draft-ietf-sipping-kpml-03

This document defines a framework and the requirements for call control and multi-party usage of the Session Initiation Protocol (SIP). To enable discussion of multi-party features and applications, we define an abstract call model for describing the media relationships required by many of these. The model and actions described here are specifically chosen to be independent of the SIP signaling and/or mixing approach chosen to actually set up the media relationships. In addition to its dialog manipulation aspect, this framework includes requirements for communicating related information and events such as conference and session state and session history. This framework also describes other goals that embody the spirit of SIP applications as used on the Internet such as the definition of primitives (not services), invoker and participant oriented primitives, signaling and mixing model independence, and others. Status of This Memo This document is not an Internet Standards Track specification; it is published for informational purposes. This document is a product of the Internet Engineering Task Force (IETF). It represents the consensus of the IETF community. It has received public review and has been approved for publication by the

In telecommunications, audio signaling is the use of the audio channel for signaling and user-interface purposes. When features use audio signaling, and are assembled in a pipesand-filters configuration, there is a potential for undesirable feature interactions. This paper analyzes the potential feature interactions. It proposes a method for eliminating some of them, as well as directions for future work on the remaining interactions. The method can be implemented in SIP, using compositional patterns of signaling that will work correctly regardless of how many features are active.

Abstract — Providing emergency services in VoIP networks is vital to the success of VoIP. It not only presents design and implementation challenges, but also gives an opportunity to enhance the existing emergency call handling infrastructure. We propose an architecture to deliver emergency services in SIP-based VoIP networks, which can accommodate PSTN calls through PSTN to SIP gateways. Our architecture addresses the issues of identifying emergency calls, determining callers’ locations, routing emergency calls to appropriate public safety access points (PSAP), and presenting required information to emergency call takers. We have developed a prototype implementation to prove our architecture’s feasibility and scalability. We expect to undertake a pilot project at a working PSAP with our implementation once it is thoroughly tested. I.

Abstract. Third-party call control (3PCC) is essential to implementing advanced services in Voice-over-IP (VoIP) networks. It allows intermediary applications to control how the media streams of endpoint devices are connected together. However the Session Initiation Protocol (SIP), the widely adopted open standard for VoIP signaling, presents a number of challenges that make 3PCC in SIP complex and prone to errors. Previously proposed solutions only address operations under ideal conditions and thus are incomplete. Furthermore, the problem of compositional 3PCC by multiple applications have not been addressed. In this paper, we propose a general solution for robust and comprehensive media connectivity control. The solution has been verified, and allows multiple applications operating concurrently in a call path to interoperate successfully. 1

live multimedia broadcasts (e.g., for applications such as TV, radio, or e-cinema) to a potentially large number of mobile hosts that roam about in a wireless internet with hotspots. We take a user-oriented approach based on an application-level delivery infrastructure consisting of and managed by (value-added) service providers. The service providers are mobility-aware and offer broadcasts in configurations that are optimized for wireless links and mobile hosts. In hotspots, mobile hosts may be able to simultaneously reach several localized service providers through different interfaces. Within this context, we present the design of a lightweight application-level protocol that enables mobile hosts to select a service provider from which they want to receive a broadcast. Mobile hosts use the protocol to begin receiving a broadcast and to remain connected to the same logical broadcast as they move across subnets. The protocol is independent of the actual stream control protocol (e.g., RTSP) that service providers might use. We show how our protocol can be realized with the existing protocols SIP and SDP. The realization uses SIP in combination with SDP's offer-answer model in a new way.

A significant deterrent to the ability to connect in a spontaneous manner in cross-enterprise collaborative applications is the difficulty in users from different domains being able to access resources or services located and owned by other entities. Coalition access control encompasses control mechanisms dealing with access between users of two or more different security domains. In this paper we present an approach to add contextual information to the distributed Role Based Access Control (dRBAC) model to support spontaneous coalition. The dRBAC model is a relatively new approach for coalition access control based on a delegation model but has been targeted towards pre-arranged delegations among distributed enterprises.A delegation architecture is presented that leverages SIP communication sessions to discover delegation security managers that can automatically exchange roles and delegations based on location and communication session contexts. 1.

Abstract—We present a location-based, ubiquitous service architecture, based on the Session Initiation Protocol (SIP) and a service discovery protocol that enables users to enhance the multimedia communications services available on their mobile devices by discovering other local devices, and including them in their active sessions, creating a “virtual device. ” We have implemented our concept based on Columbia University’s multimedia environment and we show its feasibility by a performance analysis. Index Terms—Internet multimedia, mobile communications, ubiquitous computing, location-based services

draft-ietf-sip-hitchhikers-guide-00

The IP Multimedia Subsystem (IMS) has been recognized as a reference next generation network architecture for offering multimedia services over an Internet Protocol (IP) based infrastructure. One of the key benefits of the IMS is efficient and flexible introduction of new services and access to third-party application providers, thanks to standard interfaces and standardized service capabilities. In order to support novel media-rich applications across a wide range of user devices and access networks, the IMS must support negotiable quality of service (QoS) for IP multimedia sessions. In this paper we describe the application-level QoS signaling as specified by the 3GPP, and propose some enhancements based on advanced QoS parameter matching and optimization functionality to be included along the signaling path. We outline various signaling flow scenarios and discuss them in the context of a case study involving an IMS-supported 3D virtual environment, featuring a Treasure-Hunt-like game. I.