IMS Application Servers: Roles, Requirements, and Implementation Technologies

The IP multimedia subsystem (IMS) defines a generic architecture to support communication services over a Session Initiation Protocol (SIP) infrastructure. In the IMS architecture, application servers host and execute the IMS service logic. These servers can be SIP application servers, open services architecture (OSA) application servers, or a customized applications for mobile networks using enhanced logic (Camel) service environment. Some technologies used in telephony and voice-over-IP (VoIP) application servers are also applicable to IMS application servers, but such servers have some unique requirements that could limit the extent to which these technologies can meet them.     

IPTV architecture for an IMS environment with dynamic QoS adaptation

This paper describes the development, implementation and evaluation of a SIP based IPTV architecture with a new dynamic QoS adaptation method and signaling structure. The implemented QoS adaptation method allows dynamic updates of session parameters, maximizing the QoE and turning the solution suitable for live multimedia streaming, independently of the cast mode (unicast or multicast). The new SIP signaling structure, for session and media control, was developed following an All-SIP approach and a hybrid SIP+RTSP approach, both suited for an IMS environment, in order to compare the system behavior and performance using either approach. The details of both the IPTV Application Server and the IPTV Client prototypical implementations are described, as well as the results of field tests carried out across different Fixed and Mobile access networks for each of the signaling structures. The proposed IPTV architecture revealed to be suitable for scalable converged networks, due to its flexible multimedia delivery of personalized streams over a variety of network infrastructures, namely, Mobile radio networks.     

基于SigComp的会话初始化协议压缩算法

独立媒体服务(IMS)是第三代合作伙伴计划(3GPP)在版本5中提出的支持IP多媒体业务的子系统,而基于文本的SIP消息过大成为其在IMS无线环境应用下的瓶颈,因此采用会话初始化协议（SIP）来建立和维护多媒体会话。在SigComp框架下,将改进后的LZSS算法与算术编码相结合对SIP信令进行压缩。实验表明新的算法有较高的压缩率,对改善IMS的SIP会话建立延时有一定的参考价值。     

IMS中SIP会话建立时延的研究

为了在3G核心网中支持多媒体会话,3GPP在R5中引入了IP多媒体子系统(IMS),并选择SIP作为IMS中的呼叫/会话的控制信令.IMS可以提供多媒体业务和语音业务,对QoS提出了更高的要求.主要讨论IMS QoS的一个重要参数-SIP会话建立时延.分析了SIP会话建立时延模型和相关因素并从3方面提出了优化措施,SIP的传输协议选择,信令重传机制的改进以及对SIP信令的压缩,最后进行了性能仿真.     

Interworking of heterogeneous access networks and QoS provisioning via IP multimedia core networks

Moving towards packet networks, where IP will have a prominent role, constitutes nowadays a widely accepted perception of future communications, the first instance of which has begun to materialise with the IP multimedia core network subsystem (IMS). By specification, IMS is the first implementation towards reaching converged communications which allows users to communicate with video, audio and multimedia content, via any fixed, mobile and wireless access network type, with controllable QoS. To enable IMS communications across heterogeneous networks, incorporating UMTS, WLAN and fixed IP access points, 3GPP and ETSI’s TISPAN currently work on schemes for controlling bandwidth allocation at the service level by employing logical interfaces that carry SIP messages. This article analyzes how interconnection between such heterogeneous networks may be performed on real platforms. In this effort, special attention is paid to the way the various interconnection possibilities can affect end-to-end QoS provisioning.     

IMS中基于LZSS的SIP信令压缩算法研究

3GPP在R5核心网引入了IMS结构，并选择会话初始协议（SIP）作为呼叫/会话的控制信令。但在无线环境下消息过长成为SIP在IMS中的应用瓶颈。在SigComp框架下，对各种压缩算法进行了比较，并根据SIP消息特点选择LZSS算法作为SIP信令压缩的基本算法。为了进一步提高SIP信令的压缩比，提出了LZSS＋Huffman的压缩算法及其实现流程，并对算法的压缩效果进行了仿真分析。     

VoIP系统SIP—QoS代理服务器设计方案

文中以解决VoIP系统的语音质量问题为目标,深入研究了基于SIP的VoIP系统QoS控制技术。参照IMS网络结构,考虑通信业务的QoS要求,研究以SIP为信令协议的VoIP系统如何进行呼叫控制、资源预留和策略决策,融人到SIP用户代理、SIP代理服务器,提出了一个具有QoS能力的SIP代理服务器的设计方案,增加了策略决策功能（PDF）等网络实体,支持QoS的能力得到增强。文中详细讨论了支持QoS的siP网络的增强能力,具有QoS能力的SIP代理服务器的功能结构,QoS功能模块,以及QoS资源授权和预留决策过程。     

Supporting Ubiquitous IMS-based Teleconferencing Through Discovery and Composition of IMS and Web Components

