**3.2 Project setup**

82 Mobile Networks

The loss impairment Ie captures the distortion of the original voice signal due to low-rate codec, and packet losses in both the network and the play out buffer. Currently, the E-Model (ITU-T Rec. G.107 2005) can only cope with speech distortion introduced by several codecs i.e. G.729 (ITU-T Rec. G.729 2007) or G.723 (ITU-T Rec. G.723 2006). Specific impairment factor values for codec operation under random packet-loss have formerly been treated using tabulated, packet-loss dependent Ie values. Now, the Packet-loss Robustness Factor Bpl is defined as codec specific value. The packet-loss dependent Effective Equipment Impairment Factor Ie-eff is derived using the codec specific value for the Equipment Impairment Factor at zero packet-loss. Ie and the Packet-loss Robustness Factor Bpl are listed in Table I for several codecs. With the Packet-loss Probability Ppl, (ITU-T Rec. G.113

Ppl Ie eff Ie 95 Ie Ppl Bpl

Ie,eff represents the packet loss dependent effective equipment Impairment factor,

As can be seen from this formula (44), the Effective Equipment Impairment Factor in case of Ppl = 0 (no packet-loss) is equal to the Ie value defined in Table 4.Ie represents the effect of degradation introduced by codecs, Packet Loss. (ITU-T Rec. G.113 2007) provides

Table 4. Provisional planning values for the equipment impairment factor Ie and for packet-

Coming to the VoIP traffic Characterization, Human speech is traditionally modeled as sequence of alternate talk and silence periods whose durations are exponentially distributed and referred as to ON-OFF model. On the other hand all of the presently available codecs with VAD (Voice Activity Detection) have the ability to improve the speech quality by reproducing Speakers back ground by generating special frame type called SID (Silence

The main output from the E-model is the single R value, produced by an equation combining all relevant impairments. The R factor ranges from 0-100 but is basically

Bpl is called the packet-loss robustness factor, which depends on the used codec.

derived from the value of Ie depending on codec and at zero packet loss.

parameters for use in calculating Ie from codec type and Packet Loss rate.

Insert Descriptor). SID frames are generated during Voice Inactivity Period.

(45)

**3.1.2 Calculation of the equipment impairment Ie** 

2007).Ie-eff is calculated using formula (44)

Ie is the equipment impairment factor.

loss robustness factor Bpl (ITU-T Rec. G.113 2007)

**3.1.3 The R factor** 

Ppl is the packet loss probability

The simulation is done using OPNET simulation tool IT Guru Academic Edition 9.1 for VoIP in IMS network using SIP Protocol. The network consists of IP-Telephones (VoIP or IMS Clients) connected to the Internet by routers which act as IP gateway, the network is managed by the SIP proxy server (act as P-CSCF) which uses the SIP protocol to establish the voice calls (VoIP) on the IMS network as shown in figure 3. The links between the routers and the Internet are T1 with link speed 1.544 Mbps and the links between the dialer, dialed, Proxy Server and the routers are 1000 Base-x. The idea is to configure the network with a certain parameters and run the simulation then getting from the tool the result values which used in E-Model equations to measure the Quality of service Factor R. 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). The cases considered are:

1. Find the optimal voice coder given link bandwidth, packet loss level, and background link utilization level.

Fig. 3. Network Topology

Design and Analysis of IP-Multimedia Subsystem (IMS) 85

Table 7. Codec Parameters for case1-1 (ITU-T Rec. G.107 2005) & (ITU-T Rec. G.113 2007)

Table 8. Codec Parameters for case1-2 (ITU-T Rec. G.107 2005) & (ITU-T Rec. G.113 2007)

>70)

Table 9. OPNET Results for case (1)

but lower quality of voice.

middle R value. As shown in figure 5. Figure 4 shows the average packet end to end delay for different codecs and figure 6 shows the number of connected calls for different coders. Table 9 contains data collected from OPNET in this case of 4 hours observation and shows that G.723.1 provides the maximum number of calls with accepted voice quality (R=78.2

The results of this case are shown in Figure 6 not surprising, as G.723.1 is a more efficient



Table 6 shows standard parameters for each codec used in the analysis

Table 6. Codec Parameters
