Design and Analysis of IP Multimedia Subsystem (IMS).

Abstract

IP Multimedia Subsystem (IMS) has resulted from the work of the Third Generation Partnership Project (3GPP) toward specifying an all-IP communication service infrastructure. Mainly looking at the needs and requirements of mobile operators, the 3GPP first specified IMS as a service architecture combining the Internet’s IP technology and wireless and mobility services of current mobile telephony networks. After that the IMS architecture was extended to include fixed networks as well. The Session Initiation Protocol (SIP) is used as the signaling protocol for session establishment and control in IMS.

Measuring the capacity of the (IMS) controllers is very important due to the critical role it plays in the Next Generation Network (NGN) of the Fixed and Mobile Networks. This thesis proposes a robust and scalable method that can be used to measure the capacity of the IMS controllers, Call Session Control Function (CSCF) and benchmark their different vendors. The purpose of this method is to measure the capacity of the server in terms of how many calls are routable into a defined time interval and what the consequences of overloading the system are. The proposed method was successfully tested using a software tool called SIPp and a simulation for IMS controllers called OpenSER.

The thesis introduced two robust and scalable methods that can be used to evaluate the performance of the IP Multimedia Subsystem (IMS) controllers, Call Session Control Function (CSCF) according to ETSI and IETF Standards. The first method is the FUZZING Test and the second test is using Spectra 2| SE tool, the two proposed methods were successfully tested using a software tool called SIPp and a simulation for IMS controllers called OpenSER.

In this thesis we provide also a theoretical model that can be used by operators and network designers to determine the effects of introducing IMS to their networks in terms of bandwidth usage and the estimated delays. This model uses as the input various traffic characteristics such as the number of calls per second, losses and propagation delays. The output of the model provides details on the bandwidth needed and delay expected for successfully establishing a session when using SIP over UDP in IMS networks.

Voice traffic in IP Multimedia Subsystem (IMS) will be served using Internet Protocol (IP) which is called Voice over IP (VoIP). This thesis uses the "E-Model", (ITU-T G.107), as an optimization tool to select network and voice parameters like coding scheme, packet loss limitations, and link utilization level in IMS Network. The goal is to deliver guaranteed Quality of Service for voice while maximizing the number of users served.
The objective function for all cases is to maximize the number of calls that can be active on a link while maintaining a minimum level of voice quality (R< 70).
The cases considered are:
1. Optimization: Find voice coder given link bandwidth, packet loss level, and link utilization level.
2. Optimization: Find voice coder and packet loss level given link bandwidth and background link utilization.
3. Optimization: Find voice coder and background link utilization level given link bandwidth and packet loss level
OPNET and MATLAB are the optimization tools that were used in this part of the thesis.