Tutkielma ryhmitellyistä konferensseista ja Binary Floor Control Protocol:n toteutuksesta keskitettyyn konferenssijärjestelmään

The introduction of the third generation (3G) in the mobile telecommunication world offers the possibility for a wide range of new applications and services that operators can offer to their customers. One of these services is multimedia conferencing. There is ongoing work to provide conferencing services in the IP Multimedia Subsystem (IMS) environment as one of the most significant services. This thesis focuses on providing a comprehensive overview of conferencing systems, especially of the Binary Floor Control Protocol (BFCP) and cascade conferences. The Master's Thesis consisted of two parts: The first part is a theoretical part, which provides the concepts of the centralized conferencing, known as tightly coupled conferences, and reviews the current specifications stage of the different standardization bodies. In contrast, the study of the applicability of the current centralized conferencing specifications in a cascaded conferencing environment is presented, as well as the strengths and weaknesses of them. The second part is a practical implementation of the Binary Floor Control Protocol (BFCP). BFCP is implemented in MiniSip, an existing secure open-source SIP User Agent (UA), and in Asterisk, an open source Private Branch Exchange (PBX) replacement system. BFCP is built using the specification defined by the XCON working group within the Internet Engineering Task Force (IETF). Finally, BFCP is evaluated and based on this evaluation, some conclusions are given.Kolmannen sukupolven matkapuhelinverkot mahdollistavat laajan uusien ohjelmien ja palveluiden kirjon, joita operaattorit voivat tarjota asiakkailleen. Eräs tämänlainen palvelu on multimedia konferenssi. Tällä hetkellä tehdään työtä, jonka tarkoituksena on mahdollistaa konferenssinpalvelun tarjoaminen IP Multimedia Subsystem (IMS) ympäristössä. Tämä diplomityö keskittyy konferenssijärjestelmän perusteelliseen kuvaukseen, painottuen Binary Floor Control Protocol:aan (BFCP) sekä ryhmiteltyihin konferensseihin. Työ koostuu kahdesta osasta: Ensimmäinen osa keskittyy teoriaan, joka käsittelee keskitettyjä konferenssijärjestelmiä sekä aiheen nykyistä tilaa eri standardointiorganisaatioissa. Vastakohtana tarkastellaan nykyisen keskitetyn konferenssijärjestelmän heikkouksia ja vahvuuksia. Toinen osa käsittelee käytännön toteutusta BFCP:sta, joka on toteutettu MiniSip- sekä Asterisk-ohjelmistoihin. MiniSip on avoimeen lähdekoodiin perustuva SIP käyttäjäagentti, ja Asterisk paikallisvaihdeohjelmiston (PBX) avoin korvaaja. BFCP perustuu spesifikaatioon, jonka on määritellyt XCON työryhmä IETF:ssa. Lopuksi BFCP protokollaa on arvioitu tämän toteutuksen avulla

A Floor Control Server in a Distributed Conference Service

The conferencing systems in IP Multimedia (IM) networks are going through restructuring, accomplished in the near future. One of the changes introduced is the concept of floors and floor control in its current form with matching entity roles. The Binary Floor Control Protocol (BFCP) is a novelty to be exploited in distributed tightly coupled conferencing services. The protocol defines the floor control server (FCS), which implements floor control giving access to shared resources. As the newest tendency is to distribute the conferencing services, the locations of different functionality units play an important role in developing the standards. The floor control server location is not yet single-mindedly fixed in different standardization bodies, and the debate goes on where to place it within the media server, providing the conferencing service. The thesis main objective is to evaluate two distinctive alternatives in respect the Mp interface protocol between the respective nodes, as the interface in relation to floor control is under standardization work at the moment. The thesis gives a straightforward preamble in IMS network, nodes of interest including floor control server and conferencing. Knowledge on several protocols – BFCP, SDP, SIP and H.248 provides an important background for understanding the functionality changes introduced in the Mp interface and therefore introductions on those protocols and how they are connected to the full picture is given. The actual analysis on the impact of the floor control server into the Mp reference point is concluded in relation to the locations, giving basic flows, requirements analysis including a limited implementation proposal on supporting protocol parameters. The overall conclusion of the thesis is that even if both choices are seemingly useful, not one of the locations is clearly the most suitable in the light of this work. The thesis suggests a solution having both possibilities available to be chosen from in separate circumstances, realized with consistent standardization. It is evident, that if the preliminary assumption for the analysis is kept regarding to only one right place for the floor control server, more work is to be done in connected areas to discover the one most appropriate location

