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Investigation of SIP Signaling Messages Servicing Time

机译:SIP信令消息服务时间的调查

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As a competitive alternative to traditional telephony is increasingly being used voice over internet protocol VoIP [1]. Although similar tract to transmit voice in case of VoIP than traditional voice telephony is not produced, but it is necessary to realize the conclusion of a communication session using the appropriate signaling protocol. While the end equipment manufacturers use many different protocols, but only a few of them are standardized and recognized by all equipment manufacturers. In the initial phase, the main communication protocol to establish connection has been used H.323 protocol [2], with both positive and negative features. Of the shortcomings that led to its limited use is caused by the complexity and by replacing it with Session Initiation Protocol SIP [2] and the increasing use of multimedia services over IP networks. SIP protocol is a major next-generation networks that use the IP Multimedia Subsystem IMS [3, 4]. In order to identify the characteristics of communication session, analytical [5] and simulation [6] methods are used. However, in both cases it is necessary to know the signaling message handling time statistical characteristics, including their distribution. Often in the theses are accepted, that application service time is distributed according to exponential distribution [5, 6]. At the telephone networks that use circuit switching principle handling exponential distribution is valid for the duration of the call [7], but the IP networks that use packet transfer, it may be wrong. In order to determine the appropriate use real experimental results must be used. At time voice service is handled, signaling processing nodes accepts the different types of signaling messages and their processing length depends on the complexity of the process and the calculation of nodes resources (CPU, dynamic memory, internal bus) high-speed, productivity, etc. This work provides statistical characteristics of experimental signaling alarm service servicing at nodes time, used to service voice services, investigations.
机译:作为传统电话的竞争替代品,越来越多地使用基于Internet协议的语音VoIP [1]。尽管没有产生与传统语音电话类似的在VoIP情况下传输语音的渠道,但是有必要使用适当的信令协议来实现通信会话的结论。尽管终端设备制造商使用许多不同的协议,但是只有少数几个协议被所有设备制造商标准化和认可。在初始阶段,建立连接的主要通信协议已使用H.323协议[2],具有正向和负向特性。导致其使用受限的缺点是由于其复杂性以及用会话发起协议SIP [2]代替它以及IP网络上多媒体服务的日益使用所引起的。 SIP协议是使用IP多媒体子系统IMS [3,4]的主要下一代网络。为了识别通信会话的特征,使用了解析[5]和仿真[6]方法。但是,在两种情况下,都必须知道信令消息处理时间的统计特性,包括它们的分布。通常在这样的情况下,应用服务时间是按照指数分布分配的[5,6]。在使用电路交换原理的电话网络中,处理指数分布在通话期间是有效的[7],但是使用分组传输的IP网络可能是错误的。为了确定适当的用途,必须使用真实的实验结果。在处理语音服务时,信令处理节点会接受不同类型的信令消息,并且它们的处理长度取决于处理的复杂性以及节点资源(CPU,动态内存,内部总线)的高速性,生产率等的计算。这项工作提供了在节点时间进行的实验性信令警报服务服务的统计特性,用于服务于语音服务,调查。

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