An effective IPv4–IPv6 translation mechanism for SIP applications in next generation networks

In a next generation network, the IPv6‐enabled IP multimedia subsystem (IMS) network may connect to an IPv4 network. When an IPv4/IPv6 dual‐stack user equipment (UE) initiates a call by sending an IPv6 SIP INVITE message to an IPv4‐only user agent (UA), the call cannot be established correctly. To resolve this problem, the IMS‐application layer gateway solution, the redirect solution, and the interactive connectivity establishment solution have been proposed. In this paper, we propose an effective solution where only the IPv6 INVITE message is translated into an IPv4 INVITE message. Upon receipt of the IPv4 200 OK message replied from the IPv4‐only UA, the dual‐stack UE learns that the correspondent UA supports IPv4‐only and utilizes IPv4 instead of IPv6 to send the subsequent SIP messages and real‐time transport protocol (RTP) packets. The proposed solution is compared with the existing solutions in terms of network node modification, call setup complexity, and RTP transmission latency. Our study indicates that the proposed solution outperforms the other three solutions in the call setup and the RTP transmission. Copyright © 2009 John Wiley & Sons, Ltd.     

基于Davinci平台安全VoIP系统的设计

文中在TI Davinci平台的基础上研究了安全VoIP视频电话终端的设计与实现,通过研究SIP通信中存在的注册劫持、服务器伪装、消息篡改等一系列安全威胁,针对SIP的安全问题改进了INVITE消息格式,采用优化算法的DSP加密子系统,并结合PKI、USBkey等技术提出了一套针对VoIP系统安全问题的完整解决方案,该方案包括基于PKI数字证书、扩展SIP协议格式的实体认证机制和基于PKI的SIP消息体加密机制,并使用了USBkey智能密码钥匙以保证PKI数字证书存储的安全性.     

PCS mobility management using the reverse virtual call setupalgorithm

The increased demand for wireless mobile communications coupled with the finite available spectrum has motivated investigation into alternative methods of tracking users and delivering calls. We present a new scheme for delivering mobile terminated calls named reverse virtual call setup (RVC). Only a few new fixed network SS7 signaling messages are needed to implement this protocol; we specify them here. RVC can function within the existing cellular paging network or with an integrated overlaid paging network. The relative performance of RVC for both scenarios is investigated     

Selective parsing to enhance SIP performance

The session initiation protocol (SIP) is used as the signaling protocol in the IP multimedia subsystem (IMS) and the signaling is becoming computing intensive comparing to the current telecommunication network. The SIP is a text-based protocol with characteristics of unordered and verbose headers, variable-size message, and case-insensitive keyword. It imposes challenges for an efficient message processing. The property of SIP elements being able to process SIP messages quickly is critical for the performance of IMS networks. This article investigates the performance of SIP message processed in SIP servers, mainly focusing on improving message parsing by introducing a method named selective parsing for SIP message (SP4SIP). By modeling and analyzing a SIP server with a tandem Jackson network, it is concluded that parsing messages is the bottleneck of a SIP server performance, i.e., it is the most processing intensive activity in the system. To validate the approach, it has been implemented in a high-performance SIP server in the authors' lab. The results show that selective parsing for SIP message can indeed reduce processing time.     

无线网络中SIP信令组合压缩方案研究

IMS（IP多媒体子系统）采用SIP协议建立和维护多媒体会话,但SIP是基于文本的协议,消息比较大,当应用于带宽小的无线网络时,会增加会话建立的时延。为缩短会话建立时间,有必要对SIP消息进行压缩。针对单一的使用压缩算法在SIP信令压缩性能方面的不足,本文在Deflate压缩算法的基础上,采用不同的压缩策略,对SIP消息实现了压缩。仿真结果表明,静态字典、用户自定义字典和共享压缩的组合方案得到了最好的压缩效果,压缩后的消息平均大小仅为原来消息大小的14％左右。     

On fluid‐flow modeling of priority based request scheduling for finite buffer SIP server

With the widespread deployments of voice‐over‐internet protocol services, the existing session initiation protocol (SIP) design cannot scale up for large network sizes. Events triggering a demand burst or a server slowdown can cause SIP server overload, overload propagation, and crash, thus bringing down the whole SIP network. Since the SIP retransmission mechanism exacerbates the overload condition, existing models created for a stable SIP system cannot be effectively used to analyze an overloaded server. In this paper, we propose a fluid‐flow model to characterize the behavior of the finite buffer SIP server equipped with priority‐based request scheduling mechanism (PRSM). The model for the PRSM uses primary and secondary queues for the original request messages and the retransmitted requests, respectively. The performance metrics, namely, the failed call attempts and the response delay from sending INVITE request until receiving a 100‐Trying response, are derived using the arrival time‐slot tracking and the removal processes of the proposed fluid‐flow model. We conducted test cases under the heavy traffic conditions, where the overload is caused by bulk and bursty arrivals or server slowdown. The numerical results closely match with the simulation results for all experiments, indicating that the proposed model can accurately capture the dynamic behavior of an SIP server with the PRSM. The experiments demonstrate that the number of failed call attempts is close to 0 and the mean response delay is kept constant around 175 ms for the PRSM when the buffer size is higher than 1K while both metrics are significantly higher for the conventional SIP.     

