Multimedia streaming is one of the killer applications for cellular communications. Although the 3GPP proposes the Packet-Switched Streaming (PSS) protocol to support the multimedia streaming services, the performances are still not good enough. To improve the performances, most of the researches divide the cellular networks into two parts: the wired network and the wireless network, and focus on the wireless network part. Hence the adaptive streaming was proposed. It utilizes the RTCP feedback in RTP to monitor the wireless network, and makes appropriate transmission parameters adjustments to prevent from buffer underflow and packet loss. But the overall performance may not be only limited by the wireless network part. Also, with high mobility, the link quality may be influenced severely by many factors. Hence, we will evaluate the performance of streaming services over 3G cellular networks on the train of Taiwan high speed railway in this paper and propose a mechanism to improve the performance.

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

Voice over IP (VoIP) is a real time application that allows transmitting voice through the Internet network. Recently there has been amazing progress in this field, mainly due to the development of voice codecs that react appropriately under conditions of packet loss, and the improvement of intelligent jitter buffers that perform better under conditions of variable inter packet delay. In addition, there are other factors that indirectly benefited VoIP. Today, computer networks are faster due to the advances in hardware and breakthrough algorithms. As a result, the quality of VoIP calls has improved considerably. However, the quality of VoIP calls under extreme conditions of packet loss still remains a major problem that needs to be addressed for the next generation of VoIP services. This thesis concentrates in making an analysis of the effects that network impairments, such as: delay, jitter, and packet loss have in the quality of VoIP calls and approaches to solve this problem. Finally, we analyze the impact of introducing forward error correction (FEC) Piggybacking and Reed Solomon codes for VoIP. To measure the mean opinion score of VoIP calls we develop an application based on the E-Model, and utilize perceptual evaluation of speech quality (PESQ).

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.

<p>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.</p>

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.

Thesis deals with multicast broadcasts, discusses the type of ASM and SSM. Also deals with the principle of RTP / RTCP protocol for large multicast groups, for example, broadcasting IPTV. It was created a model for simulating the behavior of the network load in a large number of receivers. The application was developed in Java and uses the recommendations of RFC3550. There are also analysed processes of communication, initialization, log out, the structure of the data packets and signaling packets. The system is designed that can simulate multicast network and obtain the estimated parameters of the operation, or can operate in real traffic, for example, as a third-party monitor. In both modes, the measurement was carried out model situations. The results of the measurements were then compared. Principle and system control was described in detail. Output measurements are graphically processed and included in the work.

The master’s thesis deals with creation of system that provides video on demand. Technologies which are used to creation mobile application are analyzed. There are also mentioned today's most used codecs in the mobile devices. There are described standards, norms, principles and recommendations for transfer multimedia data on network in real time. Technologies which are appropriate for functionality of system on server side are described. The output of the work is an operational sample of whole system and description of functionality of this system.

A signalization in an Internet protocol environment is commonly used for monitoring quality of service and other parameters of a network. This thesis is involved in transmission of signalization through internet protocol networks and proposes scalable solution for small and even for large-scale internet television broadcasting. The main contribution of this thesis lies in design and validation of optimal hierarchical tree on the basis of resources assigned. This is done in respect to geographical distance, network distance of each particular member of the hierarchical structure. For the design of algorithms simulations and global experimental network were used.