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。     

IMS中基于LZSS＋Huffman的信令压缩算法研究

SIP虽然被选定为IMS的核心呼叫/会话控制协议,但消息过长也成为SIP在IMS无线环境下应用时的瓶颈.在SigComp框架下,通过对各种压缩算法的比较,结合SIP协议消息特点,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协议消息的压缩比,提出了基于LZSS Huffman的信令压缩(SigComp)算法,并对该算法的压缩效果进行了仿真分析.     

IMS中一种新的基于SigComp的SIP压缩机制

IMS(IP Multimedia Subsystem)中采用SIP(Session Initiation Protocol)协议建立和维护多媒体会话。然而,SIP是基于文本消息的协议,在会话建立的过程中需要传输大量的比特,加大会话建立的时延。该文基于SigComp(Signaling Compression)框架结构,将改进后的LZW算法和HUFFMAN算法相结合,提出LZW-HUFFMAN算法。实验结果表明,新算法具有更高的压缩效率,有效地降低了传输时延,缩短了会话建立的时间。     

SIP信令压缩技术

信令压缩指的是将原本设计用于Internet的多媒体会话控制IP信令协议应用于无线通信时,必须采取的信息压缩技术,其目的是降低带宽开销,提高无线频谱资源利用率。文中详细介绍了信令压缩的体系结构、各实体的功能、消息格式和各种扩展操作,并使用LZHUF算法,结合静态字典和共享压缩机制,实现了SIP信令压缩。仿真结果表明,信令压缩技术取得了很好的压缩效果,压缩后消息的长度平均仅为原来消息长度的20％左右。     

不同压缩算法性能的研究

移动通信网络中，建立一次多媒体会话通常需要数十秒钟的时间。通过对信令消息的压缩可以有效地减少所需时延，改善服务质量。文章先后使用了LZ77、LZW和Deflate压缩算法对因特网协议多媒体子系统中的会话发起协议信令消息的压缩进行了研究。仿真结果显示：Deflate算法的压缩性能最好，不使用辞典时压缩率可以达到30％，使用辞典时压缩率可以达到50％以上；接着是LZW算法，而LZ77算法的压缩性能最差。     

SIP体系结构的研究

SIP协议是一个用于建立,更改和终止多媒体会话的应用层控制协议.它是IETF多媒体数据和控制体系结构的一部分并借鉴了许多已有的Internet协议,具有简单,便于扩展和扩充等特点.主要讲述了SIP消息、SIP体系结构和呼叫处理流程.     

基于代理转发的NAT穿越方案

会话启动协议(SIP)是应用层控制协议,用于在IP网上建立、修改及终止多媒体会话或呼叫。它是基于文本的协议,无法保证SIP消息和媒体流顺利穿过网络地址转换(NAT)和防火墙(FW),使位于私网内的用户不能利用SIP协议进行会话,从而限制了它的发展和应用。文章提出了一种新的私网穿越方案,该方案通过增设一个代理服务器,使私网终端将消息流和媒体流经该服务器转发,从而实现了穿越。     

基于SIP的应用层移动性管理

无线通信网络能为用户提供丰富的多媒体服务,而多媒体传输需要多媒体会话的建立和维护。由于无线终端的移动性,移动性管理成为无线网络中的重要技术。SIP是多媒体会话建立和维护的信令协议,已得到广泛应用。文章详细介绍SIP协议及基于SIP协议的应用层移动性解决方案。     

IMS中SIP会话建立时延的改进机制研究

为了减小会话建立时间，IMS中有必要对SIP消息进行压缩。文章首先给出了一种SIP会话建立时延模型，接着分析了各种与时延相关的因素并提出了改进SIP会话建立时延的方法，最后从SIP压缩角度提出了一种改进SIP会话建立时延的改进机制。     

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。     

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.     

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.     

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。     

A new call control scheme of Push-To-Talk system

For Push-To-Talk (PTT) system based on Public Mobile Data Network (PMDN), the end-to-end time delay is the key aspect of the user’s experience. The Push-Over-Cellular (POC) scheme defined by Open Mobile Alliance (OMA) is based on the VoIP phone model and use SIP protocol as the call control scheme. The call setup time delay in SIP may reach to several seconds, which is unacceptable for the PTT service. In this paper, we provide a new call control scheme for PTT system based on PLMN network. By combining the apriority knowledge of PTT call model and the priority control scheme, we encapsulate the signaling message and the voice data into a same data packet, when the user push the button, the voice and the call control signaling are sent to the server at the same time. So the long time delay of call setup procedure of POC scheme can be eliminate. The end-to-end call delay can be decreased significantly. The experiment result based on the commercial CDMA2000 1X network of China Unicom shows that the call delay can be decreased to 600 ms, which approach to the traditional trunk communication system’s requirement.