Session Initiation Protocol (SIP) is being widely adopted for VoIP, IM and other collaborative applications due to its simple yet rich functional design. However, one of the main drawbacks has been its per-application deployment (each application using its own SIP stack), leading to narrowly focussed development of SIP based services. In this paper, we propose a client-side SIP service and supporting network infrastructure blocks that provide unified mechanisms to execute generic SIP functions. The composition of these building blocks allows for creating richer applications, e.g. a conferencing server coupled with a gaming server provides dynamic conferencing between current occupants of a game room. The main feature of our framework is its availability to all applications including the ones not inherently based on SIP. Also, the SIP service API is designed to be extensible and in addition to providing novel higher level functional primitives like adhoc conferencing and seamless transition of sessions, it also exports a low level interface for specialized applications. Another feature of the service is that it allows a user to plug-in an end device of his/her choice on a per-session basis. We demonstrate the richness of the API by describing prototypes for enhancing various applications as well as new converged applications.

The public switched telephone network (PSTN) provides ubiquitous availability and very high scalability of more than a million busy hour call attempts per switch. If large carriers are to adopt Internet telephony, then Internet telephony servers should offer at least similar quantifi-able guarantees for scalability and reliability using metrics such as call setup latency, server call handling capacity, busy hour call arrivals, mean-time between failures and mean-time to recover. This thesis presents a reliable, scalable and interoperable Internet telephony architecture for user registration, call routing, conferencing and unified messaging using commodity hardware. The results extend beyond Internet telephony to encompass multimedia communication in general. The architecture presented in this thesis deals with two aspects: at least PSTN-grade re-liability and scalability of the Internet telephony servers, and interoperable Internet telephony services such as conferencing and voice mail using existing protocols. We describe the archi-tecture and implementation of our Session Initiation Protocol (SIP)-based enterprise Internet telephony architecture known as Columbia InterNet Extensible Multimedia Architecture (CIN-EMA). It consists of a SIP registration and proxy server, a multi-party conferencing server, a gateway for interworking SIP with ITU’s H.323, an interactive voice response system and a

In this paper we survey the conferencing architectures in H.323 and SIP. We give a compact review of the H.323 framework and capture the current status of the emerging SIP conferencing architecture. Subsequently we present a qualitative comparison of the schemes under study. The evaluation criteria are scalability to large conference sizes and to large geographical dispersion of participants ,as well as adaptiveness to heterogeneity in media streams format and protocol implementation. Our findings suggest that both architectures have certain advantages and limitations. Wherever possible we propose measures to overcome the latter. Concluding, we propose future work to mitigate the identified gaps in the SIP conferencing framework.

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In traditional conferencing systems, participants have little or no privacy, as their voices are by default shared with all others in a session. Such systems cannot offer participants the options of muting and deafening other members. The concept of narrowcasting can be applied to make these kinds of filters available in multimedia conferencing systems. Our system treats media sinks (in the simplest case, listeners) as full citizens, peers of the media sources (conversants ’ voices), and we defined therefore duals of mute & select: deafen & attend, which respectively block a sink or focus on it to the exclusion of others. In this article, we describe our prototyped application, which uses existing standard Session Initiation Protocol (SIP) methods to control fine-grained narrowcasting sessions. The runtime system considers the policy configured by the participants and provides a policy evaluation algorithm for media mixing and delivery. We have integrated a “virtual reality”-style interface with this SIP backend to display and control articulated narrowcasting with figurative avatars.

Push-to-Talk (PTT) is a useful capability for rapidly deployable wireless mesh networks used by first responders. PTT allows several users to speak with each other while using a single, half-duplex, communication channel, such that only one user speaks at a time while all other users listen. This paper presents the architecture and protocol of a robust distributed PTT service for wireless mesh networks. The architecture supports any 802.11 client with SIP-based (Session Initiation Protocol) VoIP software and enables the participation of regular phones. Collectively, the mesh nodes provide the illusion of a single third party call controller, enabling clients to participate via any reachable mesh node. Each PTT group instantiates its own logical floor control manager that is highly available and resilient to mesh connectivity changes such as node crashes and recoveries and network partitions and merges. Experimental results on a fully deployed mesh network consisting of 14 mesh nodes and tens of emulated clients demonstrate the scalability and robustness of the system.

For at least the next decade, we foresee two classes of packet communication networks existing side-byside and dominating the data communications scene:

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