Security in Peer-to-Peer SIP VoIP

VoIP (Voice over Internet Protocol) is one of the fastest growing technologies in the world. It is used by people all over the world for communication. But with the growing popularity of internet, security is one of the biggest concerns. It is important that the intruders are not able to sniff the packets that are transmitted over the internet through VoIP. Session Initiation Protocol (SIP) is the most popular and commonly used protocol of VoIP. Now days, companies like Skype are using Peer-to-Peer SIP VoIP for faster and better performance. Through this project I am improving an already existing Peer-to-Peer SIP VoIP called SOSIMPLE P2P VoIP by adding confidentiality in the protocol with the help of public key cryptography

WiLiTV: A Low-Cost Wireless Framework for Live TV Services

With the evolution of HDTV and Ultra HDTV, the bandwidth requirement for IP-based TV content is rapidly increasing. Consumers demand uninterrupted service with a high Quality of Experience (QoE). Service providers are constantly trying to differentiate themselves by innovating new ways of distributing content more efficiently with lower cost and higher penetration. In this work, we propose a cost-efficient wireless framework (WiLiTV) for delivering live TV services, consisting of a mix of wireless access technologies (e.g. Satellite, WiFi and LTE overlay links). In the proposed architecture, live TV content is injected into the network at a few residential locations using satellite dishes. The content is then further distributed to other homes using a house-to-house WiFi network or via an overlay LTE network. Our problem is to construct an optimal TV distribution network with the minimum number of satellite injection points, while preserving the highest QoE, for different neighborhood densities. We evaluate the framework using realistic time-varying demand patterns and a diverse set of home location data. Our study demonstrates that the architecture requires 75 - 90% fewer satellite injection points, compared to traditional architectures. Furthermore, we show that most cost savings can be obtained using simple and practical relay routing solutions

Supporting quality of service for internet applications

University of Technology, Sydney. Faculty of Information Technology.Regarding the dominance of IP applications and the requirement of providing quality of service for users, it is critical to provide an scalable network architecture capable of supporting sufficient Quality of Service (QoS). Of the two network models (Integrated Services and Differentiated Services) approved by the Internet Engineering Task Force (IETF) [1, 2], the differentiated service model has gained wider acceptance because of its scalability. Differentiated Services (DiffServ) QoS architecture is scalable but inadequate to deal with network congestion and unable to provide fairness among its traffic aggregates. Recently, IETF has recommended additional functions including admission control and resource discovery to enhance the original DiffServ [2]. In this thesis, we propose a new framework based on DiffServ. The new architecture, called Fair Intelligent Congestion Control DiffServ (FICC- DiffServ), applies the FICC algorithm and control loop to provide fairness among traffic aggregates and control congestion inside DiffServ networks. The augmented architecture is realisable within the existing IP network infrastructures. Simulation results show that the FICC-DiffServ performs excellently in terms of guaranteed fairness, minimised packet delay and jitter, as well as being robust to traffic attributes, and being simple to implement. Moreover, providing end-to-end QoS for Internet applications presents difficult problems, because the Internet is composed of many independently administrative domains called Autonomous Systems. Enabling end-to-end QoS, negotiations between domains is then crucial. As a means of negotiations, inter- autonomous system QoS routings play an important role in advertising the available network resources between domains. In this thesis, the Border Gateway Protocol (BGP) is extended to provide end-to-end QoS. The BGP is selected for two reasons: (1) BGP is an inter-domain routing protocol widely used on the Internet and (2) the use of attributes attached to routes makes BGP be a powerful and scalable inter-domain routing protocol. For end-to-end QoS, a completed framework includes a FICC-DiffServ in each domain, an extended BGP between domains and an admission control at the edge router. Via simulation, we demonstrate the reliability of the BGP-extended architecture, including route selection policy and overhead reduction issues

Agent based infrastructure for real-time applications

In this paper we propose a new infrastructure for real-time applications. As a preliminary, we describe basic characteristics of the most popular real-time services like VoIP, videoconferencing, live media streaming, and network multiplayer games. We focus on the end-to-end latency, bandwidth and efficient transmission methods. Next, we present our project concepts, infrastructure model, details of implementation and our testing environment which was designed for testing many aspects of real-time services. The system combines mechanisms for ensuring best possible connection quality (QoS), load balance of servers in infrastructure and gives control over the packet routing decisions. Additionally, provided security mechanisms make it a good choice even in the environment where a high security level is required. The system is based on the Peer-to-Peer (P2P) model and data between users is routed over an overlay network, consisting of all participating peers as network nodes. This overlay can by used for application level multicast or live media stream. In the logging process each user is assigned to a specific node (based on his geographic location and nodes load). Because nodes are participating in data transmission, we have control over the data flow route. It is possible to specify the desired route, so, regardless of the external routing protocol, we can avoid paths that are susceptible to eavesdropping. Another feature of the presented system is usage of agents. Each agent acts within the single node. Its main task is to constantly control the quality of transmission. It analyzes such parameters like link bandwidth use, number of lost packets, time interval between each packet etc. The information collected by the agents from all nodes allows to build a dynamic routing table. Every node uses the Dijkstra's algorithm to find the best at the moment route to all other nodes. The routes are constantly modified as a consequence of changes found by agents or updates sent by other nodes. In VoD services agents also analyze popularity of streamed media, which helps build intelligent video cache. To ensure greater security and high reliability of the system, we have provided a reputation mechanism. It is used during bringing up to date the information about possible routes and their quality, given by other nodes. Owing to this solution nodes and routes which are more reliable get higher priority

