This thesis describes architecture of IP telephony networks based on signaling protocol SIP and transport protocol RTP. Also tools used for analyzing VoIP traffic are described. Main objective is to design and implement a system for the detection of calls and extraction of voice payloads from  the captured packets. The system first recognizes the signaling messages of SIP protocol. These messages are analyzed afterwards. Output statistics are generated based on gathered data. Voice data will be stored in form, which is suitable for further processing.

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 / Premios Extraordinarios de tesis doctorales

<p>In the thesis the advantages with Text over IP (ToIP) is explained and it is motivated why it is a good idea to integrate this in Asterisk. It also presents an implementation of a ToIP extension in Asterisk.</p><p>ToIP means communicating over a network based on Internet protocols with real-time text. Real-time text means a character is sent to the receiving terminal as soon the sender has typed it or with a small delay.</p><p>In the thesis IM and ToIP is compared in a survey. The result point at IM is not better than ToIP even though it is much more commonly used. VoIP can not replace ToIP either because there are occasions when ToIP is better for instance if the person using it is deaf or if a person want to make a private conversation in a noisy room.</p><p>Asterisk is an IP-PBX. PBX stands for Private Branch Exchange which means a private telephone system which is part of a larger network system that exchange information.</p><p>An IP-PBX is a PBX based on the Internet. Asterisk and many other IP-PBX can also exchange calls between the PSTN ant the Internet. By including ToIP in Asterisk it will be possible to exchange ToIP calls.</p><p>The implementation described is not only including ToIP in Asterisk but also a translation function between the text format called t140 and another text format called t140 with redundancy.</p><p>The idea is to extend the translation function in the future to more text formats.</p>

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).

Following the concept utilize binary selenides as precursors with rapid thermal process (RTP) to fabricate CuInSe2 (CIS) thin film. In order to find the most promise process to get high quality CIS, several precursor stacking sequences have been tested which including SLG/In-Se/Cu-Se/Se, SLG/Cu-Se/In-Se/Se, SLG/0.1In-Se/Cu-Se/0.9/In-Se/Se, and SLG/0.5In-Se/Cu-Se/0.5/In-Se/Se, and the experiment result shows SLG/In-Se/Cu-Se/Se is the most suitable stacking sequence. Subsequently, varying Se flux to obtain several kinds copper selenides (Cu7Se4, Cu3Se2, CuSe, CuSe2) and indium selenides, try to find the suitable pairs through these binary selenides in SLG/In-Se/Cu-Se/Se structure. The suitable combination phase in Cu-Se precursor layer is CuSe blend with CuSe2. Large grain size CIS, about 1£gm, can be prepared in such precursor phase with film thickness between 700nm to 1£gm, strong (112) prefer-orientation vertical with substrate as well as good adhesion. Films were characterized through scanning election microscopy (SEM) to obtain grain size, surface morphology as well as film thickness. The X-ray diffractometer (XRD) was used to identify phase contained in whole film, and the phase constitution near surface layer was examined by Raman spectroscopy. If there are some second phases remaining in the thin film, combining the phase examination result of XRD and Raman spectroscopy, it can be estimate the second phase exist in the surface layers or internal film area.

Due to the rapid development of the Internet and the fast expansion of the bandwidth, the requirement of the real-time service for the Internet is necessary. In this way, the problem of the real-time service for the Internet becomes an important issue. Most of the applications still use TCP as the protocol, but due to the reliable property of TCP, TCP can¡¦t fit the requirement of the real-time transfer. So, we need to implement a protocol which we can use on the real-time transfer service. According to the requirement, we find an open source application layer protocol ¡V UDT (UDP-based Data Transfer). We can implement the application of real-time transfer by using the partial reliable messaging property of the protocol. In addition, user can adjust the parameters or settings of the protocol to make their application get into better performance by using the composable property of the protocol. In our research, we¡¦ll compare with RTP (Real-time Transport Protocol) and UDT, and also explain the reason why we don¡¦t choose RTP in our research. The assumption environment of our research is financial real-time service, and the protocol of such applications is TCP. In this way, we¡¦ll analyze and compare the result of the tests between TCP and UDT. We¡¦ll also adjust the parameters of the protocol to test the performance of the UDT under the environment of the real-time transfer, such as data lose rate, etc.. These results can supply the reference for the users when using UDT as their protocol to implement their real-time applications.

