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。     

移动IP与SIP集成应用中优化的AAA过程

在移动IP和SIP分别实现网络层和应用层移动性管理的多层多协议移动性管理方案中,当两种协议独立进行AAA操作时,存在缺乏效率的问题。为解决该问题,提出优化方案——“移动IP与SIP集成应用中优化的AAA过程”(OAPIMS)。在新一代AAA协议Diameter环境下,通过移动注册时,对两种协议的操作信令进行优化,减少了信令交互次数,达到提高效率的目的。分析表明,该方法可以明显降低信令开销,减少时延,提高系统性能。     

一种优化的SIP信令网过载控制算法

该文对SIP(Session Initiation Protocol)信令交互进行研究,重点对引起性能下降的原因进行分析,提出了一种对每个报文随机决定接收还是拒绝的SIP信令过载控制算法的优化方案。利用仿真对优化算法进行验证和分析,结果表明,该算法可以抑制吞吐量抖动,降低服务拒绝率,减小缓存队列的长度,缩短呼叫建立时延,能更好地适应高负载下对SIP信令网络的要求。     

支持多播的基于SIP的PTT服务器设计

PTT业务是近年来研究的一个热点,它将逐步成为人们生产、生活不可缺少的部分。本文结合现代通信技术的发展趋势,提出和设计了基于PC机和IP多播通信方式的PTT服务器方案,并将SIP协议作为呼叫控制信令,其中着重研究了IGMP多播协议在通信传输中的应用。     

基于SIP信令压缩算法的研究

SIP是基于文本消息的协议,因在会话建立的过程中需要传输大量的比特,加大了会话建立的时间,所以为了缩短会话建立时间,使SIP协议更好地运用于窄带环境,文章在SigComp框架结构下,通过扩充初始字典并优化编码,同时根据SIP消息的特点,提出了改进的LZW算法与Huffman编码相结合的方法,实现了对SIP信令的无损压缩,且压缩效果比较理想。     

Voice over Internet protocol (VoIP)

During the Internet stock bubble, articles in the trade press frequently said that, in the near future, telephone traffic would be just another application running over the Internet. Such statements gloss over many engineering details that preclude voice from being just another Internet application. This paper deals with the technical aspects of implementing voice over Internet protocol (VoIP), without speculating on the timetable for convergence. First, the paper discusses the factors involved in making a high-quality VoIP call and the engineering tradeoffs that must be made between delay and the efficient use of bandwidth. After a discussion of codec selection and the delay budget, there is a discussion of various techniques to achieve network quality of service. Since call setup is very important, the paper next gives an overview of several VoIP call signaling protocols, including H.323, SIP, MGCP, and Megaco/H.248. There is a section on telephony routing over IP (TRIP). Finally, the paper explains some VoIP issues with network address translation and firewalls     

基于信令消息的VoLTE接续时延优化

承载于LTE的语音业务(VoLTE)无线频谱利用效率优于传统CS语音,高清语音和视频编解码的引入显著提高了通信质量,VoLTE呼叫接续时长较传统CS语音大幅缩短.主要针对VoLTE的接续时延,提出基于信令消息的VoLTE接续时延优化,依据VoLTE终端在接续过程中不同阶段的网元间交互的信令消息,从寻呼、域选、资源预留3个阶段分别对影响VoLTE接续时延的因素进行针对性分析,并给出相应的解决方案.     

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.     

Energy- and Delay-Efficient Routing in Mobile Ad Hoc Networks

In this paper we discuss how we improved the MChannel group communication middleware for Mobile Ad-hoc Networks (MANETs) in order to let it become both delay- and energy-aware. MChannel makes use of the Optimized Link State Routing (OLSR) protocol, which is natively based on a simple hop-count metric for the route selection process. Based on such metric, OLSR exploits Dijkstra’s algorithm to find optimal paths across the network. We added a new module to MChannel, enabling unicast routing based on two alternative metrics, namely end-to-end delay and overall network lifetime. With such new module, we prove that network lifetime and average end-to-end delay improve, compared to the original OLSR protocol implementation included in the mentioned middleware. Thanks to MChannel’s approach, which implements routing in the user’s space, the improvements achieved in the unicast jOLSR routing protocol are transparently applied to the upstanding MChannel overlay multicast OMOLSR protocol. We also discuss how the proposed new module actually represents a general framework which can be used by programmers to introduce in MChannel novel metrics and path selection algorithms.     

