoIP kan förbättras genom att noggrant tidssynkronisera de (två eller flera) klienter som deltar i ett telefonsamtal. För detta krävs dels en återkopplingsmekanism, såsom “Real-time Control Protocol” (RTCP), samt en gemensam tidsuppfattning i de inblandade klienterna, vilket kan uppnås med hjälp av “Network Time Protocol” (NTP). Dessa protokoll, liksom de övriga Internet-standarder som VoIP baseras på (såsom “Session Initiation Protocol” (SIP) och “Real-time Transport Protocol” (RTP), beskrivs inledningsvis i uppsatsen. För studien har en SIP-klient baserad på öppen källkod använts (“Minisip”), och utökats med NTP och RCTP funktionalitet för att testa den föreslagna förbättringen av VoIP. En tydlig slutsats är att kännedom om en “global tid” möjliggör synkronisering av multipla ljudströmmar från klienter som befinner sig på olika nätverk. Möjligheten att mäta paketfördröjningen (envägs) bidrar också till en förbättrad ljudkvalitet.

<p>Nowadays, VoIP is getting more and more popular. It helps company to reduce cost, extends service to remote area, produce more service opportunities, etc. Besides these advantages, VoIP also put forward security problems.</p><p>In this paper, we introduce the popular protocols in VoIP and their security mechanisms, by introducing threats to VoIP, we point out the vulnerabilities with the security mechanisms of each VoIP protocol, and give recommendation for each VoIP protocol. In the conclusion part, we evaluate the vulnerabilities of each protocol, and point out in the future, with better protocol architecture, enhanced security policies, VoIP will has a brighter future.</p>

Nowadays, VoIP is getting more and more popular. It helps company to reduce cost, extends service to remote area, produce more service opportunities, etc. Besides these advantages, VoIP also put forward security problems. In this paper, we introduce the popular protocols in VoIP and their security mechanisms, by introducing threats to VoIP, we point out the vulnerabilities with the security mechanisms of each VoIP protocol, and give recommendation for each VoIP protocol. In the conclusion part, we evaluate the vulnerabilities of each protocol, and point out in the future, with better protocol architecture, enhanced security policies, VoIP will has a brighter future.

In IEEE 802.11 WLANs, Link Adaptation mechanisms, which choose the transmission rate of each node, provoke unexpected and random variations on the effective channel capacity. When these changes are towards lower bitrates, inelastic flows, such as VoIP, can suffer from sudden congestion, which results on higher packet delays and losses. In this thesis, a VoIP codec adaptation algorithm is proposed as a solution, based on a cross-layer feedback from RTCP packets and the MAC layer, which can adapt the codecs of active calls to adjust them to the multirate scenario. A combination of this algorithm with a call admission control mechanism is also studied. The results show an important improvement in terms of the QoS of the already active flows as also in the total hotspot's capacity. Additionally, by defining a new Grade of Service related parameter, the Q-Factor, which captures the trade-off between dropping and blocking ratio and perceived speech quality, the codec adaptation algorithm can be tuned to achieve maximum capacity without severely penalizing any of those variables, and hence satisfying both technical and user quality requirements. Finally, a new QoS-enabled AP, which implements these enhancements is designed. / En las redes inalámbricas del estándar IEEE 802.11, los mecanismos de adaptación de enlace que eligen la tasa de transmisión de cada nodo, pueden provocar variaciones aleatorias e inesperadas en la capacidad efectiva del canal. Cuando estos cambios son hacia tasas de transmisión mas bajas, los flujos inelásticos, tales como los de VoIP, pueden de repente sufrir congestión, lo que se traduce en aumento de retrasos y pérdidas de paquetes. En esa tesis, se propone un algoritmo de adaptación de codificadores de voz como solución, basado en técnicas multinivel (cross-layer) que combinan el uso de información de diferentes capas, como los paquetes RTCP y la capa MAC, y que puede adaptar los codecs de las llamadas activas para ajustarlos al escenario "multi-rate". Adicionalmente, la combinación de este algoritmo con un mecanismo de control de admisión de llamadas (CAC) se ha estudiado. Los resultados muestran una importante mejora en términos de QoS de los flujos activos como también en la capacidad total del hotspot. Además, mediante la definición de un nuevo factor, el Q-Factor, que puede captar la compensación entre la tasa de corte y de bloqueo de llamadas y de la calidad percibida por esas, el algoritmo de adaptación de codecs se puede ajustar para lograr la máxima capacidad sin penalizar severamente ninguna de esas variables y así satisfacer los requisitos técnicos de calidad y los usuarios. Por último, un nuevo punto de acceso (AP) habilitado para ofrecer calidad de servicio, ha sido diseñado que lleva a cabo estas mejoras.

The master’s thesis focuses on the virtual local computer network for laboratory usage. It aims to propose and realize proper network connection in order to monitor expected data flow. Thanks to the network analysers (software ClearSight and hardware NetTool Series II) it plans to pursue in detail the used transmission protocols of TCP/IP layers. The most decisive feature happens to be the right choice of appropriate network components and their precise configuration. Consequently, the thesis formulates a proposal of a laboratory task for the needs of students, which is also closely related to the actual problems. The assignment of the task will serve the teachers as a test pattern for measurement. The results elaborated in the form of the model protocol should enable later comparison of the recorded data. Another part of the diploma thesis is the working-out of well arranged manuals for the network analysers involved.

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