H-P2PSIP: Interconnection of P2PSIP domains for Global Multimedia Services based on a Hierarchical DHT Overlay Network

The IETF P2PSIP WG is currently standardising a protocol for distributed mul- timedia services combining the media session functionality of SIP and the decentralised distribution and localisation of resources in peer-to-peer networks. The current P2PSIP scenarios only consider the infrastructure for the connectivity inside a single domain. This paper proposes an extension of the current work to a hierarchical multi-domain scenario: a two level hierarchical peer-to-peer overlay architecture for the interconnection of different P2PSIP domains. The purpose is the creation of a global decentralised multimedia services in enterprises, ISPs or community networks. We present a study of the Routing Performance and Routing State in the particular case of a two-level Distributed Hash Table Hierarchy that uses Kademlia. The study is supported by an analytical model and its validation by a peer-to-peer simulator.En prens

Using an External DHT as a SIP Location Service

Peer-to-peer Internet telephony using the Session Initiation Protocol (P2P-SIP) can exhibit two different architectures: an existing P2P network can be used as a replacement for lookup and updates, or a P2P algorithm can be implemented using SIP messages. In this paper, we explore the first architecture using the OpenDHT service as an externally managed P2P network. We provide design details such as encryption and signing using pseudo-code and examples to provide P2P-SIP for various deployment components such as P2P client, proxy and adaptor, based on our implementation. The design can be used with other distributed hash tables (DHTs) also

An interoperable and secure architecture for internet-scale decentralized personal communication

Interpersonal network communications, including Voice over IP (VoIP) and Instant Messaging (IM), are increasingly popular communications tools. However, systems to date have generally adopted a client-server model, requiring complex centralized infrastructure, or have not adhered to any VoIP or IM standard. Many deployment scenarios either require no central equipment, or due to unique properties of the deployment, are limited or rendered unattractive by central servers. to address these scenarios, we present a solution based on the Session Initiation Protocol (SIP) standard, utilizing a decentralized Peer-to-Peer (P2P) mechanism to distribute data. Our new approach, P2PSIP, enables users to communicate with minimal or no centralized servers, while providing secure, real-time, authenticated communications comparable in security and performance to centralized solutions.;We present two complete protocol descriptions and system designs. The first, the SOSIMPLE/dSIP protocol, is a P2P-over-SIP solution, utilizing SIP both for the transport of P2P messages and personal communications, yielding an interoperable, single-stack solution for P2P communications. The RELOAD protocol is a binary P2P protocol, designed for use in a SIP-using-P2P architecture where an existing SIP application is modified to use an additional, binary RELOAD stack to distribute user information without need for a central server.;To meet the unique security needs of a fully decentralized communications system, we propose an enrollment-time certificate authority model that provides asserted identity and strong P2P and user-level security. In this model, a centralized server is contacted only at enrollment time. No run-time connections to the servers are required.;Additionally, we show that traditional P2P message routing mechanisms are inappropriate for P2PSIP. The existing mechanisms are generally optimized for file sharing and neglect critical practical elements of the open Internet --- namely link-level security and asymmetric connectivity caused by Network Address Translators (NATs). In response to these shortcomings, we introduce a new message routing paradigm, Adaptive Routing (AR), and using both analytical models and simulation show that AR significantly improves message routing performance for P2PSIP systems.;Our work has led to the creation of a new research topic within the P2P and interpersonal communications communities, P2PSIP. Our seminal publications have provided the impetus for subsequent P2PSIP publications, for the listing of P2PSIP as a topic in conference calls for papers, and for the formation of a new working group in the Internet Engineering Task Force (IETF), directed to develop an open Internet standard for P2PSIP

A service-enabling framework for the session initiation protocol (SIP)

In this dissertation, we propose a framework to provide multimedia communication services. Our proposed framework is based on SIP (Session Initiation Protocol) and has four fundamental properties: it is available, secure, high performing, and oriented to innovations. The framework is not an architecture with a rigid structure. Instead, the framework is a toolkit made up of a set of tools that can be combined in different ways. The combination of these tools provides applications and services with functionality needed to implement a wide variety of multimedia communication services. Applications and services built on top of the framework use different tools within the toolkit in order to provide their desired overall functionality. The functionality provided by the framework includes a number of primitives to be used by applications and services. These primitives mostly relate to multiparty communications and include floor control. The framework also offers support functions that relate to PSTN (Public Switched Telephony Network) interworking, policy control, and consent-based communications. Additionally, the framework contains functions that relate to signalling transport, multihoming, mobility, security, and NAT (Network Address Translation) traversal. The framework also allows building overlay networks when a SIP network infrastructure is not available. In order to test and refine the ideas presented in this dissertation, we have implemented most of them in proof-of-concept prototypes. We have used experiments and simulations to validate our assumptions and obtain new insights

