Vol. 5, No. 6 June 2014 ISSN Journal of Emerging Trends in Computing and Information Sciences CIS Journal. All rights reserved.

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1 Vol. 5, No. 6 June 04 ISSN CIS Journal. All rights reserved. Performance Evaluation of MPLS TE Signal Protocols with Different Audio Codecs Mahmoud M. Al-Quzwini, Shaimaa A. Sharafali, Computer Engineering Department, College of Engineering, Nahrain University, Baghdad, Iraq ABSTRACT This paper studies the performance of MPLS networks TE signal protocols with different voice codecs including PCM (64 Kbps), GSM FR (3 Kbps), G.73. (5.3 Kbps), G.76 (6 Kbps), G.78 (6 Kbps), G.79 (8 Kbps) and IS-64(7.4 Kbps). Simulation results show that the MPLS network with CR-LDP TE signal protocol outperforms the MPLS network with RSVP TE signal protocol in terms of the total amount of received voice packets, voice packet delay variation, voice jitter, and the number of maintained calls for all voice codecs. The results also show that G.73. codec type gives better results in terms of the number of maintained calls, but with least voice quality compared with other voice codecs. Keywords: MPLS, Traffic Engineering, VoIP, CODECS, CR-LDP, RSVP.. INTRODUCTION The high increase in the number of internet users made services such as telephone and television to reach their customers via the internet and this has been forcing Internet Service Providers (ISPs) to improve their quality of service. With this increase as well as the advances made in real-time applications (voice and video), the traditional routers have the challenges of providing the required high bandwidth, fast routing as well as quality of service support. Due to the challenges of traditional routers to provide these requirements especially for voice and video, methods such as the use of Multiprotocol Label Switching (MPLS) and so on are now used []. MPLS is not designed to replace IP; it is designed to add a set of rules to IP so that traffic can be classified, marked, and policed. MPLS as a traffic-engineering tool has emerged as an elegant solution to meet the bandwidth management and service requirements for next generation Internet Protocol (IP) based backbone networks []. WAN bandwidth is probably the most expensive and important component of an enterprise network, Network administrators must know how to calculate the total bandwidth that is required for Voice traffic and how to reduce overall utilization, a description in detail for coder-decoders (Codecs), codec complexity and the bandwidth requirements for VoIP calls. A codec is a device or program capable of performing encoding and decoding on a signal or digital data stream. Many types of codecs are used to encode-decode or compress/decompress various types of data that would otherwise use a large amount of bandwidth on WAN links. Codecs are especially important on low-speed serial links where every bit of bandwidth is needed and utilized to ensure network reliability [3].. RELATED WORKS Analyzing and optimizing voice traffic over data networks have been a major challenge to researchers and developers, many techniques have been proposed based on analyses from real word and simulated traffic. Mahesh Kr. Porwal [4] made a comparative analysis of MPLS over Non-MPLS networks and showed that MPLS have a better performance over IP networks, through this paper a comparison study has been made on MPLS signaling protocols (CR-LDP, RSVP and RSVP- TE) with Traffic Engineering by explaining their functionality and classification. The Simulation of MPLS and Non-MPLS network is done; performance is compared by with consideration of the constraints such as packet loss, throughput and end-to-end delay on the network traffic. Ravi Shankar Ramakrishnan et al.[5] analyzed three commonly used codecs using peer-to-peer network scenario. The paper presents OPNET simulator and they were considered only in Latency, Jitter and Packet loss. They were able to present from the results that G.7 is an ideal solution for PSTN networks with PCM scheme. G.73 is used for voice and video conferencing however provides lower voice quality. Music or tones such as DTMF cannot be transmitted reliably with G.73 codec. G.79 is mostly used in VoIP applications for its low bandwidth requirement that s why this type is mostly common on the WAN connections and to transport voice calls between multisite branches. Md. Arifur Rahman [6] calculated the minimum number of VoIP calls that can be created in an enterprise IP network. The paper presents OPNET simulator designing of the real-world network model. The model is designed with respect to the engineering factors needed to be reflected when implementing VoIP application in the IP network. Simulation is done based on IP network model to calculate the number of calls that can be conserved Sarmad K. Ibrahimet al. [7] studied the performance of MPLS networks with TE signal protocols in relation with voice codecs. Simulation were performed and compared for a multisite network with PCM and GSM based VoIP. Simulation results show that the MPLS network with CR-LDP TE signal protocol outperforms 447

