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

Conveying and Handling Location Information in the IP Multimedia Subsystem

The IP Multimedia Subsystem (IMS), specified by the 3rd Generation Partnership Project (3GPP), is a key element in the next-generation network (NGN) converged architecture. Extending the IMS towards provisioning support for location based services (LBS) will enable enhanced services and offer new revenues to the operator. Conveying location information in the IMS and connecting the IMS with a positioning system are still open issues. This paper presents the design and implementation of an IMS Location Server (ILS) integrating IMS with a positioning system. From the IMS perspective, the ILS serves as a service enabler for LBS. In order to demonstrate proof-of-concept in enhancing IMS-based services, two prototype service scenarios have been implemented: Location-aware Messaging (LaM), and Location-aware Push-to-Talk over cellular (LaPoC). Some work has been done by the IETF in the area of location information transport based on the Session Initiation Protocol (SIP). This paper proposes improvements in this area, primarily related to reducing the necessary amount of signaling with the specification of a new type of location filter. We have conducted measurements in a laboratory environment in order to illustrate our proposed solution and verify the benefits compared to existing solutions in terms of traffic load and session establishment time. Furthermore, we present a case study integrating the ILS with the Ericsson Mobile Positioning System (MPS)

Cooperative announcement-based caching for video-on-demand streaming

Recently, video-on-demand (VoD) streaming services like Netflix and Hulu have gained a lot of popularity. This has led to a strong increase in bandwidth capacity requirements in the network. To reduce this network load, the design of appropriate caching strategies is of utmost importance. Based on the fact that, typically, a video stream is temporally segmented into smaller chunks that can be accessed and decoded independently, cache replacement strategies have been developed that take advantage of this temporal structure in the video. In this paper, two caching strategies are proposed that additionally take advantage of the phenomenon of binge watching, where users stream multiple consecutive episodes of the same series, reported by recent user behavior studies to become the everyday behavior. Taking into account this information allows us to predict future segment requests, even before the video playout has started. Two strategies are proposed, both with a different level of coordination between the caches in the network. Using a VoD request trace based on binge watching user characteristics, the presented algorithms have been thoroughly evaluated in multiple network topologies with different characteristics, showing their general applicability. It was shown that in a realistic scenario, the proposed election-based caching strategy can outperform the state-of-the-art by 20% in terms of cache hit ratio while using 4% less network bandwidth

Multimedia session continuity in the IP multimedia subsystem : investigation and testbed implementation

Includes bibliographical references (leaves 91-94).The advent of Internet Protocol (IP) based rich multimedia services and applications has seen rapid growth and adoption in recent years, with an equally increasing user base. Voice over IP (VoIP) and IP Television (IPTV) are key examples of services that are blurring the lines between traditional stove-pipe approach network infrastructures. In these, each service required a different network technology to be provisioned, and could only be accessed through a specific end user equipment (UE) technology. The move towards an all-IP core network infrastructure and the proliferation of multi-capability multi-interface user devices has spurred a convergence trend characterized by access to services and applications through any network, any device and anywhere

Development of an Overload Management Model in IMS

Abstract— The concept of IP Multimedia Subsystem (IMS) describes a new network architecture, the main element of which is a packet transport network that supports all access technologies and provides implementation of a large number of infocommunication services. In this paper, formalization of the task of balancing the load of IMS virtual subsystems

An Architecture for Global Distributed SIP Network Using IPv4 Anycast

Tato diplomová práce se zabývá metodami pro výběr nejbližší RTP proxy k VoIP klientům s použitím IP anycastu. RTP proxy servery jsou umístěny v síti Internetu a přeposílají RTP data pro VoIP klienty za síťovými překladači adres(NAT). Bez zeměpisně rozmístěných RTP proxy serverů a metod pro nalezení nejbližšího RTP proxy serveru by došlo ke zbytečnému poklesu kvality přenosu médialních dat a velkému zpoždení. Tento dokument navrhuje 4 metody a jejich porovnání s podrobnějšími rozbory metod s využitím DNS resolvování a přímo SIP protokolu. Tento dokument také obsahuje měření chování IP anycastu v porovnání mezi metrikami směrování a metrikami časovými. Nakonec dokumentu je také uvedena implemetace na SIP Express Router platformě.This thesis is about using IP anycast-based methods for locating RTP proxy servers close to VoIP clients. The RTP proxy servers are hosts on the public Internet that relay RTP media between VoIP clients in a way that accomplishes traversal over Network Address Translators (NATs). Without geographically-dispersed RTP proxy servers and methods to find one in client's proximity, voice latency may be unbearably long and dramatically reduce perceived voice quality. This document proposes four methods their comparison with further design of DNS-based and SIP-based methods. It includes IP anycast measurements that provides an overview of IP anycast behaviour in terms of routing metrics and latency metrics. It also includes implementation on SIP Express Router platform.