Michael Adeyeye

The village telco project: a reliable and practical wireless mesh telephony infrastructure

VoIP (Voice over IP) over mesh networks could be a potential solution to the high cost of making phone calls in most parts of Africa. The Village Telco (VT) is an easy to use and scalable VoIP over meshed WLAN (Wireless Local Area Network) telephone infrastructure. It uses a mesh network of mesh potatoes to form a peer-to-peer network to relay telephone calls without landlines or cell phone towers. This paper discusses the Village Telco infrastructure, how it addresses the numerous difficulties associated with wireless mesh networks, and its efficient deployment for VoIP services in some communities around the globe. The paper also presents the architecture and functions of a mesh potato and a novel combined analog telephone adapter (ATA) and WiFi access point that routes calls. Lastly, the paper presents the results of preliminary tests that have been conducted on a mesh potato. The preliminary results indicate very good performance and user acceptance of the mesh potatoes. The results proved that the infrastructure is deployable in severe and under-resourced environments as a means to make cheap phone calls and render Internet and IP-based services. As a result, the VT project contributes to bridging the digital divide in developing areas.

A SIP-based web client for HTTP session mobility and multimedia services

This work leverages Session Initiation Protocol (SIP) transportation and mobility mechanism to transfer session data between two Web browsers. In addition, a Web browser can now act as an adaptive User Agent Client to surf the Internet and make voice calls as a SIP client. It is a novel work that uses SIP to transfer session data between Web browsers and borrows SIP Mobility types to introduce new service namely, content sharing and session hand-off, to the Web browsing experience. Referred to as a SIP-based HTTP session mobility service, it offers personal mobility to end users, and facilitates session mobility in Web browsing. While content sharing refers to the ability to view the same Web resource on two Web browsers and does not require moving session data, session hand-off refers to the migration of a Web session with its session data (cookies, hidden form elements and rewritten URL) to another Web browser. Results showed that the integration of SIP into a Web browser does not degrade the performance of a Web browser. Results also showed that the service could not work on all websites because of the Same Origin Policy (SOP) used by Web browsers to transfer cookies. The hybrid-based architectural scheme proposed and implemented here is compared with other existing Web session migration schemes. On the service commercialization, if the privacy and security of session data could be guaranteed by the implementers, a flat rate could be periodically charged regardless of the varying session data sizes. In another sense, it could be rendered as a Value Added Service (VAS) to customers.

Mapping Third Party Call Control and Session Handoff in SIP Mobility to Content Sharing and Session Handoff in the Web Browsing Context

This paper presents a new service to web browsing. The service, referred to as Session Handoff and Content Sharing between two web browsers, requires extending the capabilities of existing web browsers by integrating a Session Initiation Protocol (SIP) stack into them. An optional SIP Application Server (SIP AS) that co-ordinates HTTP session mobility between web browsers is introduced. In addition, Third-party Call Control and Session Handoff in SIP Session Mobility are successfully mapped to Content Sharing and Session Handoff between two web browsers, respectively. While content sharing refers to the ability to view same web resource on two web browsers, session handoff refers to the ability to migrate a web session between two web browsers. Until now, SIP had only been integrated into application servers. Integrating a SIP stack into a web browser has helped improve collaboration and mobility among the web users. In addition, it encourages developing adaptive User Agent Clients (UACs) that can serve two or more purposes. In this case, a web browser can also be used as a SIP client to make voice calls and be extended to perform other SIP functionalities.

Control Services for the HTTP Session Mobility Service

In our previous work, a hybrid architectural scheme was proposed to migrate web sessions between PCs. Only the client-side of the work was however achieved. The work included developing a Web browser extension and creating a new service, namely content sharing and session handoff. This further research explores the proxy-side of the architecture. It entails creating new services that prevent abuse of the services offered by the client. That is, the new services in the proxy control the interaction between the Web browsers. The services are provided to further improve the Webbrowsing experience and are derived from existing telecommunication services, such as call transfer and call blocking. Although the services - call transfer and call blocking - are peculiar to telecommunications, they are feasible in the Web-browsing context owing to identities in Web browsers, which could make them interact with one another like phones. Our approach uses the best of the Internet protocols (SIP, XML, HTTP) to provide a controllable HTTP session mobility service between two or more Web browsers.

Converged multimedia services in emerging Web 2.0 session mobility scenarios

The increasing request for converged multimedia services have motivated relevant standardization efforts, such as the Session Initiation Protocol (SIP) to support session control, mobility, and interoperability in all-IP next generation wireless networks. Notwithstanding the central role of SIP in novel converged multimedia, the potential of SIP-based service composition for the development of new classes of Web 2.0 services able to interoperate with existing HTTP-based services is still widely unexplored. The paper proposes an original solution to improve online user experience by integrating a SIP stack into the Web browser, thus enabling the execution of novel SIP-based applications directly at the client endpoint. In particular, our browser extension coordinates with our novel SIP-based Converged Application Server to enable session mobility and prevent abuses of the services available in the client. Experimental results show that our SIP-based solution is feasible in most common Internet deployment scenarios and enables session mobility with limited management cost.

