Nowadays, we are witnessing a transition from physical togetherness towards networked togetherness around media content. Novel forms of shared media experiences are gaining momentum, allowing geographically distributed users to concurrently consume the same media content while socially interacting (e.g., via text, audio or video chat). Relevant use cases are, for example, Social TV, networked games and multi-party conferencing. However, realizing enjoyable shared media services faces many challenges. In particular, a key technological enabler is the concurrent synchronization of the media playout across multiple locations, which is known as Inter-Destination Multimedia Synchronization (IDMS). This PhD thesis presents an inter-operable, adaptive and accurate IDMS solution, based on extending the capabilities of RTP/RTCP standard protocols (RFC 3550). Concretely, two new RTCP messages for IDMS have been defined to carry out the necessary information to achieve IDMS. Such RTCP extensions have been standardized within the IETF, in RFC 7272. In addition, novel standard-compliant Early Event-Driven (EED) RTCP feedback reporting mechanisms have been also designed to enhance the performance in terms of interactivity, flexibility, dynamism and accuracy when performing IDMS. The designed IDMS solution makes use of globally synchronized clocks (e.g., using NTP) and can adopt different (centralized and distributed) architectural schemes to exchange the RTCP messages for IDMS. This allows efficiently providing IDMS in a variety of networked scenarios and applications, with different requirements (e.g., interactivity, scalability, robustness…) and available resources (e.g., bandwidth, latency, multicast support…). Likewise, various monitoring and control algorithms, such as dynamic strategies for selecting the reference timing to synchronize with, and fault tolerance mechanisms, have been added. Moreover, the proposed IDMS solution includes a novel Adaptive Media Playout (AMP) technique, which aims to smoothly adjust the media playout rate, within perceptually tolerable ranges, every time an asynchrony threshold is exceeded. Prototypes of the IDMS solution have been implemented in both a simulation and in real media framework. The evaluation tests prove the consistent behavior and the satisfactory performance of each one of the designed components (e.g.,protocols, architectural schemes, master selection policies, adjustment techniques…). Likewise, comparison results between the different developed alternatives for such components are also provided. In general, the obtained results demonstrate the ability of this RTP/RTCP-based IDMS solution to concurrently and independently maintain an overall synchronization status (within allowable limits) in different logical groups of users, while avoiding annoying playout discontinuities and hardly increasing the computation and traffic load. / Montagut Climent, MA. (2015). DESIGN, DEVELOPMENT AND EVALUATION OF AN ADAPTIVE AND STANDARDIZED RTP/RTCP-BASED IDMS SOLUTION [Tesis doctoral]. Universitat Politècnica de València. https://doi.org/10.4995/Thesis/10251/48549 / TESIS / Premios Extraordinarios de tesis doctorales

This diploma thesis contains questions of simulation data transfer by ASM multicast. In simulation tool Opnet Modeler is proceed design of service IPTV. IPTV means television which is transfered in network by IP protocol. Data of IPTV service are sending by multicast transfer. Multicast is a technology which uses a group transfer. It is actually communication of a one source of data with many users. These users are receiving the same data. A main target of this technology is to decrement loading of source node and transference system in distribution of data towards group of users. Most often is multicast used in distribution of audiovisual data. Relation RTP/RTCP is simulated with a different numbers of users. Observed is interval of transmission of control RTCP packets. This will be reached by two methods which will be confront in the end. One is a theoretic calculation by course of a equation and second is a practical simulation in Opnet Modeler.

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.