Natural communication among people happens in flexible ways and is strongly affected by the users’ situation (such as communication tools available, user’s location, and user’s preferences). This situation or context information is seldom used to initiate communication sessions among users. Current communication systems are indifferent about users’ context, often require time consuming manual configurations and often result in conferencing tools not being easily accessible when needed. This leads to lower adoption of innovative communications services. IMS SIP (IP Multimedia Subsystem, Session Initiation Protocol) sessions allow users to access the session from different points of contact (home, office, etc.), however, IMS still requires a prior knowledge of all SIP components that might be used in a SIP session. Furthermore, IMS makes limited use of context information (mainly user-defined availability). To address these issues our research approach combines techniques from pervasive computing with IMS networking principles to facilitate compositions of communication sessions based on users’ context. We propose a platform and APIs for pervasive application development support to allow greater intelligence in IMS applications. We additionally provide mechanisms for IMS applications to apply their intelligence to the configuration of physical devices and web resources used to set up a conference. The innovations proposed in this paper are: (1) A new standard for intelligent IMS-based conferencing applications. (2) Application Development Interfaces (APIs) for a platform for pervasive computing. (3) An architecture for a pervasive IMS platform.

Supporting SIP-based end-to-end Data Distribution Service QoS in WANs

Assuring end-to-end QoS in enterprise distributed real-time and embedded (DRE) systems is hard due to the heterogeneity and transient behavior of communication networks, the lack of integrated mechanisms that schedule communication and computing resources holistically, and the scalability limits of IP multicast in wide-area networks (WANs). This paper makes three contributions to research on overcoming these problems in the context of enterprise DRE systems that use the OMG Data Distribution Service (DDS) quality-of-service (QoS)-enabled publish/subscribe (pub/sub) middleware over WANs. First, it codifies the limitations of conventional DDS implementations deployed over WANs. Second, it describes a middleware component called Proxy DDS that bridges multiple, isolated DDS domains deployed over WANs. Third, it describes the NetQSIP framework that combines multi-layer, standards-based technologies including the OMG-DDS, Session Initiation Protocol (SIP), and IP DiffServ to support end-to-end QoS in a WAN and shield pub/sub applications from tedious and error-prone details of network QoS mechanisms. The results of experiments using Proxy DDS and NetQSIP show how combining DDS with SIP in DiffServ networks significantly improves dynamic resource reservation in WANs and provides effective end-to-end QoS management.     

IMS中一种新的SIP压缩机制

3G IP多媒体子系统（IMS）中采用会话发起协议（SIP）建立和维护多媒体会话,然而,SIP是基于文本的协议,在会话建立过程中需要传输大量的数据,导致会话建立时延的增加。对经典的LZW算法进行改进,将LZW与HUFFMAN算法和静态字典相结合,提出S-LZW-HUFFMAN算法实现对SIP消息的压缩。仿真结果表明,新算法比目前标准化组织建议采用的压缩算法具有更高的压缩效率,能够有效降低传输时延,缩短IMS中基于SIP的会话建立时延。     

IPv6与IPv4间的SIP通信安全

SIP是一种在IP网络中建立、修改和终止多媒体会话的应用层控制协议,是下一代网络的一个重要信令协议。当前的因特网主要是基于IPv4的,IPv6被设计来代替IPv4,但是在相当长一段时间内,两者将共存。IPv4与IPv6终端间的互通显得尤为重要。本文介绍了IPv4与IPv6终端间的SIP会话,同时对其会话的安全性做了讨论,并针对其安全性提出了一种解决方案。     

一种新的基于IMS的SIP重传机制

目前，IP多媒体子系统（IMS）中基于会话初始化协议（SIP）的会话建立时间受到无线信道约束带宽、帧错误率（FER）值的影响，交换的消息数量、消息长度以及重传机制对于会话建立的时延有很大的影响，延长了会话建立的时间。将基于SIP的会话建立与无线信道的性能结合，提出一种新的SIP自适应重传机制。实验结果表明，该重传机制缩短了会话建立的时间，对IMS端到端的服务质量（QoS）性能有明显的改善。     

Mathematical theory of teletraffic and its application to the analysis of multiservice communication of next generation networks

This paper presents a survey of the authors results to discuss the model of a multiservice network with “triple play” (unicast, multicast, and elastic) traffic. The main results are given for the model with multicast traffic. Two service disciplines for multicast traffic are considered in the form of systems with transparent requests. An algorithm for the calculation of the blocking probabilities for models involving both unicast and multicast traffic is proposed. In conclusion, the lines of further research are outlined: the analysis of models of the session initiation protocol (SIP), including design problems to control overloads in SIP server networks, the planning of cross-layer interfaces for orthogonal frequency division multiplexing in mobile networks long term evolution (LTE), and the design of methods for the analysis of the quality factors of the file exchange and streaming in peer-to-peer (P2P) networks.     