UDP (User Datagram Protocol) and RTP (Real-time Transport Protocol), using fixed bit rate to convey data every time period, are the most pervasive transport protocols for multimedia traffic in communications networks. However, unexpected packet delay/jitter may occur when network becomes congested or channel interference remains unresolved. To reduce packet delay and packet loss for real-time traffic in a hybrid network from wired to wireless ad-hoc, this thesis presents RTP rate control with an ad-hoc gateway to dynamically adjust the transmission rate according to network conditions. With the proposed scheme, a source node can distinguish the two network conditions, congestion and interference, by monitoring RTCP (RTP control protocol) packets regularly reported from destination nodes and the associated ad-hoc gateway. Based on the RTCP reports, a sender node can dynamically change its encoding bit rate to improve the quality of real-time traffic. For the purpose of demonstration, we implement the proposed adaptive rate control scheme on a Linux platform for SIP-phone communications. The experimental results have shown that our proposed scheme not only relieves traffic congestion but also increases the number of received data even in the case of severe channel interference.

University, Diss., 2004--Mannheim.

Magister Legum - LLM / The aim of this mini-thesis is to examine the inconsistency in the implementation of the responsibility to protect (RTP) principle during armed conflicts with specific focus on the case of Libya and Darfur. Furthermore the mini-thesis scrutinizes the criteria which are utilized universally and questions whether the principle is determined by factors such as economics, politics and location depending on each crisis. The significance of this minithesis derives from the need to make a contribution to the new interventionism debate and contribute to the growing literature on the doctrine of the RTP especially when it comes to the inconsistencies during its application which seems to be on the rise especially in the African continent. The mini-thesis was guided by the following assumption that there are inconsistencies when it comes to the application of the RTP under humanitarian law. The mini-thesis also embarks on an enquiry into the legal aspects of the RTP doctrine and the legal status of humanitarian intervention. It is worth noting that the RTP doctrine does not concentrate on every human rights violation or abuse of power, even when these are very serious as in the case of Sudan. It certainly does not empower or establish an obligation on the international community to respond by over-riding the offending state’s sovereignty. The initial intention of the RTP was aimed at preventing mass attacks or large scale violations involving genocide, war crimes, ethnic cleansing and crimes against humanity. It is greatly disappointing to note that the international community at large tends to overlook the more severe crises which have more casualties and turn their eyes on less serious humanitarian crises. This raises concern about the extent of the inconsistency in the application of the RTP. The question that begs an answer therefore is why intervene in Libya and not Darfur? In conclusion to this mini-thesis I came to the realization that inconsistencies within the application of the RTP exist because humanitarian intervention under the RTP has a massive political element which affects implementation. The RTP is often used as a justification for states to act in conflicts when there is no domestic support for more direct political intervention. Thus, I believe that intervention can never be completely humanitarian driven until the five RTP precautionary principles are used as a guideline or criteria for interventions.

Video conferencing applications has reached worldwide usage in recent years by the help of the improvements in network infrastructures for public services. Media data covers a significant ratio of data traffic over IP networks. However, it is challenging to ensure a decent quality of service (QoS) on public networks in terms of video and audio quality. The main factor that may cause degradation in media playback quality is packet losses. There are various techniques available to conceal packet losses in lossy channels. According to the application needs and channel characteristics such as loss patterns and round trip times, retransmission or forward error correction techniques may be applied at application level. These two techniques have different challenges which lead to tradeoffs between them, thus one might be chosen over the others. In this thesis work, retransmission’s worst case performances under considered packet loss patterns and various round trip times are compared to performances of forward error correction schemes. In addition, implementation details with respect to the relevant RFCs are provided as an example to give a better judgement on the obtained results. Results obtained under the packet loss patterns that are generated with a simple Gilbert-Elliot 2-state model shows that forward error correction techniques are a reasonable choice of error concealing in the real-time transport protocol (RTP) stack where round trip time in the channel is greater than 200 ms. In addition, bandwidth overhead revealed by forward error correction stays higher than retransmission’s bandwidth overhead in all sample runs. In cases where round trip times are high, then the choice of forward error correction scheme is bound to the packet loss pattern. In the results section, it is obtained that ReedSolomon performs well in terms of residual packet losses, which are the packets not being recovered, and bandwidth overhead when losses occur in long bursts.