Optimization of Mobile IPv6-Based Handovers to Support VoIP Services in Wireless Heterogeneous Networks

The support of voice over Internet Protocol (VoIP) services in next-generation wireless systems requires the coupling of mobility with quality of service. The mobile node can experience disruptions or even intermittent disconnections of an ongoing real-time session due to handovers. The duration of such interruptions is called disruption time or handover delay and can heavily affect user satisfaction. Therefore, this delay needs to be minimized to provide good-quality VoIP services. In this paper, the focus is on the network layer mobility, specifically on mobile Internet Protocols (MIPs), since they are natural candidates for providing mobility at layer 3. Using analytical models, the authors evaluate MIPv4, MIPv6, fast MIPv6 (FMIPv6), and hierarchical MIPv6 (HMIPv6) and compare their performances in terms of handover delay for VoIP services. To optimize the handover delay, the authors propose to use the adaptive retransmission timer described in this paper. The results obtained using the adaptive timer technique show that for a 3% frame error rate and a 128-kb/s channel, the handoff delay is about 0.075 s (predictive) and 0.051 s (reactive) for FMIPv6. It is around 0.047 s [intra-mobile anchor point (MAP)] and 1.47 s (inter-MAP) for HMIPv6, around 1 s for MIPv6, and 0.26 s for MIPv4     

Mobility management for VoIP service: Mobile IP vs. SIP

Wireless Internet access has gained significant attention as wireless/mobile communications and networking become widespread. The voice over IP service is likely to play a key role in the convergence of IP-based Internet and mobile cellular networks. We explore different mobility management schemes from the perspective of VoIP services, with a focus on Mobile IP and session initiation protocol. After illustrating the signaling message flows in these two protocols for diverse cases of mobility management, we propose a shadow registration concept to reduce the interdomain handoff (macro-mobility) delay in the VoIP service in mobile environments. We also analytically compute and compare the delay and disruption time for exchanging signaling messages associated with the Mobile IP and SIP-based solutions.     

RSVP Extensions for Real-Time Services in Hierarchical Mobile IPv6

The Mobile IPv6 (MIPv6) provides many great features, such as sufficient addressing space, mobility, and security; MIPv6 is one of the most important protocols for next generation mobile Internet. Simultaneously, with the rapid improvement of wireless technologies, the real-time multi-media IP services such as video on demand, videoconference, interactive games, IP telephony and video IP phone will be delivered in the near future. Thus, to furnish accurate QoS for real-time services is one of the most important thing in the next generation mobile Internet. Although RSVP, which is a resource reservation protocol, processes signaling messages to establish QoS paths between senders and receivers, RSVP was originally designed for stationary networks and not aware of the mobility of MNs. Therefore, this paper proposes a novel RSVP extension to support real-time services in Hierarchical Mobile IPv6 (HMIPv6) environments. For intra-site mobility, the concept of QoS Agent (QA) is proposed to handle the RSVP QoS update messages and provide the advanced reservation models for real-time services. For inter-site mobility, IP multicast can help to invite inter-site QAs to make pre-reservation and minimize the service disruption caused by re-routing the data path during handover. Simulation results show that the proposed scheme over HMIPv6 is more suitable for real-time services than the famous RSVP tunnel-based solution.     

Mobile Internet telephony: mobility extension to H.323

This paper studies mobility extensions to ITU-T Rec. H.323 for the support of mobile Internet telephony. Internet telephony, also known as voice-over Internet protocol (IP) (VoIP), requires the transmission of two-way and real-time traffic over IP-based networks. The current version of H.323 allows IP telephony and the interoperability of the Internet with switched circuit networks (SCN). However, VoIP mobility has not been previously widely considered, where VoIP mobility refers to the mobility within the scope of IP telephony. We focus on terminal mobility for VoIP. We investigate the influence of mobility on the H.323 layer and propose an H.323 mobility solution to be implemented over the IP layer. Two approaches to mobility extensions to H.323 are described: using ad hoc multipoint conference expansion and using IP multicasting to emulate mobility. Besides, we have also shown that the proposed ad hoc expansion approach shares many properties with the alternative of using IP multicasting for mobility. Hence, the call signaling procedure for the ad hoc expansion approach is also applicable to the multicasting approach. Since ad hoc multipoint expansion has been defined in H.323, our solution introduces no additional entities to H.323 and requires minimal modifications to the existing H.323 protocol. Such mobility extensions can serve as a value-added feature for the Internet telephony systems compliant to the H.323 standard     

Performance evaluation of packet aggregation scheme for VoIP service in wireless multi-hop network