2 Vol. 5, No. 6 June 04 ISSN CIS Journal. All rights reserved. the MPLS network with RSVP TE signal protocol in terms of both the total amount of received voice packets and the number of maintained calls for both voice codecs. The main goal of our research is to study the performance of traffic engineering signal protocols (CR LDP and RSVP)with voice codecs for voice over MPLS network. 3. TRAFFIC ENGINEERING IN MPLS NETWORKS Traffic Engineering (TE) is a mechanism put in place to control the flow of traffic in networks and it provides the performance optimization of the network resources. The main characteristics of TE are faulttolerance, optimum resource 448utilization and resource reservation [8]. The basic objective of the consideration of TE is to improve quality of service of some applications and use the available network resources efficiently. There are some important factors, which are needed for TE. These factors are; Path Selection, Traffic Management, Direction of Traffic along Computed Paths and Distribution of Topology Information. The LSPs in the MPLS network are established and the labels are distributed on each of the hops along the LSPs before packets could be forwarded. The LSPs can be established either by explicitly routed LSP or control driven LSP. Control driven LSPs can also be referred to as hop-byhop LSP and are set by the use of LDP protocol. Explicitly routed LSPs can also be referred to as constraint based LSPS (CR-LSPs), which are specified in the setup message. At each hop, a label request is sent to the next hop along the LSP [9].There are basically two protocols used to set CR-LSPs in MPLS. These protocols are; Resource Reservation Protocol (RSVP) and Constraint based routed LDP (CR-LDP). 3. Constraint Based Routed LDP (CR-LDP) CR-LDP is an extension of LDP to support constraint based routed LSPs. The term constraint implies that in a network and for each set of nodes there exists a set of constraint that must be satisfied for the link or links between two nodes to be chosen for an LSP [0]. CR- LDP is capable of establishing both strict and loose path setups with setup and holding priority, path Preemption, and path re-optimization []. CR-LDP and LDP protocols are hard state protocols that means the signaling message are sent only once, and don t require periodic refreshing of information. In CR-LDP approach, UDP is used for peer discovery and TCP is used for session advertisement, notification and LDP messages. CR-LSPs in the CR-LDP based MPLS network are set by using Label Request message. The Label Request message is the signaling message which contains the information of the list of nodes that are along the constraint-based route. In the process of establishing the CR-LSP the Label Request message is sent along the constraint-based route towards the destination. If the route meet the requirements given by network operator or network administrator, all the nodes present in route distribute the labels by means of Label Mapping message. 3. Resource Reservation Protocol (RSVP-TE) RSVP-TE is an extension of RSVP that utilizes the RSVP mechanisms to establish LSPs, distribute labels and perform other label-related duties that satisfies the requirements of TE []. The revised RSVP protocol has been proposed to support both strict and loose explicit routed LSPs (ERLSP). For the loose segment in the ER- LSP, the hop-by hop routing can be employed to determine where to send the PATH message [0]. RSVP is a soft state protocol. It uses Path and RSVPcommands to establish path. The CR-LSPs established by RSVP signaling protocol in MPLS network is described by the following steps: a. The Ingress router in the MPLS network selects a LSP and sends the Path message to every LSR along that LSP, describing that this is the desired LSP used to establish as CR-LSP. b. In this process the Path and RSVP messages are send periodically to refresh the state maintained in all LSRs along the CR-LSP [3]. c. The LSRs along the selected LSP reserve the resources and that information is send to Ingress router using the RSVP message. 4. VOIP CODECS There are many codecs available for audio, video and text. We used in our evaluations some of G.7xx of ITU-T standards for audio compression and decompression. Table. shows number of compression schemes. Table : Common Audio Codecs Codecs types G.7 GSM FR G.73. G.76 G.78 G.79A IS-64 Bandwidth /Kbps /6.4 6/4/3/ Algorithm PCM RPE LTP ACELP ADPCM LD-CELP CS-ACELP ACELP The popular voice codecs used in the telecommunication industry are G.7 which is widely used in the PSTN environment [4, 5].G.7 represents logarithmic pulse-code modulation (PCM) with 8 bits samples for signals of voice frequencies, sampled at the rate of 8000 samples/second, on 64 kbps channel. Using G.7 audio codec for VoIP will give the best voice quality; as it uses no compression and it is the same codec used by all Public Switched network and ISDN lines. It sounds just like using a regular phone or ISDN phone. But this codec takes more bandwidth then other codecs, up to 84 Kbps including all TCP/IP overhead. However, with increasing broadband bandwidth, this should not be a problem [6]. G.73. is the ITU-T standard that specifies the coded representation for speech in PSTN using Algebraic Code-Excited Linear Prediction (CELP) coding rates at 448