Designing and prototyping WebRTC and IMS integration using open source tools

WebRTC, or Web Real-time Communications, is a collection of web standards that detail the mechanisms, architectures and protocols that work together to deliver real-time multimedia services to the web browser. It represents a significant shift from the historical approach of using browser plugins, which over time, have proven cumbersome and problematic. Furthermore, it adopts various Internet standards in areas such as identity management, peer-to-peer connectivity, data exchange and media encoding, to provide a system that is truly open and interoperable. Given that WebRTC enables the delivery of multimedia content to any Internet Protocol (IP)-enabled device capable of hosting a web browser, this technology could potentially be used and deployed over millions of smartphones, tablets and personal computers worldwide. This service and device convergence remains an important goal of telecommunication network operators who seek to enable it through a converged network that is based on the IP Multimedia Subsystem (IMS). IMS is an IP-based subsystem that sits at the core of a modern telecommunication network and acts as the main routing substrate for media services and applications such as those that WebRTC realises. The combination of WebRTC and IMS represents an attractive coupling, and as such, a protracted investigation could help to answer important questions around the technical challenges that are involved in their integration, and the merits of various design alternatives that present themselves. This thesis is the result of such an investigation and culminates in the presentation of a detailed architectural model that is validated with a prototypical implementation in an open source testbed. The model is built on six requirements which emerge from an analysis of the literature, including previous interventions in IMS networks and a key technical report on design alternatives. Furthermore, this thesis argues that the client architecture requires support for web-oriented signalling, identity and call handling techniques leading to a potential for IMS networks to natively support these techniques as operator networks continue to grow and develop. The proposed model advocates the use of SIP over WebSockets for signalling and DTLS-SRTP for media to enable one-to-one communication and can be extended through additional functions resulting in a modular architecture. The model was implemented using open source tools which were assembled to create an experimental network testbed, and tests were conducted demonstrating successful cross domain communications under various conditions. The thesis has a strong focus on enabling ordinary software developers to assemble a prototypical network such as the one that was assembled and aims to enable experimentation in application use cases for integrated environments

Analysis of RTCWeb Data Channel Transport Options

The Web has introduced a new technology in a more distributed and collaborative form of communication, where the browser and the user replace the web server as the nexus of communications in a way that after the call establishment through web servers, the communication is performed directly between browsers as peer to peer fashion without intervention of the web servers. The goal of Real Time Collaboration on the World Wide Web (RTCWeb) project is to allow browsers to natively support voice, video, and gaming in interactive peer to peer communications and real time data collaboration. Several transport protocols such as TCP, UDP, RTP, SRTP, SCTP, DCCP presently exist for communication of media and non-media data. However, a single protocol alone can not meet all the requirements of RTCWeb. Moreover, the deployment of a new transport protocol experiences problems traversing middle boxes such as Network Address Translation (NAT) box, firewall. Nevertheless, the current implementation for transportation of non-media in the very first versions of RTCWeb data does not include any congestion control on the end-points. With media (i.e., audio, video) the amount of traffic can be determined and limited by the codec and profile used during communication, whereas RTCWeb user could generate as much as non-media data to create congestion on the networks. Therefore, a suitable transport protocol stack is required that will provide congestion control, NAT traversal solution, and authentication, integrity, and privacy of user data. This master's thesis will give emphasis on the analysis of transport protocol stack for data channel in RTCWeb and selects Stream Control Transmission Protocol (SCTP), which is a reliable, message oriented general-purpose transport layer protocol, operating on top of both IPv4 and IPv6, providing congestion control similar to TCP and additionally, some new functionalities regarding security, multihoming, multistreaming, mobility, and partial reliability. However, due to the lack of universal availability of SCTP within the OS(s), it has been decided to use the SCTP userland implementation. WebKit is an open source web browser engine for rendering web pages used by Safari, Dashboard, Mail, and many other OS X applications. In WebKit RTCWeb implementation using GStreamer multimedia framework, RTP/UDP is utilized for the communication of media data and UDP tunnelling for non-media data. Therefore, in order to allow a smooth integration of the implementation within WebKit, we have decided to implement GStreamer plugins using SCTP userland stack. This thesis work also investigates the way Mozilla has integrated those protocols in the browser's network stack and how the Data Channel has been designed and implemented using SCTP userland stack

