Architecture and testbed implementation of vertical handovers based on SIP session border controllers

This paper describes a Session Initiation Protocol (SIP) based solution for mobility management that provides seamless mobile multimedia services in a heterogeneous scenario where different radio access technologies are used (802.11/ WiFi, Bluetooth, 2.5G/3G networks). The solution relies on the so called “Session Border Controllers” which are now widely used in many commercial SIP telephony solutions, mainly to deal with NAT traversal. Session Border Controller functionality has been extended to support seamless mobility for multimedia applications. A prototype of the proposed solution focused on VoIP services has been implemented in a test bed which is able to perform seamless handovers (and NAT traversal) using the 802.11, Bluetooth and 3G (UMTS) access networks. Measurements results are reported which analyze the performance of the solution in a real world environment, using commercial WiFi and 3G services.     

The 3G-324M protocol for conversational video telephony

As mobile operators worldwide migrate to third-generation (3G) networks, conversational video-telephony services are becoming a key differentiator between new 3G offerings and existing 2G/2.5G services. Although it's possible to have limited video-based services - such as a multimedia messaging service - that deliver pictures and video clips over 2.5G services, these are delay-insensitive applications that could run over a packet-based wireless network like general packet radio service (GPRS) or code division multiple access (CDMA)'s 1XRTT. For delay-sensitive applications such as conversational video telephony, present 3G packet bearers are inadequate, and the Third Generation Partnership Project (3GPP; http://www.3gpp.org) mandates using the 3G bandwidth-guaranteed circuit-switched bearer and the 3G-324M system. The 3G-324M system is a derivative of the International Telecommunication Union (ITU) H.324 protocol standard for low-bitrate multimedia communication, which ITU-T developed for the public switched telephone network (PSTN). This article describes the 3G-324M system, which has been adopted by both 3GPP and 3GPP2 (htpp://www.3gpp2.org), as well as its H.324 roots.     

Wireless loops: What are they?

Several loop applications of wireless technology are aimed at reducing the cost of deploying communications services ranging from telephone to wideband video. In these applications, wireless links replace a portion of a wireline loop from a central location (a central office or cable headend) to a subscriber. The replacement of labor-intensive wireline technology by complex mass-produced integrated electronics in wireless transceivers is projected to reduce the overall cost of the resulting loop. These wireless loop applications attempt to provide existing communications services or small modifications to existing communications services. A different interpretation of a wireless loop makes use of low-power digital radio technology to provide the last thousand feet or so of a loop. Low-power low-complexity wireless loop technology in small base units can be integrated with network intelligence to provide the fixed-infrastructure network needed to support economical personal communications services (PCS) to small, lightweight, low-power personal voice and/or data communicators. Low-complexity communicators can provide many hours of “talk time” or data transmission time and perhaps several days of standby time from small batteries (≤ 1.5 oz). Because this application of wireless loop technology can reduce the inherent costs in several parts of a wireline loop, it has the potential to provide convenient widespread PCS at less costs than providing telephone services over conventional wireline loops. This low-power wireless loop application does not fit into any existing communications system paradigm. Wireless technology with tetherless access and wide-ranging mobility, e.g., the personal access communications system (PACS), does not fit the accumulated wisdom of the wireline telephony paradigm. It also does not fit the paradigm of existing cellular radio that has sparsely distributed expensive cell sites, and it is not targeted at fixed video services as is wireless cable. Because a significant change in thinking is required in addressing this new low-power low-complexity widespread wireless loop paradigm, its large economic advantages and service benefits have not yet been embraced by many of the existing communications providers, who appear to be more comfortable pursuing the better-known paradigms of video using wireless cable, or of cellular radio in the guise of high-tier PCS, or in the guise of rapid economical deployment of telephone services in developing nations. This paper discusses the inherent economic advantages and service benefits of low-power low-complexity wireless loop technology integrated with network intelligence aimed at providing economical low-tier PCS to everyone.     