Technologies of data transmission in video conferencing systems

The methods and technologies, associated with video and audio data transmission in video conferencing systems were discovered. Existing problems in the field of data transmission in conferencing systems via the Internet and methods for their solution were considered. The most widely used technologies of data transmission in the video conferencing systems via the Internet were analyzed. The concept of telepresence and features of video conferencing systems that it provides were considered. Conclusions about the appropriateness of using different technologies of video and audio data transfer depending on the type of video conferencing system were made.Досліджено методи і технології передачі відео- та аудіоданих у системах відеоконференцзв’язку. Розглянуто існуючі проблеми у сфері передачі даних у системах конференцзв'язку через Інтернет і методи їх вирішення. Проаналізовано найбільш поширені технології передачі відео- та аудіоданих у системах відеоконференцзв'язку через Інтернет. Розглянуті поняття телеприсутності і особливості систем відеоконференцзв'язку, що її забезпечують. Зроблено висновки про доцільність використання різних технологій передачі відео- та аудіоданих залежно від типу системи відеоконференцзв'язку.Исследованы методы и технологии передачи видео- и аудиоданных в системах видеоконференцсвязи. Рассмотрены существующие проблемы в сфере передачи данных в системах конференцсвязи через Интернет и методы их решения. Проанализированы наиболее распространенные технологии передачи видео- и аудиоданных в системах видеоконференцсвязи через Интернет. Рассмотрены понятие телеприсутствия и особенности систем видеоконференцсвязи, которые его обеспечивают. Сделаны выводы о целесообразности использования различных технологий передачи видео- и аудиоданных в зависимости от типа системы видеоконференцсвязи

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

A SOAP Web Services-Based Architecture for Floor Control in Multimedia Conferencing

Multimedia conferencing applications are an important and widely-used category of Web applications. Floor control is a significant and advanced feature of multimedia conferencing applications. Floor control mechanisms, when introduced in audio/video conferencing, control the media streams such as identifying which participant is allowed to send and who can be seen or heard. This prevents conflict and ensures an optimized use of resources between the conference participants. Floor control is composed of three logical entities: a single floor control server (i.e. entity responsible for managing the floors and their status), one or more floor chairs (moderators), and any number of regular conference participants. This thesis proposes a SOAP Web services based architecture for floor control in multimedia conferencing. Web services are designed to support interoperable machine-to-machine interaction over a network. They are attractive because of their flexibility. There are two types of web services: SOAP Web services and RESTful Web services. In SOAP Web services, interactions between the entities are based on XML and use SOAP, which is embedded in HTTP. RESTful web services are an architectural design style that rely on HTTP, but do not use SOAP. XML is also optional. We propose a set of floor control requirements and use them to review the related work and pinpoint the weaknesses. The proposed architecture includes the main components of floor control. It also includes a comprehensive set of server-side and client-side SOAP web service APIs that expose the floor control capabilities to application developers. The proposed APIs are programming language-independent and provide a higher level of abstraction to the application developers, which enables the interoperability. Furthermore, in the proposed architecture the floor control clients do not interact directly with the floor control server (FCS) but through a gateway accessible using SOAP web services. This opens up the possibility to use different floor control protocols transparently to the floor control clients. Application portability is no longer a problem because floor clients access the floor capabilities independently of the protocol supported by the FCS. We have built a conferencing application with floor control as a proof of concept to demonstrate the new interface for floor control and the feasibility of the proposed architecture. In addition, performance measurements have also been made to evaluate the viability of the architecture