Bandwidth-efficient Video Streaming with Network Coding on Peer-to-Peer Networks

PhDOver the last decade, live video streaming applications have gained great popularity among users but put great pressure on video servers and the Internet. In order to satisfy the growing demands for live video streaming, Peer-to-Peer(P2P) has been developed to relieve the video servers of bandwidth bottlenecks and computational load. Furthermore, Network Coding (NC) has been proposed and proved as a significant breakthrough in information theory and coding theory. According to previous research, NC not only brings substantial improvements regarding throughput and delay in data transmission, but also provides innovative solutions for multiple issues related to resource allocation, such as the coupon-collection problem, allocation and scheduling procedure. However, the complex NC-driven P2P streaming network poses substantial challenges to the packet scheduling algorithm. This thesis focuses on the packet scheduling algorithm for video multicast in NC-driven P2P streaming network. It determines how upload bandwidth resources of peer nodes are allocated in different transmission scenarios to achieve a better Quality of Service(QoS). First, an optimized rate allocation algorithm is proposed for scalable video transmission (SVT) in the NC-based lossy streaming network. This algorithm is developed to achieve the tradeoffs between average video distortion and average bandwidth redundancy in each generation. It determines how senders allocate their upload bandwidth to different classes in scalable data so that the sum of the distortion and the weighted redundancy ratio can be minimized. Second, in the NC-based non-scalable video transmission system, the bandwidth ineffi- ciency which is caused by the asynchronization communication among peers is reduced. First, a scalable compensation model and an adaptive push algorithm are proposed to reduce the unrecoverable transmission caused by network loss and insufficient bandwidth resources. Then a centralized packet scheduling algorithm is proposed to reduce the unin- formative transmission caused by the asynchronized communication among sender nodes. Subsequently, we further propose a distributed packet scheduling algorithm, which adds a critical scalability property to the packet scheduling model. Third, the bandwidth resource scheduling for SVT is further studied. A novel multiple- generation scheduling algorithm is proposed to determine the quality classes that the receiver node can subscribe to so that the overall perceived video quality can be maxi- mized. A single generation scheduling algorithm for SVT is also proposed to provide a faster and easier solution to the video quality maximization function. Thorough theoretical analysis is conducted in the development of all proposed algorithms, and their performance is evaluated via comprehensive simulations. We have demon- strated, by adjusting the conventional transmission model and involving new packet scheduling models, the overall QoS and bandwidth efficiency are dramatically improved. In non-scalable video streaming system, the maximum video quality gain can be around 5dB compared with the random push method, and the overall uninformative transmiss- sion ratio are reduced to 1% - 2%. In scalable video streaming system, the maximum video quality gain can be around 7dB, and the overall uninformative transmission ratio are reduced to 2% - 3%

User generated content for IMS-based IPTV

Includes abstract.Includes bibliographical references.Web 2.0 services have been on the rise due to improved bandwidth availability. Users can now connect to the internet with a variety of portable devices which are capable of performing multiple tasks. Due to this, services such as Voice over IP (VoIP), presence, social networks, instant messaging (IM) and Internet Protocol television (IPTV) to mention but a few, started to emerge...This thesis proposed a framework that will offer user-generated content on an IMS-Based IPTV and the framework will include a personalised advertising system..

Peer-to-peer multimedia communication

I sistemi Peer-to-Peer (P2P) sono stati inventi, messi in campo e studiati da più di dieci anni, andando al di là della semplice applicazione per scambio di file. Nelle reti P2P i partecipanti si organizzano in una rete "overlay" che è astratta rispetto alle caratteristiche della sottostante rete fisica. Scopo di questi sistemi è la distribuzione di risorse quali contenuti, spazio di memorizzazione o cicli macchina. Gli utenti quindi giocano un ruolo attivo e possono essere considerati come sia clienti che serventi allo stesso tempo per il particolare servizio che la rete P2P offre. Lo scopo di questa tesi di dottorato è lo studio di questi sistemi ed il dare un contributo nella loro analisi prestazionale. L'analisi mira a valutare le prestazioni raggiunte dai sistemi e/o i limiti teorici raggiungibili. Infatti, nonostante esistano diversi meccanismi per il peer-to-peer streaming, l'analisi prestazionale di questo tipo di sistemi può essere considerata ancora nella sua infanzia. A questo scopo, i contributi principali di questa tesi di dottorato sono: i)la derivazione di un limite teorico per il ritardo nei sistemi di P2P streaming, ii) la creazione di un algoritmo che sfrutti le conoscenze acquisite attraverso il lavoro teorico, iii) l'analisi prestazionale dell'algoritmo utilizzando un simulatore espressamente progettato per riprodurre le caratteristiche delle reti P2P reali composte da centinaia di migliaia di nodi che si connettono e disconnettono in continuazione.Peer-to-Peer (P2P) systems have been invented, deployed and researched for more than ten years and went far beyond the simple file sharing applications. In P2P networks, participants organize themselves in an overlay network that abstracts from the topological characteristics of the underlying physical network. Aim of these systems is the distribution of some kind of resources like contents, storage, or CPU cycles. Users, therefore, play an active role so that they can be considered as client and server at the same time, for the particular service that is provided through the P2P paradigm. Goal of this dissertation thesis is to study these systems, and give contributes in their performance evaluation. The analysis will aim to evaluate the achieved performance of a system and/or the performance bounds that could be achievable. In fact, even if there are several proposals of different systems, peer-to-peer streaming performance analysis can be considered still in its infancy and there is still a lot of work to do. To this aim, the main contributes of this dissertation thesis are i) the derivation of a theoretical delay bounds for P2P streaming system ii) II the creation of an algorithm that exploits the new insights that come out from the theoretical study iii) the performance evaluation of this algorithm using an ad-hoc simulator, expressly tailored to reproduce the characteristics of the real-world P2P streaming systems, composed by hundred thousands of intermittently connected users