移动自组网中基于AODV的SIP呼叫建立的性能仿真与分析

为了能够在移动自组网中使用SIP(sessioninitiationprotocol),提出了一种新的SIP呼叫建立机制。通过对AODV路由协议进行扩展使得路由发现过程与SIP呼叫建立过程同时完成。仿真结果与性能分析表明,与运行于标准AODV上的SIP呼叫建立相比,这种新提出的机制具有较低的呼叫建立时延和较高的呼叫建立成功率。     

Enabling SIP-based sessions in ad hoc networks

The deployment of infrastructure-less ad hoc networks is suffering from the lack of applications in spite of active research over a decade. This problem can be solved to a certain extent by porting successful legacy Internet applications and protocols to the ad hoc network domain. Session Initiation Protocol (SIP) is designed to provide the signaling support for multimedia applications such as Internet telephony, Instant Messaging, Presence etc. SIP relies on the infrastructure of the Internet and an overlay of centralized SIP servers to enable the SIP endpoints discover each other and establish a session by exchanging SIP messages. However, such an infrastructure is unavailable in ad hoc networks. In this paper, we propose two approaches to solve this problem and enable SIP-based session setup in ad hoc networks (i) a loosely coupled approach, where the SIP endpoint discovery is decoupled from the routing procedure and (ii) a tightly coupled approach, which integrates the endpoint discovery with a fully distributed cluster based routing protocol that builds a virtual topology for efficient routing. Simulation experiments show that the tightly coupled approach performs better for (relatively) static multihop wireless networks than the loosely coupled approach in terms of the latency in SIP session setup. The loosely coupled approach, on the other hand, generally performs better in networks with random node mobility. The tightly coupled approach, however, has lower control overhead in both the cases. This work was partially done while the author was a graduate student in CReWMaN, University of Texas at Arlington. Dr. Nilanjan Banerjee is a Senior Research Engineer in the Networks Research group at Motorola India Research Labs. He is currently working on converged network systems. He received his Ph.D. and M.S. in computer science and engineering from University of Texas at Arlington. He received his B.E. degree in the same discipline from Jadavpur University, India. His research interests include telecom network architectures and protocols, identity management and network security, mobile and pervasive computing, measures for performance, modeling and simulation, and optimization in dynamic systems. Dr Arup Acharya is a Research Staff Member in the Internet Infrastructure and Computing Utilities group at IBM T.J. Watson Research Center and leads the Advanced Networking micropractice in On-Demand Innovation Services. His current work includes SIP-based services such as VoIP, Instant Messaging and Presence, and includes customer consulting engagements and providing subject matter expertise in corporate strategy teams. Presently, he is leading a IBM Research project on scalability and performance of SIP servers for large workloads. In addition, he also works on different topics in mobile/wireless networking such as mesh networks. He has published extensively in conferences/journals and has been awarded seven patents. Before joining IBM in 2000, he was with NEC C&C Research Laboratories, Princeton. He received a B.Tech degree in Computer Science from the Indian Institute of Technology, Kharagpur and a PhD in Computer Science from Rutgers University in 1995. Further information is available at Dr. Sajal K. Das is a Professor of Computer Science and Engineering and also the Founding Director of the Center for Research in Wireless Mobility and Networking (CReWMaN) at the University of Texas at Arlington (UTA). His current research interests include sensor networks, resource and mobility management in wireless networks, mobile and pervasive computing, wireless multimedia and QoS provisioning, wireless internet architectures and protocols, grid computing, applied graph theory and game theory. He has published over 400 research papers in these areas, holds four US patents in wireless internet and mobile networks. He received Best Paper Awards in IEEE PerCom’06, ACM MobiCom’99, ICOIN’02, ACM MSwiM’00 and ACM/IEEE PADS’97. He is also recipient of UTA’s Outstanding Faculty Research Award in Computer Science (2001 and 2003), College of Engineering Research Excellence Award (2003), the University Award for Distinguished record of Research (2005), and UTA Academy of Distinguished Scholars Award (2006). He serves as the Editor-in-Chief of Pervasive and Mobile Computing journal, and as Associate Editor of IEEE Transactions on Mobile Computing, ACM/Springer Wireless Networks, IEEE Transactions on Parallel and Distributed Systems. He has served as General or Program Chair and TPC member of numerous IEEE and ACM conferences. He is a member of IEEE TCCC and TCPP Executive Committees.     

