**3.6 Conclusions**

IP Multimedia Subsystem (IMS) is very important due to the critical role it plays in the Next Generation Network (NGN) of the Fixed and Mobile Networks.

In this chapter we provide a theoretical model that can be used by operators and network designers to determine the effects of introducing IMS to their networks in term of bandwidth usage needed to establish IMS session. The inputs of this model are the required number of Calls or Sessions per Second, Network losses, SIP Messages size, Number of Network hops and number of ringing times. The output of this model is the bandwidth needed to insert IMS in the network.

Voice traffic in IMS will be served using Internet protocol (IP) which is called Voice over IP (VoIP). This chapter uses the "E-Model" developed by ITU-T as design tool to select network and voice parameters like coding scheme, packet loss limitations, and link utilization level in IMS Network.

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:


OPNET and MATLAB are the optimization tool that is used in this chapter.

In case 1, we found that G.723.1 is the optimized coder as it gives the maximum number of calls keeping its R factor more than 70. The quality of speech is generally higher with G.729A and G.711. But G.729A and G.711 uses more bandwidth than G.723.1. In Case 2, both G.729A and G.723.1 were sensitive to changes in packet loss, but G.711 was not as sensitive. In Case 3, voice quality was not sensitive to changes in the link load until the link load grew above approximately 94%.

The chapter also provides new equestrians can be added to enhance E-Model to relate packet loss to the level of Equipment Impairment (Ie) with different codecs.
