The IETF Internet telephony architecture and protocols

Internet telephony was first used as a simple way to provide point-to-point voice transport between two IP hosts. However, the growing interest in providing integrated voice, data, and video services has caused its scope to be extended. Internet telephony now encompasses a range of services, including not only traditional conferencing, call control, multimedia, and mobility services, but also new ones that integrate Web, e-mail, presence, and instant messaging applications with telephony. Internet telephony and traditional circuit-switched telephony will coexist for quite some time, requiring interworking between the two. In this article we present a suite of protocols, developed in the IETF, which provide a partial solution to this complex problem     

QoS issues in the converged 3G wireless and wired networks

The Internet evolution delineated through the last years has urged the wireless network community to support the deployment of IP multimedia services with guaranteed quality of service (QoS) in 3G wireless networks. This article copes with the interoperability between 3G wireless networks and wired next-generation IP networks, for the provision of services with an a priori known quality level over both environments. More specifically, the UMTS architecture as well as a prototypical implementation of the next-generation Internet based on DiffServ are considered. The article focuses on the mapping among the traffic classes of the two networks at the point where the networks converge, and discusses the requirements and possible solutions for their proper interworking at the signaling and user levels. Simulations prove that proper mapping among the traffic classes of each world is necessary in order to achieve the desired end-to-end traffic characteristics.     

Voice and Internet Multimedia in UMTS Networks

Voice telephony is the predominant service on today's cellular mobile networks, in terms of number of customers, revenues and network usage. However, it is difficult to predict how long this will be the case given the rising demand for new Internet multimedia services. It is therefore essential that 3rd generation (3G) mobile networks support a voice telephony service, but also that these networks are also capable of providing Internet multimedia services using the same technology.This paper provides an overview of how voice telephony is provided in the initial phase of the universal mobile telecommunications system (UMTS). It then describes how this is expected to evolve in later phases — so that voice telephony becomes one of a large number of multimedia services provided from a common Internet protocol-based mobile network.     

An IPv4-IPv6 translation mechanism for SIP overlay network in UMTS all-IP environment

Both IPv6 and session initiation protocol (SIP) are default protocols for Universal Mobile Telecommunications System (UMTS) all-Internet protocol (IP) network. In the existing mobile telecommunications environments, an IPv6-based UMTS all-IP network needs to interwork with other Internet protocol version 4 (IPv4)-based SIP networks. Therefore, mobile SIP applications are typically offered through an overlay structure over the IPv4-Internet protocol version 6 (IPv6) interworking environments. Based on 3GPP 23.228, we propose an IPv4-IPv6 translation mechanism (i.e., SIPv6 translator) that integrates different IP infrastructures (i.e., IPv4 and IPv6) to provide an overlay network for transparent SIP application deployment. In this paper, we present the architecture and the call flows of the SIPv6 translator. An analytic model is proposed to investigate the fault tolerance issue of our approach. Our study provides guidelines to select appropriate number of processors for fault tolerance.     

Toward an all-IP-based UMTS system architecture

Looking into the future, two main drivers for the mobile telecommunications market can be identified: third-generation mobile systems (e.g., UMTS) and the Internet (e.g., the introduction of IP technologies like voice/multimedia over IP in mobile networks). UMTS is seen as the enabler of wireless multimedia applications and portability of a personalized service set across network/terminal boundaries, as defined within the virtual home environment (VHE) system concept. In light of these evolutions, this article investigates the impact of the evolution toward an all-IP UMTS network architecture on the UMTS service architecture, which is based on the VHE concept. The article discusses two possible scenarios for supporting VoIP services in the UMTS service architecture and analyzes their applicability in an all-IP-based UMTS network. The first is based on the traditional centralized IN service architecture. The second proposes a new decentralized architecture based on direct control of VoIP call control equipment by open service architecture interfaces     

