ALEX: Improving SIP Support in Systems with Multiple Network Addresses

The successful and increasingly adopted session initiation protocol (SIP) does not adequately support hosts with multiple network addresses, such as dual-stack (IPv4-IPv6) or IPv6 multi-homed devices. This paper presents the Address List Extension (ALEX) to SIP that adds effective support to systems with multiple addresses, such as dual-stack hosts or multi-homed IPv6 hosts. ALEX enables IPv6 transport to be used for SIP messages, as well as for communication sessions between SIP user agents (UAs), whenever possible and without compromising compatibility with ALEX-unaware UAs and SIP servers

Delivering Live Multimedia Streams to Mobile Hosts in a Wireless Internet with Multiple Content Aggregators

We consider the distribution of channels of live multimedia content (e.g., radio or TV broadcasts) via multiple content aggregators. In our work, an aggregator receives channels from content sources and redistributes them to a potentially large number of mobile hosts. Each aggregator can offer a channel in various configurations to cater for different wireless links, mobile hosts, and user preferences. As a result, a mobile host can generally choose from different configurations of the same channel offered by multiple alternative aggregators, which may be available through different interfaces (e.g., in a hotspot). A mobile host may need to handoff to another aggregator once it receives a channel. To prevent service disruption, a mobile host may for instance need to handoff to another aggregator when it leaves the subnets that make up its current aggregator�s service area (e.g., a hotspot or a cellular network).\ud In this paper, we present the design of a system that enables (multi-homed) mobile hosts to seamlessly handoff from one aggregator to another so that they can continue to receive a channel wherever they go. We concentrate on handoffs between aggregators as a result of a mobile host crossing a subnet boundary. As part of the system, we discuss a lightweight application-level protocol that enables mobile hosts to select the aggregator that provides the �best� configuration of a channel. The protocol comes into play when a mobile host begins to receive a channel and when it crosses a subnet boundary while receiving the channel. We show how our protocol can be implemented using the standard IETF session control and description protocols SIP and SDP. The implementation combines SIP and SDP�s offer-answer model in a novel way

A new scheme to reduce session establishment time in session initiation protocol (SIP)

The session Initiation Protocol (SIP) has been developed by Internet Engineering Taskforce standard (IETF) with the main purpose of establishing and managing sessions between two or more parties wishing to communicate. SIP is a signaling protocol which is used for the current and future Internet Protocol (IP) telephony services, video services, and integrated web and multimedia services. SIP is an application layer protcol, thus it can run over Transmission Control Protocol(TCP) or User Datagram Protocol (UDP). When the packets are sent over the network, a form of congestion control mechanism is necessary to prevent from network collapse. TCP is a reliable protocl and provides the congestion control by adjusting the size of the congestion windows. UDP is an unreliable protocol and no flow control mechanism is provided. Many applications of the Internet require the establishment and management of sessions. The purpose of the thesis is to study the session establishnment procedure in SIP and try to reduce the time taken for the session setup in two different conditions. One, when there is no congestion in the network, and the other is when there is a network congestion. We have simulated the behaviour of session establishment in SIP using Network Simulator (NS2). UDP is used as the transport protocol. We have created different network topologies. In the topology we had created SIP user agents who wants to communicte, proxy servers for forwarding the requests on behalf of the user agents, and a Domain Name Server (DNS) which maintains the location information of all proxy servers. We tried to reduce the time taken for the session establishment. As UDP does not provide any congestion control mechanisms, we used the binary exponential backoff (BEB) algorithm to set the timers. In our network topolgy when there is no packet loss in the network, the time taken for the session establishment is reduced from 0.86 sec to 0.574 sec. In case of network congestion the setup time is reduced from 4.55 sec to 2.86 sec. From the simulation, we conclude that the session establishment time can be reduced by reducing the number of message exchanges required for session setup

Interworking Architectures in Heterogeneous Wireless Networks: An Algorithmic Overview

The scarce availability of spectrum and the proliferation of smartphones, social networking applications, online gaming etc., mobile network operators (MNOs) are faced with an exponential growth in packet switched data requirements on their networks. Haven invested in legacy systems (such as HSPA, WCDMA, WiMAX, Cdma2000, LTE, etc.) that have hitherto withstood the current and imminent data usage demand, future and projected usage surpass the capabilities of the evolution of these individual technologies. Hence, a more critical, cost-effective and flexible approach to provide ubiquitous coverage for the user using available spectrum is of high demand. Heterogeneous Networks make use of these legacy systems by allowing users to connect to the best network available and most importantly seamlessly handover active sessions amidst them. This paper presents a survey of interworking architectures between IMT 2000 candidate networks that employ the use of IEFT protocols such as MIP, mSCTP, HIP, MOBIKE, IKEV2 and SIP etc. to bring about this much needed capacity

Signaling for Internet Telephony

Internet telephony must offer the standard telephony services.However, the transition to Internet-based telephony services also provides an opportunity to create new services more rapidly and with lower complexity than in the existing public switched telephone network(PSTN). The Session Initiation Protocol (SIP) is a signaling protocol that creates, modifies and terminates associations between Internet end systems, including conferences and point-to-point calls. SIP supports unicast, mesh and multicast conferences, as well as combinations of these modes. SIP implements services such as call forwarding and transfer, placing calls on hold, camp-on and call queueing by a small set of call handling primitives. SIP implementations can re-use parts of other Internet service protocols such as HTTP and the Real-Time Stream Protocol (RTSP). In this paper, we describe SIP, and show how its basic primitives can be used to construct a wide range of telephony services

The Virtual Device: Expanding Wireless Communication Services Through Service Discovery and Session Mobility

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

Study, Analysis and Modeling of IP Multimedia systems on next generation networks providing mobile and fixed multimedia services

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