In a wireless multi-hop network environment, energy consumption of mobile nodes is an important factor for the performance evaluation of network life-time. In Voice over IP (VoIP) service, the redundant data size of a VoIP packet such as TCP/IP headers is much larger than the voice data size of a VoIP packet. Such an inefficient structure of VoIP packet causes heavy energy waste in mobile nodes. In order to alleviate the effect of VoIP packet transmission on energy consumption, a packet aggregation algorithm that transmits one large VoIP packet by combining multiple small VoIP packets has been studied. However, when excessively many VoIP packets are combined, it may cause deterioration of the QoS of VoIP service, especially for end-to-end delay. In this paper, we analyze the effect of the packet aggregation algorithm on both VoIP service quality and the energy consumption of mobile nodes in a wireless multi-hop environment. We build the cost function that describes the degree of trade-off between the QoS of VoIP services and the energy consumption of a mobile node. By using this cost function, we get the optimum number of VoIP packets to be combined in the packet aggregation scheme under various wireless channel conditions. We expect this study to contribute to providing guidance on balancing the QoS of VoIP service and energy consumption of a mobile node when the packet aggregation algorithm is applied to VoIP service in a wireless multi-hop networks.     

WLAN中基于MIPv6的移动VoIP切换性能仿真分析

移动IP是一个在Internet上基于网络层提供移动性支持功能的要求较高的VoIP业务,切换延迟将直接影响到话音质量,严重时甚至会中断正在进行的会话.文章借助ns2网络模拟器仿真分析了WLAN中基于MIPv6的移动VoIP切换性能.结果表明,MIPv6及其扩展协议的切换性能优劣顺序依次为:F-HMIPv6、FMIPv6、HMIPv6、MIPv6.尤其是F-HMIPv6协议,无论端到端延迟还是切换延迟,都得到了最大的改善.所得结论能为网络切换性能的进一步优化提供重要依据.     

Hybrid mobility management schemes integrating mobile IP and SIP for seamless invocation of services in All-IP network

The growing demand for seamless invocation of different multimedia services from handheld devices anytime anywhere is the main driving force for drawing attention in the area of mobility management. Although Session Initiation Protocol (SIP) based mobility solution is very efficient for real-time services, Mobile IP is required to handle mobility of the mobile node (MN) at the network layer. We have extensively studied and explored some existing mobility management methods integrating the functionalities of Mobile IP and SIP in this paper. These schemes require support of IP encapsulation at the protocol stack of correspondent node (CN). To address the above problem, we have proposed in this paper, certain modification at the IP layer of Base Station (BS) that also reduces the bandwidth consumption. Moreover, service provisioning in a continuous way in public places like airport, university campus etc., requires to integrate some micro-mobility protocol with the existing mobility management methods to reduce the handoff disruption time in case of intradomain handoff. Thus, in this paper, we have proposed two new hybrid mobility management schemes that integrate two existing Mobile IP and SIP-based schemes where the proposed modification in the IP layer of BS is incorporated separately with the micro-mobility protocol Hierarchical Mobile IP (HMIP). The numerical results show that the integration of HMIP into the existing methods reduces both the signaling cost and the delay, mainly the active handoff disruption time. Simulation results on NS-2 demonstrate the performance improvement of the proposed mobility management schemes over the existing methods in terms of handoff delay.     

Mobility Management for VoIP on Heterogeneous Networks: Evaluation of Adaptive Schemes

The introduction of the IP multimedia subsystem on 3G cellular networks and the integration with other widely deployed wireless networks based on the IEEE 802.11 protocol family require support for both mobility and quality of service. When mobile systems move across heterogeneous networks, ongoing real-time sessions are affected not only by handoff delay but also by different packet delay and bit rate. In this paper, we propose a cross-layer mechanism that takes into account mobility at different layers of the network stack in order to yield better quality for VoIP, videoconferencing, and other real-time applications. We describe our cross-layer architecture, adaptation techniques, a prototype implementation, and experimental results.     

VoIP over DVB-RCS with QoS and bandwidth on demand

Motivated by the need for compliance/interoperability above the satellite-specific layers, this article proposes a consolidated approach for voice over IP over satellite networks based on the ETSI DVB-RCS standard. Voice communication is a real-time service that needs priority over other services in IP environments with limited bandwidth, such as IP satellite networks. Bandwidth utilization in such networks needs to be optimized in order to reduce service costs, and this requires the use of dynamic bandwidth allocation schemes. This article therefore addresses the role of bandwidth on demand in the optimization of bandwidth allocation for VoIP and assesses the impact of BoD mechanisms on voice quality. The trade-off between voice quality and bandwidth efficiency is investigated under different DVB-RCS-specific capacity request/allocation strategies, and it is demonstrated that DVB-RCS provides an efficient platform for integrated support for a variety of VoIP applications over satellite. The main contribution of this article consists of the identification of the mechanisms capable of responding to the key challenges raised by the VoIP application in the satellite environment.     