Extending web browsers architectures to support HTTP session mobility

In a bid to extend web browsers capabilities, a new extension that transfers HTTP session between two web browsers also knows as User Agents will be developed. The capabilities of these web browsers will be extended by integrating a Session Initiation Protocol (SIP) stack into them. SIP has been chosen because it has clearly defined session mobility types namely Third-party Call Control and Session Hand-off. This paper identifies the modifications that will be made to the present-day web browsers architectures and describes the two services that can be provided namely content sharing and session transfer between any two user agents.

Determining the signalling overhead of two common WebRTC methods: JSON via XMLHttpRequest and SIP over WebSocket

Web Real-Time Communication (WebRTC) introduces real-time multimedia communication as native capabilities of Web browsers. With the adoption of WebRTC the Web browsers will be able to use WebRTC to communicate with one another (peer-to-peer), and with WebSocket servers such as Mobicents SIP Servlets and other server technologies that support WebSocket communication to enable SIP-to-WebRTC communication. This paper outlines the potential of WebRTC and discusses the two common methods of doing real-time communication in Web browsers through WebRTC. The methods are JavaScript Object Notation (JSON) via XMLHttpRequest (XHR) and Session Initiation Protocol (SIP) via WebSocket. A three-user WebRTC video chat prototype application was developed and used to evaluate both methods. Additional signalling overhead introduced into a browser by each method was determined. The results showed WebRTC-SIP/WS has more overhead than WebRTC-JSON/XHR. These signalling overhead findings are useful in that they could help application developers make decision on their choice of technologies and protocols when developing WebRTC-supported applications.

CAS: A SIP-based proxy for the provisioning of HTTP session mobility

On one hand, the convergence of the Internet, Telecommunication and Broadcasting is generating new services over the Internet, and on the other hand, the online experience is getting improved via the introduction of user profile mobility and web session mobility. While different architectural schemes for HTTP session mobility have been proposed and implemented in the academia, the industry has introduced user profile mobility solutions, such as Mozilla Weave and Google Browser Sync. This paper presents our effort to improve the web browsing experience via a SIP-based web session mobility service. It discusses the client and proxy features of our session mobility system. The system architecture is service-oriented and easy to integrate in any existing SIP-based deployment environment. Experimental results about CPU usage, memory consumption, and number of requests processed at the proxy side confirm the feasibility of our solution from a service provider perspective in all most common Internet deployment scenarios.

Emerging research areas in SIP-based converged services for extended Web clients

The convergence of Next Generation Networks and Internet-based rich applications are generating relevant industrial opportunities in the market of mobility-enabled services. Even if this trend is widely recognized, there are still a few industrial-level solutions that effectively support session mobility in a transparent way and with the capability of openly integrating with existing and legacy applications. In this paper we propose a SIP-based hybrid architecture for Web session mobility that offers content sharing and session handoff between Web browsers. In addition, its technical originality includes integrating a SIP stack into a Web browser, thus offering the advantage of extending a Web browser to act as a SIP client. Lastly, a rich set of control services that prevent abuse of content sharing and session handoff are introduced into the proposed system. The implemented solution uses SIP in a standard way to migrate Web sessions between Web browsers; it is made up of a SIP integrated Web client and a converged (SIP and HTTP) Application Server that can be easily used to enable session mobility in any kind of Web-based application. In addition, the implemented system has recently evolved to a framework for developing different kinds of converged services over the Internet, analogously to what is possible with Google Wave and the existing telephony APIs. Finally, the paper reports the evaluation of the proposed framework and of the employed technologies, together with directions of future work, in terms of both extension to other application domains and exploration of research areas/models that can benefit form the adoption of SIP and Web-related solutions.

Routing Cost and Latency in the VillageTelco Wireless Mesh Network

Afrimesh is the Network Management System (NMS) used in the VillageTelco (VT) project. The NMS uses both Simple Network Management Protocol (SNMP) and Internet Control Message Protocol (ICMP) as its network management protocols. It provides a thorough report on network health, such as interference, noise level and latency. In addition, it displays the current network topology and can run at a node thereby making it efficient to identify compromised or isolated node(s). This article presents the performance of a typical mesh network. The performance metrics include signalling overhead and network latency of a small scale mesh network with few nodes. Experiments showed that an additional hop in a network increases the network latency by averagely 35ms. In addition, the signalling overhead of a VT network increases with time at every node. However, the footprint is small and would not impede the performance of a network in a large-scale deployment.