3 Vol. 5, No. 6 June 04 ISSN CIS Journal. All rights reserved. 5.3Kbit/s and Multiples Maximum Likelihood Quantization (MP-MLQ) at 6.3 Kbit/s. The 6.3 Kbit/s provides very good voice quality whereas the lower bit rate provides good quality with some more functionality [7]. G.76 is the Recommendation for speech coding at 40, 3, 4, and 6 Kbit/s (variable bit rates) using Adaptive Differential Pulse Code Modulation (ADPCM) transcoding technique [8]. routers in each network are connected with DS3 cable with data rate of G.78 is the ITU-T Recommendation for speech coding at 6 Kbit/s utilizing Low-Delay Code-Excited Linear Prediction Coding (LD-CELP) [9]. G.79a this annex provides the high level description of a reduced complexity version of the G.79 speech codec. This version is bit stream interoperable with the full version, i.e. a reduced complexity encoder may be used with a full implementation of the decoder, and vice versa [0]. Offers toll quality speech at a low bit rate of 8Kbps using CS-ACELP (Conjugate Structure Algebraic Code Excited Linear Prediction). However, it is a rather "costly" codec in terms of CPU processing time; therefore some VoIP phones and adapters can only handle one G.79 call (channel) at a time. This codec provides robust performance but at the price of its complexity. This can cause calls to fail if the user attempts to use three-way calling, or place simultaneous calls on both lines of a two-line device, and G.79 is the only allowed codec [6]. Another standard used in our evaluations is ETSI GSM. The GSM system uses Linear Predictive Coding with Regular Pulse Excitation (LPC-RPE codec). It is a full rate speech codec and operates at 3 Kbits/sec. As a comparison, the old public telephone networks use speech coding with bit rate of 64 Kbit/s [6]. And IS-64 speech coding is based on the ACELP (Algebraic-code-excited linear prediction) []. The bit rate of the speech codec is 7.4 Kbit/s. The standard has been superseded by TIA/EIA SIMULATION The simulation environment employed in this paper is based on OPNET 4.5 simulator which is extensive and powerful simulation software. Figures and show two different MPLS networks, each one is simulated with CR-LDP and RSVP TE signal protocols. To simulate real network environments voice, video, HTTP, FTP, DB, Telnet and applications are used in each network. The VoIP traffic is sent from source (voice ) to destination (voice ), the video traffic is sent from source (video ) to destination (video ), DB and HTTP traffic is sent from source (DB, HTTP) to destination (DB, HTTP server), remote traffic is sent from source (remote) and FTP traffic is sent from source (FTP) to destination (FTP, remote server). The network in figure. consists of six routers and four switches and in figure. consists of eight routers and four switches. These Fig : first MPLS network topology Fig : Second MPLS network topology Mbit/s. The end nodes are connected to the network via switches. Links of each switch are 00BaseT. The voice workstations use different types of codecs, namely, PCM (64 kbps), GSM FR (3 kbps), G.73. (5.3 kbps), G.76 (6 kbps), G.78 (6 kbps), G.79a (8 kbps), and IS-64 (7.4 kbps) each type of codecs simulated individually and their results shown in figures 3 through 6, for the sent and received voice packet. Figures 7 through 30 show results of packet delay variation and end to end delay. Voice jitter and main opinion score results are depicted in figures 3 through PERFORMANCE METRICS In our simulations, we use the following metrics to evaluate the performance of MPLS network. 6. Mean Opinion Score (MOS) MOS provides a numerical measure of the quality of human speech in voice telecommunications, with value ranging from to 5 where is the worst quality and 5 is the best quality. In our simulation, we compute MOS through a non-linear mapping from R- factor as in []: Where; R : the effect of impairments that occur with the voice signal, : the impairments caused by different types of losses occurred due to codec's and network, and : represents the impairment caused by delay particularly mouth-to-ear delay. Using the default setting for and A Eq. () can be reduced to: 449