An introduction to TSAPI and network telephony

Discusses the integration of the telephone into the desktop work environment. This objective can be achieved in a number of different ways. The approach described uses the client-server model. Network telephony is a service that provides the capability to establish, answer, route, and terminate telephone calls under the control of applications on either desktop computers or servers resident in the network. It is accessed by applications via a standard programming interface (telephony services application programming interface, TSAPI) and utilizes a centralized server-based interface with the private branch exchange (PBX) to access the switching and telephone control services that the PBX provides. Creating a logical control link between the application software on the desktop computer and the telephone on the desk eliminates the need to physically connect the telephone to the desktop computer. The author distinguishes between the application programming interface and the network telephony service provider. The author assumes the services are provided by NTS R2.2 release and a PBX-based switching service. However, the author uses the term PBX to include switching services provided by key telephone or hybrid systems, PC-based telephony cards, or other appropriate technologies     

An expandable time-division circuit switching LSI and networkarchitecture for broadband ISDN

A broadband network architecture is proposed that integrates multimedia services, such as data, video, and telephony information, using 52-Mb/s based STM-paths at the user network interface (UNI). The user can access any new service via the STM-based access network via either synchronous transfer mode (STM) switching or asynchronous transfer mode (ATM) switching. STM circuit switching supports long duration, constant bandwidth data transfer services such as video and high-definition television (HDTV) distribution and will also be used for the crossconnect system. Circuit switching can provide transparent transmission during long connection periods. This paper also proposes an expandable time-division switch architecture, an expandable time-division switching LSI, and an expandable switching module for small to large size system applications. The proposed time-division switching LSI, module, and system handle 52-Mb/s bearer signals and have throughputs of 2.4 Gb/s, 10 Gb/s, and 40 Gb/s, respectively. The time-division switch realizes video distribution with 1:n connections. Finally, a local switching node that features an expandable 52-Mb/s time-division circuit switching network is shown for multimedia access networking     

Performance of Video Telephony Services in UMTS using Live Measurements and Network Emulation

While speech services in mobile communication systems are investigated quite well, video telephony services are relatively novel in this sector. Because of the market penetration of camera equipped mobile phones the video telephony service is expected to become a widely used service. In this article an introduction of the protocols used for video telephony in UMTS is given. Concepts for the performance evaluation in terms of video (PEVQ) and audio quality (PESQ) are presented and utilized. Evaluations are performed by both live measurements and network emulation. The results show that there is quite potential for improvements related to video telephony in UMTS in terms of video quality and channel setup time. Finally, an improved radio bearer configuration is provided which aims at a better integration of video telephony services into the UMTS architecture.     

Location Management Strategies for Cellular Overlay Networks—A Signaling Cost Analysis

In a world driven by the accessibility of data for business, for communication and entertainment, but one populated by an increasingly nomadic society, it is no surprise that the appetite for wireless services continues to grow. The limitations of existing technologies however, particularly for wireless multi-casting and broadcasting, suggest that new approaches are needed to allow users to access services in the all wireless world. A popular approach for improving wireless services proposes to equip mobile terminals (MTs) with more than one radio interface to enable access to services in an always best connected paradigm. One such architecture envisages a co-operating overlay network, or inter-network, that comprises of a cellular telephony network and a digital broadcasting network. MTs in the inter-network have two corresponding radio interfaces and co-operation between the networks provide the user with seamless access to diverse services across the inter-worked platform. In this paper, we introduce novel location management proposals for co-operating overlay networks that incorporate a uni-directional broadcast overlay and provide a cost and latency analysis of the schemes. Our cost analysis demonstrates the soundness of the techniques and the value of the analyses as comparative evaluation tools. In particular, it is shown that there are a wide range of realistic operating conditions where the novel proposals presented here provide an optimum performance in cost terms.     

Evaluation of header compression schemes for IP-based wireless access systems

For mobile IP-based telephony (voice over IP) and IP-based real-time multimedia over cellular radio systems, an economically viable solution is needed. It is an absolute requirement that, for example, the 60-octet IPv6/UPD/RTP headers on IP telephony packets be reduced in size to conserve bandwidth in the radio spectrum. We evaluate the performance of two header compression schemes, based on RFCs 2508 and 3095, under the conditions of cellular radio access technology. The results presented in this article refer to voice and Web browsing traffic and are based on the implementation of compression/decompression algorithms for the aforementioned standards. We find that RFC 3095 performs significantly better if used for mobile communications.     

Using PARLAY APIs Over a SIP System in a Distributed Service Platform for Carrier Grade Multimedia Services

The implementation of new mobile communication technologies developed in the third generation partnership project (3GPP) will allow to access the Internet not only from a PC but also via mobile phones, palmtops and other devices. New applications will emerge, combining several basic services like voice telephony, e-mail, voice over IP, mobility or web-browsing, and thus wiping out the borders between the fixed telephone network, mobile radio and the Internet. Offering those value-added services will be the key factor for success of network and service providers in an increasingly competitive market. In 3GPP's service framework the use of the Parlay APIs is proposed that allow application development by third parties in order to speed up service creation and deployment. 3GPP has also adopted SIP for session control of multimedia communications in an IP network. This article proposes a mapping of SIP functionality to Parlay services and describes a prototype implementation using the SIP Servlet API. Furthermore, an architecture of a Service Platform is presented that offers a framework for the creation, execution and management of carrier grade multimedia services in heterogeneous networks.     