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.     

A Novel SIP-Based Route Optimization for Network Mobility

With the provision of various wireless services, e.g., third-generation (3G) and wireless local area network (WLAN), more and more people request to access the Internet anywhere at anytime. For example, people want to check their e-mails on the bus or watch online news while traveling in the train. For this purpose, the Internet Engineering Task Force (IETF) proposed the concept of network mobility, i.e., a set of users move as a unit. We motivate the network mobility problem by considering the state-of-the-art scenario of the network mobility (NEMO) basic support protocol that is extended from the Mobile IPv6 (MIPv6). In order to avoid the same problems suffered by MIPv6, a new session initiation protocol (SIP)-based network mobility management scheme called SIP-NEMO is designed and proposed in this paper. The proposed SIP-NEMO not only copes with the movement of a mobile network but also achieves the route optimization between two SIP clients without too many signaling messages over wireless links, even if the mobile network is nested. In this paper, we also analytically compute and simulate the performance of SIP-NEMO with the NEMO basic support protocol proposed by the IETF.     

基于SIP的手机视频监控系统设计

介绍了一种基于SIP协议的手机视频监控系统的设计方法,并主要讨论了如何利用SIP扩展方法的消息体来定义控制指令和告警短信。该系统具有极大的灵活性、可扩展性、安全性和跨平台性,可以应用于互联网和3G移动网,并可平滑过渡到下一代核心网IMS。     

Session integration service over multiple devices

One of the most important research subjects in session initiation protocol (SIP) is mobility management. Session mobility is a key issue in SIP mobility management. Session mobility is an advanced feature that maintains an ongoing media session from one device to another. In this paper, we consider the issue of session mobility and propose a complete integrated solution, referred to as ‘session integration service’, to transfer an ongoing media session over multiple devices. To provide flexibility in session mobility, three new agent states are introduced based on the user agent definition. They consist of session manager, session user, and free node. This paper also proposes two split session integration methods for session integration service based on these three agents. The proposed method provides session mobility flexibility and allows the user the ability to transfer, split, and retrieve a session over multiple devices. Moreover, the split session can be integrated in any devices. To ensure session mobility over multiple devices, the session integration service will be implemented through a modification of the open source project ‘SIP‐communicator’. Copyright © 2010 John Wiley & Sons, Ltd.     

An outband paging protocol for energy-efficient mobile communications

A new protocol is proposed for reducing the power consumption of battery-powered terminals in a mobile computing environment. We exploit the fact that, in a mobile data network, mobile terminals do not continuously receive data and therefore they need not continuously operate their receivers. Nevertheless, they need to check their traffic condition periodically, that is, whether there are pending data for them or not. The proposed energy-efficient protocol is based on a paging procedure wherein a dedicated channel is used to alert (page) terminals with pending traffic. Each terminal may check its traffic condition whenever it decides to by monitoring the paging channel. The protocol is evaluated through an approximated theoretical model and through computer simulation. We focus on deriving approximate formulas for the mean message delay, the message delay variance and the power consumption. It is shown that the proposed protocol can achieve considerable power saving at a cost of increased message delivery delay.     

Paging strategy optimization in personal communication systems

Mobility tracking is concerned with finding a mobile subscriber (MS) within the area serviced by the wireless network. The two basic operations for tracking an MS, location updating and paging, constitute additional load on the wireless network. The total cost of updating and paging can be minimized by optimally dividing the cellular area into location registration (LR) areas. In current systems broadcast paging messages are sent within the LR area to alert the MS of an incoming call. In this paper we propose a selective paging strategy which uses the MS mobility and call patterns to minimize the cost of locating an MS within an LR area subject to a constraint on the delay in locating the MS. This revised version was published online in July 2006 with corrections to the Cover Date.     

