该文研究了IMS(IP Multimedia Subsystem)中的呼叫建立流程,以确定可以优化的信令流量,研究结果显示:S-CSCF(Serving Call Session Control Function)是IMS中的瓶颈点。为了降低会话建立时延,提高系统性能,提出了一种基于组的业务触发算...该文研究了IMS(IP Multimedia Subsystem)中的呼叫建立流程,以确定可以优化的信令流量,研究结果显示:S-CSCF(Serving Call Session Control Function)是IMS中的瓶颈点。为了降低会话建立时延,提高系统性能,提出了一种基于组的业务触发算法(Group based Service Triggering Algorithm,GSTA),然后对现有3GPP(3rd Generation Partnership Project)提出的业务触发算法(3GPP STA,3GPP Service Triggering Algorithm)和GSTA进行了性能建模,理论分析和仿真结果表明GSTA可以有效地降低S-CSCF的信令流量,增加了整个系统的吞吐量,同时显著减少了会话建立时延,提高了IMS网络的服务质量。展开更多
To initiate voice, image, instant messaging and general multimedia communication, the Session communication must initiate between two participants. SIP (Session initiation protocol) is an application layer control, wh...To initiate voice, image, instant messaging and general multimedia communication, the Session communication must initiate between two participants. SIP (Session initiation protocol) is an application layer control, which task is creating management, and terminating this kind of Sessions. With regard to the independence of SIP from the Transport layer protocols, the SIP messages can be transferred on a variety of Transport layer protocols such as TCP or UDP. The mechanism of Retransmission, which has been embedded in SIP, is able to compensate the missing Packet loss, if needed. The application of this mechanism is when SIP messages are transmitted on an unreliable transmission layer protocol such as UDP. This mechanism, while facing with SIP proxy with overload, causes excessive filling of proxy queue, delays the increase of other contacts and adds the amount of the proxy overload. We in this article, while using UDP, as the Transport layer protocol, by regulating the Invite Retransmission Timer appropriately (T1), have improved the SIP functionality. Therefore, by proposing an Adaptive Timer of Invite message retransmission, we have tried to improve the time of Session initiation and as a result, improving the performance. The performance of the proposed SIP, by the SIPP software in a real network environment has been implemented and evaluated and its accuracy and performance has been demonstrated.展开更多
文摘该文研究了IMS(IP Multimedia Subsystem)中的呼叫建立流程,以确定可以优化的信令流量,研究结果显示:S-CSCF(Serving Call Session Control Function)是IMS中的瓶颈点。为了降低会话建立时延,提高系统性能,提出了一种基于组的业务触发算法(Group based Service Triggering Algorithm,GSTA),然后对现有3GPP(3rd Generation Partnership Project)提出的业务触发算法(3GPP STA,3GPP Service Triggering Algorithm)和GSTA进行了性能建模,理论分析和仿真结果表明GSTA可以有效地降低S-CSCF的信令流量,增加了整个系统的吞吐量,同时显著减少了会话建立时延,提高了IMS网络的服务质量。
文摘To initiate voice, image, instant messaging and general multimedia communication, the Session communication must initiate between two participants. SIP (Session initiation protocol) is an application layer control, which task is creating management, and terminating this kind of Sessions. With regard to the independence of SIP from the Transport layer protocols, the SIP messages can be transferred on a variety of Transport layer protocols such as TCP or UDP. The mechanism of Retransmission, which has been embedded in SIP, is able to compensate the missing Packet loss, if needed. The application of this mechanism is when SIP messages are transmitted on an unreliable transmission layer protocol such as UDP. This mechanism, while facing with SIP proxy with overload, causes excessive filling of proxy queue, delays the increase of other contacts and adds the amount of the proxy overload. We in this article, while using UDP, as the Transport layer protocol, by regulating the Invite Retransmission Timer appropriately (T1), have improved the SIP functionality. Therefore, by proposing an Adaptive Timer of Invite message retransmission, we have tried to improve the time of Session initiation and as a result, improving the performance. The performance of the proposed SIP, by the SIPP software in a real network environment has been implemented and evaluated and its accuracy and performance has been demonstrated.