Advancing Multimedia: Application Sharing, Latency Measurements and User-Created Services

Online collaboration tools exist and have been used since the early days of the Internet. Asynchronous tools such as wikis and discussion boards and real-time tools such as instant messaging and voice conferencing have been the only viable collaboration solutions up until recently, due to the low bandwidth between participants. With the increasing bandwidth in computer networks, multimedia collaboration such as application sharing and video conferencing have become feasible. Application and desktop sharing allows sharing of any application with one or more people over the Internet. The participants receive the screen-view of the shared application from the server. Their mouse and keyboard events are delivered and regenerated at the server. Application and desktop sharing enables collaborative work, software tutoring, and e-learning over the Internet. I have developed a high performance application and desktop sharing system called BASS which is efficient, reliable, independent of the operating system, scales well via heterogeneous multicast, supports all applications, and features true application sharing. Most of the time an application sharing session requires audio and video conferencing to be more useful. High quality video conferencing requires a fair amount of bandwidth and unfortunately Internet bandwidth of home users is still limited and shared by more than one application and user. Therefore, I measured the performance of popular video conferencing applications under congestion to understand whether they are flexible enough to adapt to fluctuating and limited bandwidth conditions. In particular, I analyzed how Skype, Windows Live Messenger, Eyebeam and X-Lite react to changes in available bandwidth, presence of HTTP and BitTorrent traffic and wireless packet losses. To perform these measurements more effectively, I have also developed vDelay, a novel tool for measuring the capture-to-display latency (CDL) and frame rate of real-time video conferencing sessions. vDelay enables developers and testers to measure the CDL and frame rate of any video conferencing application without modifying the source code. Further, it does not require any specialized hardware. I have used vDelay to measure the CDL and frame rate of popular video chat applications including Skype, Windows Live Messenger, and GMail video chat. vDelay can also be used to measure the CDL and frame rate of these applications in the presence of bandwidth variations. The results from the performance study showed that existing products, such as Skype, adapt to bandwidth fluctuations fairly well and can differentiate wireless and congestion-based packet losses. Therefore, rather than trying to improve video conferencing tools, I changed my focus to end-user created communication-related services to increase the utility of existing stand alone Internet services, devices in the physical world, communication and online social networks. I have developed SECE (Sense Everything, Control Everything), a new language and its supporting software infrastructure for user created services. SECE allows non-technical end-users to create services that combine communication, social networks, presence, calendaring, location and devices in the physical world. SECE is an event-driven system that uses a natural-English-like language to trigger action scripts. Users associate actions with events and when an event happens its associated action is executed. Presence updates, social network updates, incoming calls, email, calendar and time events, sensor inputs and location updates can trigger rules. SECE retrieves all this information from multiple sources to personalize services and to adapt them to changes in the user's context and preferences. Actions can control the delivery of email, change the handling of phone calls, update social network status and set the state of actuators such as lights, thermostats and electrical appliances

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

Media Processing in Video Conferences for Cooperating Over the Top and Operator Based Networks

Telecom operators have dominated the communication industry for a long time by providing services with guaranteed quality of service. Such services are provided by the operator at the cost of maintaining a high grade network. With the introduction of broadband and internet, many over the top (OTT) services have emerged. These services use the underlying operator networks as a mere bit pipe while all service intelligence resides in the application running on the client device. Introduction of OTT services has seen a good response from general users who are no longer bound to services provided by the network operator. This in turn has caused operators and telecom companies to loose the ownership of their customers. This thesis takes media processing in video conferencing as a case study to compare the two competing domains of operator networks and OTT networks. Both domains offer video conferencing to end users, but they follow different architectures. The study shows that OTT services can perform much better if they utilize support of the underlying network. This will also bring the user base back to the network operator. The proposal is to turn the competition into cooperation between both parties. Assessments are done from both technical as well as business perspectives to assert that such cooperative agreements are possible and should be experimented in real life