Supplementary services in the H.323 IP telephony network

Traditionally, different networks were developed to handle voice, data, and video. The circuit-switched telephone network carried voice and the packet network carried data. Due to different deployment of these networks, different services were developed, such as voice mail in the telephone network and electronic mail on the Internet. With the revolution of multimedia in the computer industry, voice, video, and data are now being carried on both networks. Supplementary services, such as transfer and forwarding (which were originally developed for private telephone networks and later migrated to public telephone networks) are now being developed for packet networks. The standards for packet networks are being defined in the H.323-based series of ITU-T recommendations. This article provides the H.323 architecture for supplementary services, the differences in deployment of these services between the circuit-switched and packet-switched networks, and interworking of these services across hybrid networks     

Transport of mobile application part signaling over Internetprotocol

The convergence of voice, data, and video networks is creating a new environment for telecommunications. In response to the changes, telecommunications equipment manufacturers and service providers are competing fiercely to bring an optimum solution to customers. The evolution of GSM to GPRS and to UMTS is a cellular wireless industry endeavour to meet this demand. This evolution will see the core wireless network infrastructure change from circuit-switched to packet-switched where voice and data are transported using IP as the common protocol. However, this poses a number of challenges, one of which is how to run the key mobile application part signaling protocols over IP. MAP defines the application protocols between switches and databases (e.g., MSC, VLR, SGSN, HLR) for supporting mobility management, security management, radio resource management, and mobile equipment management. UMTS supports both circuit-switched and packet-switched services     

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.     

Architecture for interworking data over PCS

This article presents a new approach for wireless service providers to offer data services by taking advantage of the existing infrastructure for voice services and interworking with existing wireline-based data services. The article presents a framework: for interworking between any wireless radio system and any data application on the wireline network. The interworking is provided by a common interworking function that uses a generic interworking control protocol (ICP). Any radio system capable of using the ISDN-based C-interface and implementing ICP can take advantage of the proposed approach. ICP is a generic protocol and can be implemented using different networks. The article considers an ISDN network for the lower-layer transport of ICP. Though the article focuses on interworking with the PSTN and the Internet, the architecture also allows access to other data networks     

Wireless Internet access for mobile subscribers based on theGPRS/UMTS network

With the advent of IP technologies and the tremendous growth in data traffic, the wireless industry is evolving its core networks toward IP technology. Enabling wireless Internet access is one of the upcoming challenges for mobile radio network operators. The General Packet Radio Service is the packet-switched extension of GSM and was developed to facilitate access to IP-based services better than existing circuit-switched services provided by GSM. We illustrate how a visited mobile subscriber on a GPRS/UMTS network can access his/her home network via the gateway GPRS support node (GGSN). We also propose some implementation ideas on wireless Internet access for a remote mobile subscriber based on a GPRS/UMTS network     

VoIP mobility in IP/cellular network internetworking

This article explores VoIP mobility in the context of IP and cellular networks interworking. ITU-T Rec. H.323 gateways provide the interconnection between IP networks and switched circuit networks. They allow a call originating from an SCN phone to be transmitted over an IP network to an H.323 terminal, or bridged to another SCN phone. While H.323 provides interoperability with other SCN terminals, the major efforts have been focused on IP/wired SCN (PSTN, ISDN, etc.) interworking. In this article we discuss the challenges associated with the interworking between IP networks and cellular networks through H.323 gateways, and propose an innovative approach using the existing call transfer supplementary service to provide VoIP mobility in the H.323 IP telephony networks. The proposed approach uses existing components in the H.323 standard, thereby allowing VoIP mobility service in hybrid IP/cellular networks to be a value-added feature in the existing H.323-compliant Internet telephony systems     

VoIP信令协议SIP和H.323的比较

VoIP系统凭借其低廉的话费和较好的语音质量,已经成为重要的电信业务,并有取代传统长途业务的趋势.许多组织研究并制定了IP网络上呼叫的协议标准,但有两种IP电话信令和控制标准最具有影响力.一种是ITU推荐的H.323协议,另一种是IETF的SIP.这两种协议代表了解决同一问题的两种不同的方法:H.323是信令基于ISDN Q.931和早期推荐的H系列协议的传统的电路交换的方法,而SIP是一种支持基于HTTP的IP网络的超轻量协议标准.本文,我们主要针对SIP和H.323的体系结构,可靠性,复杂性,可扩展性,可伸缩性以及支持业务类型方面进行比较.     