ADAPTIVE UNMANNED VEHICLE AUTOPILOTING USING WEBRTC VIDEO ANALYSIS

Εκμεταλλευόμαστε τις νέες δυνατότητες που παρέχονται από το WebRTC, υπό την έννοια της διαλειτουργικότητας και των τελευταίας γενιάς επικοινωνιών σε πραγματικό χρόνο, προκειμένου να αναπτύξουμε ένα σύστημα για το πιλοτάρισμα μη επανδρωμένων οχημάτων χρησιμοποιώντας ανάλυση βίντεο. Συγκεκριμένα, ορίζουμε μια τοπολογία όπου ένα ROS όχημα μεταδίδει βίντεο μέσω WebRTC προς έναν ενδιάμεσο εξυπηρετητή, ο οποίος με τη σειρά του το μεταβιβάζει σε έναν πελάτη. Ο εξυπηρετητής εκμεταλλεύεται τη βιβλιοθήκη OpenCV και εφαρμόζει ανάλυση βίντεο, με τέτοιο τρόπο ώστε να εξυπηρετήσει ένα επιλεγμένο από τον πελάτη σενάριο. Οι αντίστοιχες εντολές μεταδίδονται στο όχημα, με αποτέλεσμα να έχουμε ένα αυτόματα οδηγούμενο όχημα. Ο πελάτης παρακολουθεί την πορεία του οχήματος και μπορεί να αλλάξει δυναμικά το επιλεγμένο σενάριο – αυτό σημαίνει είτε να αλλάξει ελαφρώς τη λειτουργία του (π.χ. από παρακολούθηση ανθρώπων σε παρακολούθηση παιδιών) είτε να ενεργοποιήσει μια εντελώς διαφορετική φιλοσοφία λειτουργίας – στέλνοντας τα κατάλληλα αιτήματα στον εξυπηρετητή. Μόλις ο εξυπηρετητής λάβει αυτά τα αιτήματα, χρησιμοποιεί τις αντίστοιχες λειτουργίες το OpenCV για να εξυπηρετήσει το νέο σενάριο, και στέλνει τις νέες εντολές οδήγησης στο όχημα, αναγκάζοντας το σύστημα να υιοθετήσει μια νέα λειτουργία αυτόματου πιλότου. Η επικοινωνία μεταξύ του οχήματος, του εξυπηρετητή και του πελάτη εδραιώνεται μέσω των SIP/SDP και ενορχηστρώνεται μέσω ενός Web-Socket εξυπηρετητή που επιτελεί το ρόλο του Signaling Server, ενώ οι εντολές μεταφέρονται μέσω του WebRTC Data Channel πάνω από το SCTP. Περιγράφουμε και αναλύουμε το πώς όλα αυτά τα ετερογενή συστατικά (WebRTC – OpenCV – ROS) συνδυάζονται για τη δημιουργία μιας δικτυακής υποδομής, για το αυτόματο πιλοτάρισμα ROS οχημάτων σύμφωνα με ένα συγκεκριμένο σενάριο χρήσης. Τέλος, τα αποτελέσματα αποδεικνύουν την ιδέα μας, δηλαδή μια οριζόντια υποδομή που (α) αποτελείται από μια ευέλικτη/αρθρωτή αρχιτεκτονική, (β) παρέχει τα απαραίτητα στοιχεία για την μηχανή-σε-μηχανή επικοινωνία, (γ) χρησιμοποιεί τελευταίας γενιάς τεχνολογίες, (δ) επιτρέπει σε έναν προγραμματιστή να εφαρμόσει τη δική του λογική κατακόρυφα σε βάθος και (ε) παρέχει στον τομέα του IoT μια λύση που μπορεί εύκολα να αξιοποιηθεί με πολλούς τρόπους.We exploit the new features provided by WebRTC in terms of interoperability and state-of-the-art real-time communications, in order to develop a system for piloting unmanned vehicles using video analysis. Specifically, we define a topology where a ROS-based vehicle transmits its video using WebRTC to an intermediate server, who in turn relays it to a client. The server takes advantage of the OpenCV library and applies video analysis, with respect to a selected task (i.e. face detection) defined by the client. The corresponding commands are transmitted to the vehicle, resulting in an automatically driven unmanned vehicle. The client monitors the vehicle’s movement and can dynamically change the selected use case; that is, either change slightly its operation (i.e. from human tracking to children tracking) or enable an entirely new core philosophy (i.e. to fire detection) by sending the appropriate requests to the server. Upon reception of these requests, the server utilizes the corresponding OpenCV functionalities to serve the new task, and sends the new piloting commands to the vehicle, forcing the system to adopt a new autopiloting mode. This communication between the vehicle, the server and the client is established using SIP/SDP and orchestrated via a WebSocket server that serves as a Signaling Server, the media are transferred through SRTP/UDP, and the commands are carried via the WebRTC Data Channel over SCTP. We explain and describe how to combine all of these heterogeneous components (WebRTC – OpenCV – ROS), in order to compose a web-based infrastructure for autopiloting ROS-based vehicles upon a specific use case. Finally, the results prove our concept, meaning a horizontal infrastructure that (a) consists of a modular architecture, (b) provides the necessary components for machine-to-machine communication, (c) uses state-of-the-art technologies, (d) allows a developer to implement her own logic vertically, and (e) provides IoT with a solution that can be easily exploited in numerous ways