4 Vol. 5, No. 6 June 04 ISSN CIS Journal. All rights reserved. 6. Packet end-to-end delay (Dee) The total voice packet delay; Dee represent in this formula: Where,,, and represent the network, encoding, decoding, compression and decompression delay, respectively [3]. 6.3 Jitter The jitter is defined as the signed maximum difference in one-way delay of the packets over a particular time interval. Let t(i) and t (i) be the time transmitted at the transmitter and the time received at the receiver, respectively. Jitter is then, calculated as in [4]: According to equation (3), the jitter value can be negative which means that the time difference between the packets at the destination is less than that at the source. 6.4 Packet delay variation (PDV) Packet delay variation plays a crucial role in the network performance degradation and affects the userperceptual quality. Higher packet delay variation results in congestion of the packets, which can results in the network overhead. PDV is defined as the variance of the packet delay, which can be, calculated from the following Equation [4]. Where is the average delay of the n selected packets. 7. RESULTS AND DISCUSSION 7. MOS The MOS values shown in table. and figures 3 through 44 indicate that the voice calls with G.73. codec have the less quality than calls with other types of codecs. 7. Number of Maintained Calls The voice delay can be divided into three contributing components which are described as follows [3, 5]: () (3) (4) The delay introduced by the G.7 codec for encoding and packetization are ms and 0 ms respectively. The delay at the sender considering above two delays along with compression is approximated to a fixed delay of 5 ms; At the receiver the delay introduced is from buffering, decompression, depacketization and playback delay. The total delay due to the above factors is approximated to a fixed delay of 45 ms. The overall network delay can be calculated from the above sender and receiver delays to be 80 ms approximately ( ). The 50 ms represents the maximum acceptable end-to-end delay so that the quality of the established VoIP call is acceptable [5]. In this paper the traffic drop time is used to calculate the number of maintained calls. In all simulations, the values of the packet end-to-end delay when the traffic drops were all less than the maximum acceptable end to end delay. This will show the difference among different codecs. Then the number of maintained calls = (drop time start time) / (5) The voice call start at 0 sec, and the drop time for each scenario is shown in table.5, and the number of calls maintained are shown in table.6. The results show that the number of maintained calls when using CR LDP TE signaling protocol is greater than those with using RSVP TE signaling protocol with all codecs. 7.3 Jitter Table.3 and figures 3 through 44shows the results of maximum jitter values for different scenarios. Except the GSM FR and G.76 codecs in the first network, it is observed that CRLDP TE signal protocol has lower values than RSVP TE. 7.4 Packet Delay Variation (PDV) Table.4 and figures 7 through 30show the results of packet delay variation for different scenarios. Except the GSM FR codec, it is observed that CRLDP TE signal protocol has lower values than RSVP TE. Table : Summary statistics of MOS values experienced Codecs Network MOS value Types number CR LDP RSVP PCM GSM G G

5 Vol. 5, No. 6 June 04 ISSN CIS Journal. All rights reserved G G.79A IS Table 3: Summary of maximum jitter values experienced Codecs Types PCM GSM FR G.73. G.76 G.78 G.79A IS-64 Network number Jitter value (sec) Max. CR LDP RSVP Table 4: summary of voice packet delay variations Codecs types PCM GSM FR G.73. G.76 G.78 G.79A IS-64 Network number Packet Delay Variations (sec) CR LDP RSVP Codecs types PCM GSM FR G.73. G.76 G.78 G.79A IS-64 Table 5: summary of traffic drop time Table 6: summary of number of maintained calls Codecs types PCM GSM FR G.73. G.76 G.78 G.79A IS-64 Network number Network number Traffic drop time(sec) CR LDP RSVP Number of calls/sec CR LDP RSVP Table 7: summary of End-to-End delay values Codecs Network End-to-End delay (sec) types number CRLDP RSVP PCM GSM FR G G G G.79A IS

6 Vol. 5, No. 6 June 04 ISSN CIS Journal. All rights reserved. Fig 3: st network PCM send and received voice traffic Fig 5: st network G.76 send and received voice traffic Fig 6: st network G.78 send and received voice traffic Fig 4: st network G.73.send and received voice traffic 45