An intrusion detection and prevention system for IMS and VoIP services

The Voice Over IP (VoIP) environments and the most contemporary ones such as the IP Multimedia Subsystem (IMS) are deployed in order to provide cheap and at the same time high quality services to their users. Video calls, conferences, and applications can be provided to mobile devices with the lowest possible delay, while the Quality of Service (QoS) remains as the top priority for users and providers. Toward this objective, these infrastructures utilize the Session Initiation Protocol (SIP) for signaling handshakes since it is the most flexible and lightweight protocol available. However, according to many researches, it happens to be vulnerable to many attacks that threaten system’s security and availability. In this paper, we introduce a cross-layer mechanism that is able to mitigate in real-time spoofing attacks such as SIP signaling, identity theft, masquerading, and Man in the middle, and also single and distributed source flooding. It consists of three components: the policy enforcer which acts as a black list, and the spoofing and flooding modules. We also introduce a classification of SIP flooding attacks for better representation of the detection coverage. To the best of our knowledge, the proposed detection system is the most complete and accurate in terms of the attack range that is able to deter. Concerning its performance, it does not require computational expensive calculations nor resource demanding security protocols, thus being a lightweight mechanism. The experimental results have demonstrated high detection rates with false alarm rates approaching zero. Finally, it is platform independent and transparent to networks’ operations and thus can be deployed in both VoIP and IMS environments.     

在Linux环境下SIP通信的实现

SIP(Session Initiatioan Protocol)协议是由IETF工作组于1997年7月提出的,是应用层的控制协议,能建立、调整和终止多媒体的呼叫和会话。SIP协议是基于文本方式的,即以明文方式传输。SIP消息包括请求消息、应答消息。SIP协议侧重于将IP电话作为因特网上的一个应用,并且也采用RTP作为媒体传输的协议。本文在SIP通信过程中,采用面向连接的TCP来传输SIP的交互信令,采用面向无连接的UDP协议进行实时音频流传输。本设计是在Linux操作系统下,用套接字(socket)来实现的。设计实现了SIP协议的整个通信过程。最后,提出了进一步开发的设想。     

对SIP协议的研究及其基本功能的一个实例

会话初始协议SIP是IETF的多媒体数据和控制体系的一部分,它是一个基于应用层的控制协议,用来邀请用户参加多媒体会议。SIP不依附于任何一种特定的会议控制协议。SIP是被设计成独立于低层的传输协议的并且可以用一些额外的功能来进行扩展。本文介绍了SIP的工作原理和一些操作并且给出了一人两个用户SIP进行初始化的简单例子。     

下一代网络多媒体会话中H.248与SIP之间协议协作的研究

H.248和SIP是下一代电信网络的两个核心协议,通过H.248和SIP之间的协作来完成多媒体会话的建立、调整和删除是在下一代网络中实现多媒体业务需面临的重要问题。文章在深入研究H.248和SIP协议的基础上针对协议协作的应用环境抽象了一个协作模型。依据此模型,对两方多媒体会话和多方多媒体会话中各信令交互阶段H.248和SIP之间的协作进行了详细阐述。对H.248和SIP协议之间协作的研究可以应用到下一代电信网络的工程技术领域。     

Performance evaluation of a mesh-evolving quality-of-service-aware multicast routing protocol for mobile ad hoc networks

The tremendous amount of multimedia applications running across the wireless communication medium makes quality of service (QoS) a fundamental requirement for mobile ad hoc networks. However, it is not easy to incorporate QoS into these networks. Moreover, the growing number of group-oriented applications also necessitates the efficient utilisation of network resources. The multicast model is a promising technique which can achieve this efficiency by facilitating the inherent broadcast capability of the wireless medium. The mesh-evolving ad hoc QoS multicast (MAQM) routing protocol is developed to address the resource efficiency and QoS problems with one, integrated solution. MAQM achieves multicast efficiency by tracking the availability of resources for each node within its neighbourhood. The QoS status is monitored continuously and announced periodically to the extent of QoS provision. Using these features, MAQM nodes can make their decisions on joining a new multicast session based on the sustainability of their perceived QoS. MAQM also evolves the initial multicast tree into a mesh during the course of an ongoing session to achieve a more robust network topology. Thus, MAQM integrates the concept of QoS-awareness into multicast routing in mobile ad hoc networks. Since ad hoc networks require the protocol control overhead to be as small as possible, we analyse the multicast session establishment process of MAQM to see its impact on the protocol performance in terms of system control overhead. We also evaluate the performance of MAQM through computer simulations using various qualitative and quantitative criteria. The simulation results validate our mathematical analysis of the control overhead and show that MAQM significantly improves multicast efficiency through its QoS-aware admission and routing decisions with an acceptably small overhead. Thus, MAQM shows that QoS is not only essential for, but also applicable to mobile ad hoc networks.