SIP协议栈探讨与应用

SIP(会话初始化协议)是伴随着Internet的发展同时借鉴了Web业务成功经验的、由IETF制定的一套网络多媒体信令协议,主要用于创建、修改和终止多媒体呼叫与会话,是一个与HTTP和SMTP类似的、基于文本的协议,具有易读取、易扩展以及易于调试的特性。简单介绍了SIP协议的功能组件以及消息机制,提出了SIP协议栈实现的层次结构模型,并给出了SIP协议栈的结构以及软件流程。     

IP Paging Service for Mobile Hosts

In wireless networks, mobile hosts must update the network with their current location in order to get packets delivered. Paging facilitates efficient power management at the mobile host by allowing the host to update the network less frequently at the cost of providing the network with only approximate location information. The network determines the exact location of a mobile host through paging before delivering packets destined to the mobile host. In this paper, we propose the concept of paging as an IP service. IP paging enables a common infrastructure and protocol to support the different wireless interfaces such as CDMA, GPRS, wireless LAN, avoiding the duplication of several application layer paging implementations and the inter-operability issues that exist today. We present the design, implementation, and detailed qualitative and quantitative evaluation, using measurements and simulation, of three IP-based paging protocols for mobile hosts.     

无线自组网SIP协议可靠性机制设计

随着无线自组网的发展,无线自组网提供实时多媒体业务成为重要的应用需求。作为下一代网络NGN中提供多媒体业务的SIP协议,应用于无线自组网时（特别是网络不可用时）,会出现不提示用户、提示不及时、误振铃、资源泄漏等问题。通过深入分析发现标准SIP协议的可靠性机制还不够完善,因此设计了proceeding状态超时、180响应重复发送、实时感应、紧急巡查等新的可靠性机制,建议提高180响应的重要性并将可靠临时响应机制运用到180响应,建议有状态SIP服务器必须使用SessionTimer（会话保持）机制,由此形成了比较完善的SIP协议可靠性机制。测试结果表明,完善后的SIP协议可靠性机制有效解决了网络不可用时引起的问题,在无线自组网中具有较好的适应性。     

Mobility management architectures based on joint mobile IP and SIP protocols

In the all-IP wireless networks beyond the third generation, mobility management can be effectively achieved by applying mobile IP (MIP) and the session initiation protocol (SIP) jointly. Nevertheless, an efficient combination of both protocols remains an open research issue. Conventional hybrid MIP-SIP mobility architectures operate MIP and SIP almost independently, resulting in significant redundant costs. This article investigates the representative hybrid MIP-SIP architectures and explores the joint optimizations between MIP and SIP for a more cost-efficient mobility support whilst utilizing their complementary power. Two novel design approaches are presented. The first approach culminates in a tightly integrated architecture, which merges the redundant mobility entities in MIP and SIP to yield maximum system efficiency. The other approach leads to a loosely integrated architecture, where necessary interactions are introduced between MIP and SIP mobility servers while their physical entities are kept intact. Major mobility procedures, including location update, session setup and handoff, are discussed in these architectures. The analytical results demonstrate that both proposed architectures outperform typical hybrid MIP-SIP architectures in terms of clear-cut reduced signaling costs     

Bio-inspired auto-adaptive SIP overload controller

In this article, a Session Initiation Protocol (SIP) overload control solution is proposed. It considers all the types of SIP requests. This is really what a SIP load is composed of, in an industrial environment. So far, the specialized literature considered INVITE messages only. So, we think that SIP servers are required to be dynamically adaptive to the diversity of the incoming load content. In the latter, the rate of a given SIP message type may be more or less than the other message types, depending on the services provided by the SIP server. Sometimes, it also depends on the time of the day. The auto-adaptation ability of the proposed overload control mechanism is designed after the immune system metaphor. The solution is validated through load tests and compared with a well known SIP overload control algorithm. Test load arrival patterns have been chosen to simulate three different service packages known in the SIP industry world as: Hosted Private Branch Exchange, Prepaid Calling Card Service, and Call-Shop Service.     

Optimization of SIP Session Setup Delay for VoIP in 3G Wireless Networks

Wireless networks beyond 2G aim at supporting real-time applications such as VoIP. Before a user can start a VoIP session, the end-user terminal has to establish the session using signaling protocols such as H.323 and session initiation protocol (SIP) in order to negotiate media parameters. The time interval to perform the session setup is called the session setup time. It can be affected by the quality of the wireless link, measured in terms of frame error rate (FER), which can result in retransmissions of packets lost and can lengthen the session setup time. Therefore, such protocols should have a session setup time optimized against loss. One way to do so is by choosing the appropriate retransmission timer and the underlying protocols. In this paper, we focus on SIP session setup delay and propose optimizing it using an adaptive retransmission timer. We also evaluate SIP session setup performances with various underlying protocols (transport control protocol (TCP), user datagram protocol (UDP), radio link protocols (RLPs)) as a function of the FER. For 19.2 Kbps channel, the SIP session setup time can be up to 6.12s with UDP and 7s with TCP when the FER is up to 10 percent. The use of RLP (1, 2, 3) and RLP (1, 1, 1, 1, 1, 1) puts the session setup time down to 3.4s under UDP and 4s under TCP for the same FER and the same channel bandwidth. We also compare SIP and H.323 performances using an adaptive retransmission timer: SIP outperforms H.323, especially for a FER higher than 2 percent.     

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.