7 Vol. 5, No. 6 June 04 ISSN CIS Journal. All rights reserved. Fig 7: st network G.79 send and received voice traffic Fig 9: st network IS-64 send and received voice traffic Fig 8: st network GSM FR send and received voice traffic Fig 0: nd network PCM send and received voice traffic 453

8 Vol. 5, No. 6 June 04 ISSN CIS Journal. All rights reserved. Fig : nd network G.73. send and received voice traffic Fig 3: nd network G.78 send and received voice traffic Fig : nd network G.76 send and received voice traffic Fig 4: nd network G.79 send and received voice traffic 454

9 Vol. 5, No. 6 June 04 ISSN CIS Journal. All rights reserved. Fig 5: nd network GSM FR send and received voice traffic Fig 7: st network PCM PDV and EE Delay Fig 8: st network G.73. PDV and EE Delay Fig 6: nd network IS-64 send and received voice traffic 455

10 Vol. 5, No. 6 June 04 ISSN CIS Journal. All rights reserved. Fig 9: st network G.76 PDV and EE Delay Fig : st network G.79 PDV and EE Delay Fig 0: st network G.78 PDV and EE Delay Fig : st network GSM FR PDV and EE Delay 456

11 Vol. 5, No. 6 June 04 ISSN CIS Journal. All rights reserved. Fig 3: st network IS-64 PDV and EE Delay Fig 5: nd network G.73. PDV and EE Delay Fig 4: nd network PCM PDV and EE Delay Fig 6: nd network G.76 PDV and EE Delay 457

12 Vol. 5, No. 6 June 04 ISSN CIS Journal. All rights reserved. Fig 7: nd network G.78 PDV and EE Delay Fig 9: nd network GSM FR PDV and EE Delay Fig 30: nd network IS-64 PDV and EE Delay Fig 8: nd network G.79 PDV and EE Delay 458

13 Vol. 5, No. 6 June 04 ISSN CIS Journal. All rights reserved. Fig 3: st network PCM jitter and MOS values Fig 33: st network G.76 jitter and MOS values Fig 34: st network G.78 jitter and MOS values Fig 3: st network G.73. jitter and MOS values 459

14 Vol. 5, No. 6 June 04 ISSN CIS Journal. All rights reserved. Fig 37: st network IS-64 jitter and MOS values Fig 35: st network PG.79 jitter and MOS values Fig 36: st network GSM FR jitter and MOS values Fig 38: nd network PCM jitter and MOS values 460

15 Vol. 5, No. 6 June 04 ISSN CIS Journal. All rights reserved. Fig 39: nd network G.73. jitter and MOS values Fig 4: nd network G.78 jitter and MOS values Fig 40: nd network G.76 jitter and MOS values Fig 4: nd network G.79 jitter and MOS values 46

16 Vol. 5, No. 6 June 04 ISSN CIS Journal. All rights reserved. Performance analysis focused on voice metrics included voice MOS values, voice end-to-end delay, voice jitter, voice packet delay variation, and voice sent/received packets. The number of calls is calculated and compared for each codec. Results have showed that the MPLS network with CR-LDP TE signal protocol has a noticeable performance advantage compared to the MPLS network with RSVP TE signal protocol. It is five times more than RSVP in terms of number of maintained calls in the st network, this performance difference increases as the network becomes larger as it becomes seven times in the second network. CRLDP has.% less end to end delay of that of RSVP 9 in the st network and by 8.09% in the nd network. For voice jitter, CRLDP is 34.34% of that of RSVP in the st network and 8.3% in the nd network. The voice packet delay variation of CRLDP is.4% of that of RSVP in the st network and 44.88% in the nd one. Fig 43: nd network GSM FR jitter and MOS values The performance of the G.73. codec has the highest number of calls than other codecs but with poor voice quality. The IS-64 codec has higher number of calls % of that of G.73. codec with fair voice quality. Other codecs have less number of calls approximately % of that of G.73. codec but with fair voice quality. These results are for the respective codecs when applied with CRLDP protocol, however a similar performance difference between the codecs is obtained with RSVP protocols. Increasing the number of paths in the nd network increases the performance advantage of the CRLDP over the RSVP. This is due to the RSVP scalability issue when there are a large number of paths passing through a node due to the periodical refreshing of the state for each path. REFERENCES [] O. Akinsipeet al., Comparison of IP, MPLS and MPLS RSVP-TE Networks using OPNET, International Journal of Computer Applications, Vol. 58, No., 0. [] UYLESS BLACK MPLS and Label Switching Network, nd Edition, 00 [3] Reyadh Shaker Naoum, and Mohanand Maswadym, Performance Evaluation for VOIP over IP and MPLS, World of Computer Science and Information Technology Journal (WCSIT) ISSN: -074 Vol., No. 3, 04, 0 Fig 44: nd network IS-64 jitter and MOS values 8. CONCLUSIONS In this paper, the performance of MPLS traffic engineering signaling protocols CRLDP and RSVP have been investigated with seven types of codecs PCM, GSM FR, G.73., G.76, G.78, G.79a, and IS-64. The performances of these codecs have been presented and compared. [4] Mahesh Kr. Porwal, et al., Traffic Analysis of MPLS and Non MPLS Network including MPLS Signaling Protocols and Traffic distribution in OSPF and MPLS, International Conference on Emerging Trends in Engineering and Technology, 008 [5] Ravi Shankar Ramakrishnan and P Vinodkumar, Performance Analysis of Different Codecs in 46