Handoff with DSP Support: Enabling Seamless Voice Communications across Heterogeneous Telephony Systems on Dual-Mode Mobile Devices

In this paper we investigate the problem of voice communications across heterogeneous telephony systems on dual-mode (WiFi and GSM) mobile devices. Since GSM is a circuit-switched telephony system, existing solutions that are based on packet-switched network protocols cannot be used. We show in this paper that an enabling technology for seamless voice communications across circuit-switched and packet-switched telephony systems is the support of digital signal processing (DSP) techniques during handoffs. To substantiate our argument, we start with a framework based on the Session Initiation Protocol (SIP) for vertical handoffs on dual-mode mobile devices. We then identify the key obstacle in achieving seamless handoffs across circuit-switched and packet-switched systems, and explain why DSP support is necessary in this context. We propose a solution that incorporates time alignment and time scaling algorithms during handoffs for supporting seamless voice communications across heterogeneous telephony systems. We conduct testbed experiments using a GSM-WiFi dual-mode notebook and evaluate the quality of speech when the call is migrated from WiFi to GSM networks. Evaluation results show that such a cross-disciplinary solution involving signal processing and networking can effectively support seamless voice communications across heterogeneous telephony systems.     

QoS enabled voice support in the next generation Internet: issues,existing approaches and challenges

The Internet is under rapid growth and continuous evolution in order to accommodate an increasingly large number of applications with diverse service requirements. In particular, Internet telephony, or voice over IP is one of the most promising services currently being deployed. Besides the potentially significant cost reduction, Internet telephony can offer many new features and easier integration with widely adopted Web-based services. Despite these advantages, there still exist a number of barriers to the widespread deployment of Internet telephony. The most prominent one, however, is how to ensure the QoS needed for voice conversation. The purpose of this article is to survey the state-of-the-art technologies in enabling the QoS support for voice communications in the next-generation Internet. In this article, we first review the existing technologies in supporting voice over IP networks, including the basic mechanisms in the IETF Internet telephony architecture and ITU-T H.323-related Recommendations. We then discuss the IETF QoS framework, specifically the Intserv and Diffserv framework. Finally, we present two leading companies' (Cisco and Lucent) solutions to offering IP telephony services as examples to illustrate how real systems are implemented     

Mobile IP-DECT internetworking architecture supporting IMT-2000applications

Realistic realization and mass acceptance of mobile data services require networking architectures offering acceptable quality of service and attractive tariffs. A novel strategy for this goal is maximum integration of popular data networking standards and their infrastructure into wireless networks. This article discusses a Mobile IP-based network architecture to provide IP services in DECT to support IMT-2000 applications. DECT offers micromobility within multicell subnets, while Mobile IP supports macromobility between multicell subnets. Incorporating Mobile IP into the DECT handoff mechanism in this way extends DECT micromobility with IP macromobility. Also, utilizing fast, seamless DECT handoff management reduces Mobile IP handoff delay to circumvent TCP throughput degradation during handoff and reduce frequency of Mobile IP signaling over the ether to conserve spectral efficiency. This feature seamlessly unifies DECT with the global Internet. Seamless integration of DECT with the Internet is crucial due to the continuing phenomenal popularity of the Internet and wireless communications, and ubiquity of DECT systems. To achieve the above DECT/IP interworking efficiently, the architecture introduces a network entity called a DECT service switching point, which is an extended DECT central control fixed part. DECT network-level services are mapped onto those of the IETF integrated services architecture to maintain QoS provided by DECT in the backbone Internet. Mobile Resource Reservation Protocol, an extended RSVP tailored to mobile networking, is adopted to provide the needed signaling in IntServ. The proposed architecture preserves traditional non-IP based services such as PSTN voice     