Integrated services packet networks with mobile hosts: Architecture and performance

This paper considers the support of real-time services to mobile users in an Integrated Services Packet Network. In the currently existing architectures, the service guarantees provided to the mobile hosts are mobility dependent, i.e., mobile hosts experience wide variation in the quality of service and often service disruption when hosts move from one location to another. The network performance degrades significantly when mobile hosts are provided with mobility independent service guarantees. In this paper we have proposed a service model for mobile hosts that can support adaptive applications which can withstand service degradation and disruption, as well as applications which require mobility independent service guarantees. We describe an admission control scheme for implementing this service model and evaluate its performance by simulation experiments. Simulation results show that, if sufficient degree of multiplexing of the mobility dependent and independent services are allowed, the network does not suffer any significant performance degradation and in particular our admission control scheme achieves high utilization of network resources.     

面向固定移动融合环境的P2PSIP系统设计

全业务运营是电信市场继语音和宽带接入服务之后的下一个增长点,而基于IP的融合有线网络和无线网络的语音服务则是全业务的重点之一。本文通过分析现有VoIP网络存在的问题以及固定移动融合网络环境下VoIP的特点,提出一种新型双层重叠网架构的P2PSIP架构,并阐述了新型架构的优点及双层重叠网之间的通信机制。新型架构能有效提高系统的安全性、健壮性和用户节点资源利用效率,更好的满足固定移动融合网络环境下VoIP对带宽、网络质量和安全性的要求。     

IP multimedia services: analysis of mobile IP and SIP interactions in 3G networks

Third-generation cellular networks have been designed to provide a variety of IP data services. Both IPv4 and IPv6 are supported in order to provide future-proof solutions. Mobility is supported through both cellular-specific and IP mechanisms. Mobile IP is becoming a key technology for managing mobility wireless networks. At the same time, the session initiation protocol is the key to realizing and provisioning services in IP-based cellular networks. The need for mobility of future real-time service independent of terminal mobility requires SIP to seamlessly interwork with mobile IP operations. In this article, we investigate the issues related to interworking between SIP and mobile IP, with a focus on IPv6 and the applicability to 3G networks being standardized in 3GPP and 3GPP2.     

Integrated Mobility Management Methods for Mobile IP and SIP in IP based Wireless Data Networks

With developments in voice over IP (VoIP), IP-based wireless data networks and their application services have received increased attention. While multimedia applications of mobile nodes are served by Session Initiation Protocol (SIP) as a signaling protocol, the mobility of mobile nodes may be supported via Mobile IP protocol. For a mobile node that uses both Mobile IP and SIP, there is a severe redundant registration overhead because the mobile node has to make location registration separately to a home agent for Mobile IP and to a home registrar for SIP, respectively. Therefore, we propose two new schemes that integrate mobility management functionality in Mobile IP and SIP. We show performance comparisons among the previous method, which makes separate registration for Mobile IP and SIP without integration, and our two integrated methods. Numerical results show that the proposed methods efficiently reduce the amount of signaling messages and delay time related to the idle handoff and the active handoff.     

Topology-aided cross-layer fast handoff designs for IEEE 802.11/mobile IP environments

This study first reviews state-of-the-art fast handoff techniques for IEEE 802.11 or Mobile IP networks. Based on that review, topology-aided cross-layer fast handoff designs are proposed for Mobile IP over IEEE 802.1.1 networks. Time-sensitive applications, such as voice over IP (VoIP), cannot tolerate the long layer-2 plus layer-3 handoff delays that arise in IEEE 802.11/Mobile IP environments. Cross-layer designs are increasingly adopted to shorten the handoff latency time. Handoff-related layer-2 triggers may reduce the delay between layer-2 handoff completion and the associated layer-3 handoff activation. Cross-layer topology information, such as the association between 802.11 access points and Mobile IP mobility agents, together with layer-2 triggers, can be utilized by a mobile node to start layer-3 handoff-related activities, such as agent discovery, address configuration, and registration, in parallel with or prior to those of layer-2 handoff. Experimental results indicate that the whole handoff. delay can meet the delay requirement of VoIP applications when layer-3 handoff activities occur prior to layer-2 handoffs.     