17 Vol. 5, No. 6 June 04 ISSN CIS Journal. All rights reserved. VoIP Using SIP, National Conference on Mobile and Pervasive Computing(CoMPC-008).India [6] Md. Arifur Rahman, et al., Performance Analysis and the Study of the behavior of MPLS Protocols, Proc. of the International Conference on Computer and Communication Engineering 008, Kuala Lumpur, Malaysia [6] Priyanka Luthra and Manju Sharma, Performance Evaluation of Audio Codecs using VoIP Traffic in Wireless LAN using RSVP, International Journal of Computer Applicatio [7] ITU-T Recommendation G.73.: Dual rate speech coder for multimedia communications transmitting at 5.3 and 6.3 Kbit/s; 03/996. [7] Sarmad K. Ibrahim, and Mahmoud M. AL- Quzwini, Performance Evaluation of MPLS TE Signal Protocols with Different Audio Codecs for Voice Application, International Journal of Computer Applications, Vol. 57, No., 0. [8] X. Xiao et al., Traffic Engineering with MPLS in the Internet, Global Center Inc. and Michigan State University, USA, vol. 4, pp. 8-33, Mar [9] A. Ghanwani et al., Traffic Engineering Standards in IP Networks Using MPLS, IEEE Communication Mag. Dec [0] B. Jamoussi, et al., Constraint-Based LSP Setup Using LDP, IETF RFC 3, Janaury [] E. Rosen, et al., Multiprotocol Label Switching Architecture, RFC 303, Janaury [] D. Awduche, et al., Applicability Statement for Extensions to RSVP for LSP-Tunnels (RFC 30), [3] N. F. Mir and A. Chien, Simulation of Voice over MPLS communications Networks, IEEE ICSS conf., CA, 00, pp [4] ITU-T Recommendation G.4: One-way transmission time;05/000. [5] ITU-T Recommendation G.7: Pulse code modulation (pcm) of voice frequencies; /988. [8] ITU-T Recommendation G.76: 40, 3, 4, 6 Kbit/s adaptive differential pulse code modulation (ADPCM); /990. [9] ITU-T Recommendation G.78: Coding of speech at 6 Kbit/s using low-delay code excited linear prediction; 06/0. [0] ITU-T Recommendation G.79: Coding of speech at 8 Kbit/s using conjugate-structure algebraiccode-excited linear prediction (CSACELP); 03/996. [] R. Salami, et al, A toll quality 8 kb/s speech codec for the personal communications system (PCS), IEEE Trans. Veh. Technol.,vol43, no. 3, pp , Aug [] The e-model, a computational model for use in transmission planning. ITU-T recommendation g.07, May000. [3] Jadhav S., et al., Performance Evaluation of Quality of VoIP in WiMAX and UMTS PDCAT 0, pp. 378 [4] M.A. Mohamed, et al., Performance Analysis of VoIP Codecs over WiMAX Networks, IJCSI International Journal of Computer Science Issues, Vol. 9, Issue 6, No 3, November 0, pp [5] K. Salah and A. Alkhoraidly, An Opnet-Based Simulation Approach for Deploying VoIP, International Journal of Network Management, Vol. 6, No. 3, 006, pp