Mobile and Multicast IP Services in PACS: System Architecture, Prototype, and Performance

Traditionally, wireless cellular communication systems have been engineered for voice. With the explosive growth of Internet applications and users, there is an increasing demand on providing Internet services to mobile users based on the voice-oriented cellular networks. However, Internet services add a set of radically different requirements on to the cellular wireless networks, because the nature of communication is very different from voice. It is a challenge to develop an adequate network architecture and necessary systems components to meet those requirements.This paper describes our experience on developing Internet services, in particular, mobile and multicast IP services, in PACS (Personal Access Communication Systems). Our major contributions are five-fold: (i) PACS system architecture that provides wireless Internet and Intranet access by augmenting the voice network with IP routers and backbone links to connect to the Internet; (ii) simplified design of RPCU (Radio Port Controller Unit) for easy service maintenance and migration to future IP standards such as IPv6; (iii) native PACS multicast to efficiently support dynamic IP multicast and MBone connectivity; (iv) optimization and incorporation of Mobile IP into PACS handoff mechanism to efficiently support roaming within a PACS network as well as global mobility between PACS networks and the Internet; (v) successful prototype design of the new architecture and services verified by extensive performance measurements of IP applications. Our design experience and measurement results demonstrate that it is highly feasible to seamlessly integrate the PACS networks into the Internet with global IP mobility and IP multicast services.     

Real-time voice over packet-switched networks

We discuss the architecture and technical viability of transporting real-time voice over packet-switched networks such as the Internet. The value of integrating voice and data networks onto a common platform is well known. The telephony industry has proposed the ATM standard as a means of upgrading the Internet to provide both real-time and data services. In contrast, voice services may be added to traditional IP networks that were originally designed for data transmission alone. We consider the feasibility and expected quality of service of audio applications over IP networks such as the Internet. In particular, we examine possible architectures for voice over IP and discuss measured Internet delay and loss characteristics     

VTOA/VolP/N-ISDN Telephony Gateway

This article presents the architecture and implementation of a telephony gateway for interworking between N- ISDN, ATM and IP telephony. In this way, interworking is achieved both within private networks and with the PSTN, address translation being performed according to both the vtoa (atm interface) and H .323 (ip interface) specifications. The gateway implementation is based on a PC, presenting a cost- effective alternative to the equipment currently available on the market. Moreover, its highly modular software architecture allows new telephony interfaces to be easily added.     

Internet telephony: going like crazy

Interet telephony is possibly the fastest-growing part of communications today. This article discusses what exactly it is, who needs it, and how it works. Internet telephony, or voice over Internet protocol (VoIP), is the provision of phone service over the Internet. But in sharp contrast with conventional telephony, it carries voice traffic as data packets over a packet-switched data network instead of as a synchronous stream of binary data over a circuit-switched, time-division multiplexed (TDM) voice network. There are some substantial benefits (as well as some sticky problems) to the scheme, which is why companies and individuals are finding it increasingly attractive     

VoWLAN通信平台的设计与实现

VoWLAN也叫VoWiFi或者WiFi VoIP。它是基于无线网络技术和VoIP网络,是两者的有机结合。即是通过WLAN提供VoIP业务,使得终端用户通过WLAN拨打IP电话成为现实。本文提出了在基于Linux操作系统的SIP应用服务器及VoIP网关中,如何通过SIP信令和传统的PSTN数据通信线路与无线网络无缝连接方案,从而实现IP网络与传统电话间的实时语音通信、电话会议、语音信箱、视频通信、短消息、数据传输等业务。本设计已成功应用于某企业的实时语音通信平台,获得良好的效果。