Second generation cordless (CT-2) telephony in the UK: telepointservices and the common air-interface

The background to second generation cordless telephony (CT-2) is set out to show the relationship between telepoint and cellular radio services. The physical layer of the telepoint air-interface is described with justification of the technical choices made in working towards the common air-interface which will supercede initial proprietary standards. The various channels and signalling multiplexes are then discussed in sufficient detail to permit understanding of the procedures by which portable terminals establish connections through base stations to the public switched network. The author concludes with an optimistic projection of the future for telepoint services in the UK and other European countries     

Radio interfaces make the difference in 3G cellular systems

Third-generation cellular telephony is on its way-not, unfortunately, as a single worldwide system, but as three incompatible ones. The main difference between the three lies in their choice of radio interface technology. This fact is crucial for several reasons, since the radio interface determines not only the fundamental capacity of a mobile radio network, but also how it deals with such issues as interference, multipath distortion, and handing off calls from one base station to another as users move around. Consequently, as might be expected, the choice of radio interface has a dramatic effect on the complexity of the system and its cost. Also, global travellers will need more than one phone with which to communicate, at least until trimode phones reach the market. To understand what is being developed, and why, the author begins with one of the stated goals of third-generation (3G) systems, namely to support variable user data rates as high as 2 Mb/s. In one way or another, all three approaches provide for adaptive bandwidth-on-demand. Two of the systems use wideband code-division multiple access (WCDMA) for the radio interface. The other uses two variations of time-division multiple access (TDMA)     

Developing applications for Internet telephony: a case study on the use of parlay call control APIs in SIP networks

Applications offered to end users as value-added services, or more simple, services, are crucial for the survival and success of service providers. Two main sets of standards have emerged for Internet telephony: H.323 from the ITU-T and SIP from the IETF. Unfortunately, the related application development frameworks are rather weak. Parlay, a set of standard object-oriented and signaling protocol-neural APIs, is an alternative. It allows applications to access network functionality, including call control, in a controller manner. Call control makes it possible to establish, modify, and tear down calls. It is the main functionality offered by Internet telephony networks. We have built a call control application in a SIP environment, using the call control APIs offered by Parlay. The application is a multiparty game. This article describes the case study. The mapping of the APIs onto SIP is presented, and its implementation is described. Related work reviewed, and the lessons learned are discussed. Parlay call control APIs are suitable for application development in Internet telephony. However, well isolated extensions are needed to realize their full potential.     

ISDN computer-aided telephony

It is argued that ISDN computer-aided telephony requires properly architected platforms to satisfy changing application needs during the 1990s. Proper architecting necessitates the use of functionally rich and consistent telephony application programming interfaces (APIs). Other APIs are also needed to support integrated applications. The coexistence of telephony and other APIs must be accommodated in the ISDN driver architecture to make efficient use of D-channel signaling and voice, data, or image communications on the associated B/H channels. This driver may support Open Systems Interconnection (OSI), Systems Network Architecture (SNA), X.25, or other protocol stacks in the same computer using a single ISDN access link. Applications being currently explored show that significant benefits can be realized using incoming call management and LAN-based image server access by means of ISDN. It is envisioned that by the year 2001, a common API will facilitate multimedia applications on multivendor platforms architected within the OSI framework. These platforms will support interconnections of public and private ISDNs and bridging to BISDNs     

You'll never talk alone [cellular radio networks]

The transition in the 1990s from analogue to digital technology in cellular radio networks marked a turning point in mobile telephony, heralding a future of ubiquitous, low-cost, personal communications. The global growth of personal communications has continued at an incredibly rapid pace. The commercial pressures in today's markets are such that the development of future standards and systems are driven not simply by technology, but by commercial opportunity and market pull. In this respect personal communications has gone up a gear since the development of the early GSM digital cellular standards. The opportunities to apply PCS technology into adjacent markets and applications are huge, with new products and services limited only by imagination     

Mobility using IEEE 802.21 in a heterogeneous IEEE 802.16/802.11-based, IMT-advanced (4g) network

Industry is defining a new generation of mobile wireless technologies, called in cellular terminology "fourth generation" or "4G." This article shows that a system combining extensions of two radio access technologies, IEEE 802.11 and IEEE 802.16, meets the ITU-R's "IMT-Advanced" or 4G requirements. The extensions are 802.16 m (100 Mb/s, 250 km/h) and 802.11VHT (1 Gb/s, low velocity). The focus of this article is to show how IEEE 802.21 (the emerging IEEE standard for media-independent handover services) supports ";seamless"; mobility between these two radio access technologies. This mobility integrates the two radio access technologies into one system. We conclude that an 802.11VHT + 802.16 m + 802.21 system is likely to be proposed to the ITU-R for IMT- Advanced 4G.     