Inter-domain interoperability framework based on WebRTC

Nowadays, the communications paradigm is changing with the convergence of communication services to a model based on IP networks. Applications such as messaging or voice over IP are increasing its popularity and Communication Service Providers are focusing on offering this kind of services. Moreover, Web Real Time Communication (WebRTC) has emerged as a technology that eases the creation of web applications featuring Real-Time Communications over IP networks without the need to develop and install any plug-in. It lacks of specifications in the control plane, leaving the possibility to use WebRTC over tailored web signalling solutions or legacy networks such as IP Multimedia Subsystem (IMS). This technology brings a wide range of possibilities for web developers, but Communication Service Providers are adviced to develop solutions based on the WebRTC technology as described in the Eurescom Study P2252. The lack of WebRTC specifications on the signalling platform together with the threats and opportunities that this technology represents for Communication Service Providers, makes evident the need of research on interoperability solutions for the different kind of signalling implementations and experimentation on the best way for Communication Service Providers to obtain the maximum benefit from WebRTC technology. The main goal of this thesis is precisely to develop a WebRTC interoperability framework and perform experiments on whether the Communication Service Providers should use their existing IMS solutions or develop tailored web signalling platforms for WebRTC deployments. In particular, the work developed in this thesis was completed under the framework of the Webrtc interOperability tested in coNtradictive DEployment scenaRios (WONDER) experimentation for the OpenLab project. OpenLab is a Large-scale integrating project (IP) and is part of the European Union Framework Programme 7 for Research and Development (FP7) addressing the work programme topic Future Internet Research and Experimentation.Actualmente, el paradigma de comunicaciones está cambiando gracias a la convergencia de los servicios de comunicaciones hacia un modelo basado en redes IP. Aplicaciones tales como la mensajería y la voz sobre IP están creciendo en popularidad mientras los proveedores de servicios de comunicaciones se centran en ofrecer este tipo de servicios basados en redes IP. Por otra parte, la tecnología WebRTC ha surgido para facilitar la creación de aplicaciones web que incluyan comunicaciones en tiempo real sobre redes IP sin la necesidad de desarrollar o instalar ningún complemento. Esta tecnología no especifica los protocolos o sistemas a utilizar en el plano de control, dejando a los desarrolladores la posibilidad de usar WebRTC sobre soluciones de señalizaci on web específicas o utilizar las redes de señalización existentes, tales como IMS. WebRTC abre un gran abanico de posibilidades a los desarrolladores web, aunque también se recomienda a los proveedores de servicios de comunicaciones que desarrollen soluciones basadas en WebRTC como se describe en el estudio P2252 de Eurescom. La falta de especificaciones en el plano de señalización junto a las oportunidades y amenazas que WebRTC representa para los proveedores de servicios de comunicaciones, hacen evidente la necesidad de investigar soluciones de interoperabilidad para las distintas implementaciones de las plataformas de señalización y de experimentar c omo los proveedores de servicios de comunicaciones pueden obtener el máximo provecho de la tecnología WebRTC. El objetivo principal de este Proyecto Fin de Carrera es desarrollar un marco de interoperabilidad para WebRTC y realizar experimentos que permitan determinar bajo que condiciones los proveedores de servicios de comunicaciones deben utilizar las plataformas de se~nalizaci on existentes (en este caso IMS) o desarrollar plataformas de señalización a medida basadas en tecnologías web para sus despliegues de WebRTC. En particular, el trabajo realizado en este Proyecto Fin de Carrera se llevó a cabo bajo el marco del proyecto WONDER para el programa OpenLab. OpenLab es un proyecto de integración a gran escala en el cual se desarrollan investigaciones y experimentos en el ámbito del futuro Internet y que forma parte del programa FP7 de la Unión Europea.Ingeniería de Telecomunicació

Dynamic media stream mobility with TURN

Multi party videoconference systems use MCU (Multipoint Control Unit) devices to forward media streams. In this paper we describe a mechanism that allows the mobility of such streams between MCU devices. This mobility is especially useful when redistribution of streams is needed due to scalability requirements. These requirements are mandatory in Cloud scenarios to adapt the number of MCUs and their capabilities to variations in the user demand. Our mechanism is based on TURN (Traversal Using Relay around NAT) standard and adapts MICE (Mobility with ICE) specification to the requirements of this kind of scenarios. We conclude that this mechanism achieves the stream mobility in a transparent way for client nodes and without interruptions for the users