The influence of QoS routing on the achievable capacity in TDMA based Ad hoc wireless networks

The issue of providing Quality of Service (QoS) guarantees in an Ad hoc wireless network is a very challenging problem. In this paper, we make the following contributions: (i) analytically derive bounds for the end-to-end call acceptance rate using existing queueing theory methods, (ii) study the impact of the routing scheme on the end-to-end call acceptance rate, and (iii) propose a differentiated services scheme for deterministically providing QoS guarantees. Unlike the existing studies which analyze the transport capacity, we focus on the end-to-end call acceptance. The framework that we assume is that of a TDMA based Ad hoc wireless network. The routing scheme employed influences the end-to-end call acceptance of the network. The metrics that we consider are the call acceptance probability and the system saturation probability (i.e., the probability that the network is in a state in which every new call is rejected). We derive general bounds on the call acceptance and the system saturation for the case of differentiated-classes of users in the network. These bounds indicate the number of calls of the highest priority class that can be admitted into the network. Simulation studies were carried out to study the effect of load, hopcount, and the influence of the routing protocol on the call acceptance. The increase in the call acceptance rate with the introduction of load-balancing highlights the importance of load-balancing in enhancing the system performance. From these studies, we arrive at the following results: (i) load-balancing leads to significant improvement in the end-to-end call acceptance rate, and is an important factor in attaining the maximum end-to-end call acceptance rate in a given network and (ii) it is indeed possible to provide deterministic QoS guarantees for a designated set of nodes which are characterized by “deterministic guarantee limit”.     

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.     

IP多媒体子系统的安全问题剖析

IP多媒体子系统(IMS)是一种非常重要的全新体系框架,同时也是全IP网络发展的终极目标。这种框架为移动应用融合语音、视频和数据服务。IMS采用SIP作为主要信令机制,并覆盖于IP网络之上。鉴于其所处的全IP环境以及以SIP为基础的设计架构,IMS在核心网层和应用层面都呈现出了多种安全挑战。本文对IMS的体系框架和由此产生的安全挑战做了深入探讨和研究。     

Data-flow in mobile edge computing networks: end-to-end performance analysis using stochastic network calculus

Mobile edge computing (MEC) networks can provide a variety of services for different applications. End-to-end performance analysis of these services serves as a benchmark for the efficient planning of network resource allocation and routing strategies. In this paper, a performance analysis framework is proposed for the end-to-end data-flows in MEC networks based on stochastic network calculus (SNC). Due to the random nature of routing in MEC networks, probability parameters are introduced in the proposed analysis model to characterize this randomness into the derived expressions. Taking actual communication scenarios into consideration, the end-to-end performance of three network data-flows is analyzed, namely, voice over Internet protocol (VoIP), video, and file transfer protocol (FTP). These network data-flows adopt the preemptive priority scheduling scheme. Based on the arrival processes of these three data-flows, the effect of interference on their performances and the service capacity of each node in the MEC networks, closed-form expressions are derived for showing the relationship between delay, backlog upper bounds, and violation probability of the data-flows. Analytical and simulation results show that delay and backlog performances of the data-flows are influenced by the number of hops in the network and the random probability parameters of interference-flow (IF).     

SIP终端语音聊天的设计与实现

SIP协议是多媒体通信网的控制协议，在分组交换网中，它提供基本的呼叫控制，负责建立、修改和终止多媒体（话音、数据、视频等）会话等应用。概述了SIP终端软件的设计思想，并详细介绍了SIP终端软件中语音聊天功能的关键技术，从宏观和微观2个角度阐述了语音聊天功能的实现过程，构造了系统测试环境。SIP终端软件实现了语音聊天功能，具有良好的性能。     

VoLTE信令和媒体流路径优化方法研究

VoLTE是运营商在4G下的目标话音解决方案,在LTE覆盖范围内通过LTE网络提供基于IMS的话音业务,同时通过锚定和域选功能,实现VoLTE用户在CS覆盖下的主被叫业务.本文分析了现网VoLTE被叫的信令和媒体流路径情况,发现了路径迂回的问题,并且提出了一种利用S-CSCF的增强功能来实现信令和媒体流的最佳路径,经过测试验证,该方法很好地解决了路径迂回问题,缩减了端到端的接续时延.     

Seamless session mobility scheme in heterogeneous wireless networks

The next generation wireless communication system will likely be heterogeneous networks, as various technologies can be integrated on heterogeneous networks. A mobile multiple‐mode device can easily access the Internet through different wireless interfaces. The mobile multiple‐mode device thus could switch to different access points to maintain the robustness of the connection when it can acquire more resources from other heterogeneous wireless networks. The mobile multiple‐mode device therefore needs to face the handover problem in such environment. This work introduces Session Initiation Protocol (SIP)‐based cross‐layer scheme to support seamless handover scheme over heterogeneous networks. The proposed scheme consists of a battery lifetime‐based handover policy and cross‐layer fast handover scheme, called the SIP‐based mobile stream control transmission protocol (SmSCTP). This work describes the major idea of the proposed scheme and infrastructure. The proposed scheme has been implemented in Linux system. The simulation and numerical results demonstrate that the proposed SmSCTP scheme yields better signaling cost, hand‐off delay time, packet loss and delay jitter than SIP and mSCTP protocols. Copyright © 2010 John Wiley & Sons, Ltd.     

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.