Telephones in the sky

Following the rapid growth of land mobile radio, increased attention is being paid to the provision of aircraft passenger telephony services. The author reviews the requirements for such services and outlines how modern technology will shortly enable worldwide aeronautical telephony to become a reality-via satellite. The current service provisions and enabling technologies are described and an insight provided into an experimental airline passenger telephone system     

Trends in digital communication by wire

The introduction and rapid spread of pulse code modulation must be examined in the framework of overall capability, efficiency, and cost. However, apart from television, the only field of application that can presently justify a massive investment in a new transmission system is telephony, because it represents the dominant communication market. Other communication services?such as data and facsimile transmission, telegraphy, broadcast sound, and closed-circuit television?must in general travel as passengers on telephony networks if they are to be economically feasible. Hence, the most fruitful approach to an examination of digital communication is to begin by studying PCM telephony and then to see how other services can be coordinated with it.     

A direct signaling system for flexible access and deployment oftelecommunication services

In current telecommunication networks, the value-added services available to subscribers are almost exclusively offered by the carrier providing local telephony service. This results from two factors: the processing for services is either based in or triggered from software executing on the access telecommunication switches, and signaling links over which subscribers request services are terminated on these access switches. In a direct signaling system, the signaling link from a subscriber does not necessarily terminate on an access switch, but rather on an intelligent server we call a call server. The call server invokes service functions and coordinates their interactions. This direct signaling system may be overlaid on various types of access networks, including circuit-switched telephony, packet telephony, wireless local loop, cable, etc. We discuss the design, implementation, and performance of a direct signaling system, including procedures that provide varying amounts of integration with current telecommunication systems. Our results show that a direct signaling system using a low-delay direct signaling link and in which no changes are made to existing telecommunicatione switches incurs approximately 100 ms higher postdial delay than a standard ISDN system. A direct signaling system in which telecommunication switches provide open interfaces and coordinate processing with direct signaling servers incurs approximately 100 ms lower postdial delay than a standard ISDN system     

Spectrum inventory issues - [spectrum policy and regulatory issues]

In the United States the issue of a "spectrum inventory" is of much current interest. A bill has been introduced in Congress, S. 649 (http://frwebgate.access.gpo.gov/cgi-bin/getdoc.cgi?dbname=111_cong_bills&docid=f:s649is.txt.pdf), that would require both U.S. spectrum regulators, the National Telecommunications and Information Administration/NTIA (for federal government spectrum users) and Federal Communications Commission/FCC (for all other spectrum users) to complete a "radio spectrum inventory" within 180 days. In this column we discuss the potential implications of such legislation. It is hoped that such a spectrum inventory will ultimately result in new spectrum for commercial applications. Since spectrum is a key raw ingredient for wireless systems, this could spur both technical innovation and the provision of new wireless services to the public. Finally, since telecommunications systems are themselves a key economic infrastructure, new and improved services could make existing users more efficient and competitive, as well as open the door to the creation of new businesses that use wireless services to offer nontelecommunications products and services (as Amazon uses the Internet).     

OpenRAN: a new architecture for mobile wireless Internet radioaccess networks

Cellular telephony networks depend on an extensive wired network to provide access to the radio link. The wired network, called a radio access network, provides such functions as power control and, in CDMA networks, combination of soft handoff legs (also known as macrodiversity resolution) that require coordination between multiple radio base stations and multiple mobile terminals. Existing RAN architectures for cellular systems are based on a centralized radio network controller connected by point-to-point links with the radio base transceiver stations. The existing architecture is subject to a single point of failure if the RNC fails, and is difficult to expand because adding an RNC is expensive. Also, although a network operator may have multiple radio link protocols available, most RAN architectures treat each protocol separately and require a separate RAN control protocol for each. We describe a new architecture, the OpenRAN architecture, based on a distributed processing model with a routed IP network as the underlying transport fabric. OpenRAN was developed by the Mobile Wireless Internet Forum IP in the RAN working group. The OpenRAN architecture applies principles to the radio access network that have been successful in reducing cost and increasing reliability in data communications networks. The result is an architecture that can serve as the basis for an integrated next-generation cellular radio access network