DocumentCode
3371086
Title
The performance analysis of SIP-T signaling system in carrier class VoIP network
Author
Wu, Jung-Shyr ; Wang, Peir-Yuan
fYear
2003
fDate
27-29 March 2003
Firstpage
39
Lastpage
44
Abstract
The paper presents the performance modeling, analysis, and simulation of SIP-T (Session Initiation Protocol for Telephones) signaling system in carrier class VoIP (Voice over IP) network. The SIP-T signaling system defined in IETF (Internet Engineering Task Force) draft is a mechanism that uses SIP (Session Initiation Protocol) to facilitate the interconnection of PSTN with carrier class VoIP network. Based on IETF, the SIP-T signaling system not only promises scalability, flexibility, and interoperability with PSTN but also provides call control function of MGC (media gateway controller) to set up, tear down, and manage VoIP calls in carrier class VoIP network. In this paper, we analyze the queueing size (i.e. buffer size), the mean of queueing delay, and the variance of queueing delay of SIP-T signaling system that are the major performance evaluation parameters for improving QoS (quality of service) and system performance of MGC in carrier class VoIP network focused on toll by-pass or tandem by-pass of PSTN. First, we assume a mathematical model of the M/G/1 queue with non-preemptive priority assignment to represent SIP-T signaling system. Second, we present the formulas of queueing size, queueing delay, and delay variation for the nonpreemptive priority queue by queueing theory respectively. Besides, some numerical examples of queueing size, queueing delay, and delay variation are presented as well. Finally, the theoretical estimates are shown to be in excellent consistency with simulation results.
Keywords
Internet telephony; buffer storage; delays; performance evaluation; protocols; quality of service; queueing theory; telecommunication signalling; IETF; Internet Engineering Task Force; M/G/1 queue; MGC; Media Gateway Controller; PSTN interconnection; QoS; SIP-T signaling system; Session Initiation Protocol for Telephones; Voice over IP; buffer size; carrier class VoIP network; delay variation; nonpreemptive priority assignment; performance analysis; performance modeling; quality of service; queueing delay mean; queueing delay variance; queueing size; simulation; tandem by-pass; toll by-pass; Communication system signaling; Control systems; Delay; Internet telephony; Performance analysis; Protocols; Quality of service; Queueing analysis; Signal analysis; Speech analysis;
fLanguage
English
Publisher
ieee
Conference_Titel
Advanced Information Networking and Applications, 2003. AINA 2003. 17th International Conference on
Print_ISBN
0-7695-1906-7
Type
conf
DOI
10.1109/AINA.2003.1192841
Filename
1192841
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