Multicast Broadcast Services Support in OFDMA-Based WiMAX Systems [Advances in Mobile Multimedia]

Multimedia stream service provided by broadband wireless networks has emerged as an important technology and has attracted much attention. An all-IP network architecture with reliable high-throughput air interface makes orthogonal frequency division multiplexing access (OFDMA)-based mobile worldwide interoperability for microwave access (mobile WiMAX) a viable technology for wireless multimedia services, such as voice over IP (VoIP), mobile TV, and so on. One of the main features in a WiMAX MAC layer is that it can provide'differentiated services among different traffic categories with individual QoS requirements. In this article, we first give an overview of the key aspects of WiMAX and describe multimedia broadcast multicast service (MBMS) architecture of the 3GPP. Then, we propose a multicast and broadcast service (MBS) architecture for WiMAX that is based on MBMS. Moreover, we enhance the MBS architecture for mobile WiMAX to overcome the shortcoming of limited video broadcast performance over the baseline MBS model. We also give examples to demonstrate that the proposed architecture can support better mobility and offer higher power efficiency.     

MRSVP: A Resource Reservation Protocol for an Integrated Services Network with Mobile Hosts

This paper describes a reservation protocol to provide real-time services to mobile users in an Integrated Services Packet Network. Mobility of hosts has significant impact on the quality of service provided to a real-time application. The currently proposed network system architecture and mechanisms to provide real-time services to fixed hosts are inadequate to accommodate the mobile hosts which can frequently change their point of attachments to the fixed network. Mobile hosts may experience wide variations of quality of service due to mobility. To reduce the impacts of mobility on QoS guarantees, a mobile host needs to make advance resource reservations at multiple locations it may possibly visit during the lifetime of the connection. The currently proposed reservation protocol in the Internet, RSVP, is not adequate to make such reservations for mobile hosts. In this paper, we describe a new reservation protocol, MRSVP, for supporting integrated services in a network with mobile hosts.     

Performance Optimizations for Deploying VoIP Services in Mesh Networks

In the recent past, there has been a tremendous increase in the popularity of VoIP services as a result of huge growth in broadband access. The same voice-over-Internet protocol (VoIP) service poses new challenges when deployed over a wireless mesh network, while enabling users to make voice calls using WiFi phones. Packet losses and delay due to interference in a multiple-hop mesh network with limited capacity can significantly degrade the end-to-end VoIP call quality. In this work, we discuss the basic requirements for efficient deployment of VoIP services over a mesh network. We present and evaluate practical optimizing techniques that can enhance the network capacity, maintain the VoIP quality and handle user mobility efficiently. Extensive experiments conducted on a real testbed and ns-2 provide insights into the performance issues and demonstrate the level of improvement that can be obtained by the proposed techniques. Specifically, we find that packet aggregation along with header compression can increase the number of supported VoIP calls in a multihop network by 2-3 times. The proposed fast path switching is highly effective in maintaining the VoIP quality. Our fast handoff scheme achieves almost negligible disruption during calls to roaming clients     

Deployment of IP multimedia streaming services in third-generation mobile networks

In this article, an end-to-end quality of service framework for streaming services in 3G mobile networks is considered. Under this scenario, the interaction between UMTS and IETF's protocols and mechanisms for a streaming session is analyzed. By signaling flowcharts, it is shown that both groups of protocols and mechanisms can co-operate to provide seamless end-to-end real-time services. Specifically, the article proposes to make the IP multimedia subsystem aware of the real time streaming protocol, in order to extend its control from SIP to RTSP-based services, such as multimedia streaming services. Supported by this proposed framework, provisioning of audio streaming services over 3G mobile networks is also outlined.     

基于RTP协议的IP电话QoS监测及提高策略

介绍了多媒体通信的发展趋势和当前存在的问题,对基于RTP协议的网络电话中音频数据传输技术进行了研究,对影响实时传输质量QoS的典型因素进行了分析。在局域网的环境下进行了语音包分析实验,探讨了基于RTP协议的QoS动态监测方法,并提出可行的改进方案。     

Network mobility protocol for vehicular ad hoc networks

The goal of the network mobility management is to effectively reduce the complexity of handoff procedure and keep mobile devices connecting to the Internet. When users are going to leave an old subnet and enter a new subnet, the handoff procedure is executed on the mobile device, and it may break off the real‐time service, such as VoIP or mobile TV, because of the mobility of mobile devices. Because a vehicle is moving so fast, it may cause the handoff and packet loss problems. Both of the problems will lower down the throughput of the network. To overcome these problems, we propose a novel network mobility protocol for vehicular ad hoc networks. In a highway, because every car is moving in a fixed direction at a high speed, a car adopting our protocol can acquire an IP address from the vehicular ad hoc network through the vehicle‐to‐vehicle communications. The vehicle can rely on the assistance of a front vehicle to execute the prehandoff procedure, or it may acquire a new IP address through multihop relays from the car on the lanes of the same or opposite direction and thus may reduce the handoff delay and maintain the connectivity to the Internet. Simulation results have shown that the proposed scheme is able to reduce both the handoff delay and packet loss rate. Copyright © 2013 John Wiley & Sons, Ltd.