{"title":"Comparative Study of Secure vs. Non-secure Transport Protocols on the SIP Proxy Server Performance: An Experimental Approach","authors":"Sureshkumar V. Subramanian, R. Dutta","doi":"10.1109/ARTCOM.2010.90","DOIUrl":null,"url":null,"abstract":"The wide scale deployment of Internet combined with several advancements in hardware and software technologies created opportunities for several Internet based applications such as Voice Over IP (VoIP) that involves the delivery of voice, video and data to the end user. SIP, the Session Initiation Protocol, is a signaling protocol for Internet conferencing, telephony, presence, events notification and instant messaging. The SIP Proxy Server is a call control software package that enables service providers to build scalable, reliable Voice over IP networks today and it provides a full array of call routing capabilities to maximize network performance in both small and large packet voice networks. The SIP Proxy Server can perform a digest authentication of SIP Register, invite requests, and can encrypt SIP requests and responses using Transport Layer Security (TLS). For secure communication, user authentication, confidentiality and integrity of signaling message and SIP session establishment are essential. TLS is used for the secured SIP signaling and Secure Real Time Protocol (SRTP) for secured SIP session establishment. This paper is focused on evaluating the performance impacts of SIP proxy servers when secured (TLS) over non-secured transport protocols (TCP/UDP) are used to transport SIP messages. Several experiments were conducted in a lab environment to evaluate key performance parameters such as Call setup time, Mean number of calls, Memory utilization, CPU utilization and queue size.","PeriodicalId":398854,"journal":{"name":"2010 International Conference on Advances in Recent Technologies in Communication and Computing","volume":"60 1","pages":"0"},"PeriodicalIF":0.0000,"publicationDate":"2010-10-16","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":"18","resultStr":null,"platform":"Semanticscholar","paperid":null,"PeriodicalName":"2010 International Conference on Advances in Recent Technologies in Communication and Computing","FirstCategoryId":"1085","ListUrlMain":"https://doi.org/10.1109/ARTCOM.2010.90","RegionNum":0,"RegionCategory":null,"ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":null,"EPubDate":"","PubModel":"","JCR":"","JCRName":"","Score":null,"Total":0}
引用次数: 18
Abstract
The wide scale deployment of Internet combined with several advancements in hardware and software technologies created opportunities for several Internet based applications such as Voice Over IP (VoIP) that involves the delivery of voice, video and data to the end user. SIP, the Session Initiation Protocol, is a signaling protocol for Internet conferencing, telephony, presence, events notification and instant messaging. The SIP Proxy Server is a call control software package that enables service providers to build scalable, reliable Voice over IP networks today and it provides a full array of call routing capabilities to maximize network performance in both small and large packet voice networks. The SIP Proxy Server can perform a digest authentication of SIP Register, invite requests, and can encrypt SIP requests and responses using Transport Layer Security (TLS). For secure communication, user authentication, confidentiality and integrity of signaling message and SIP session establishment are essential. TLS is used for the secured SIP signaling and Secure Real Time Protocol (SRTP) for secured SIP session establishment. This paper is focused on evaluating the performance impacts of SIP proxy servers when secured (TLS) over non-secured transport protocols (TCP/UDP) are used to transport SIP messages. Several experiments were conducted in a lab environment to evaluate key performance parameters such as Call setup time, Mean number of calls, Memory utilization, CPU utilization and queue size.
互联网的大规模部署与硬件和软件技术的一些进步相结合,为几种基于互联网的应用创造了机会,例如IP语音(VoIP),它涉及向最终用户传输语音、视频和数据。SIP,即会话发起协议,是一种用于互联网会议、电话、到场、事件通知和即时消息传递的信令协议。SIP代理服务器是一个呼叫控制软件包,使服务提供商能够构建可扩展的、可靠的IP语音网络,它提供了完整的呼叫路由功能,以最大限度地提高小型和大型分组语音网络的网络性能。SIP代理服务器可以执行SIP注册的摘要身份验证,邀请请求,并可以使用传输层安全(TLS)加密SIP请求和响应。为了实现安全通信,用户认证、信令消息的保密性和完整性以及SIP会话的建立是必不可少的。安全的SIP信令采用TLS,建立安全的SIP会话采用SRTP (Secure Real Time Protocol)。本文的重点是评估当使用安全(TLS)而不是非安全传输协议(TCP/UDP)来传输SIP消息时SIP代理服务器的性能影响。在实验室环境中进行了几个实验,以评估关键性能参数,如呼叫建立时间,平均呼叫数,内存利用率,CPU利用率和队列大小。