Performance Evaluation and Delay Modelling of VoIP Traffic over Wireless Mesh Network

Transcription

1 International Journal of Computer Application ( ) Volume 1 No.9, May 11 Performance Evaluation and Delay Modelling of VoIP Traffic over 8.11 Wirele Meh Network Amit Chhabra Dept. of CSE SDDIET, Barwala Panchkula, India Dr. Gurpal Singh Dept. of CSE & IT BBSBEC, Fategarh Sahib Punjab, India ABSTRACT Thi paper analyze voice quality in term of R-Factor and MOS in IEEE 8.11 Wirele Meh Network (WMN). Another iue addreed in thi paper i the effect of increaing node on voice tranmiion in wirele meh network. The imulation model developed allow identifying the main reaon for voice quality degradation in Meh network. Reult how that voice quality meaured in term of R-Factor and MOS degrade with increae in number of hop (Wirele Meh Point) in Wirele Meh network. Wirele meh network ha been modeled uing M/D/1 queue and baed on thi analytical delay model, the average one way delay uffered by Voice over IP (VoIP) traffic ha been calculated. Simulation have been conducted to validate the correctne of the analytical model. General Term Wirele Meh Network, VoIP, Delay, Jitter, Packet Lo, Codec Keyword Wirele Meh Network, Voice over IP (VoIP), R-Factor, Mean Opinion Score (MOS), H.33, G INTRODUCTION Wirele Meh Network have recently emerged a an important Reearch area worldwide. A WMN conit of tatic meh router and meh client [1]. Meh router or Meh Point have minimal mobility i.e. they work without any energy contraint and form the meh backbone for meh client. However, the meh client can be either tationary or mobile node. Wirele meh network are an attractive communication paradigm becaue of their low cot and relative eae of deployment []. The capability of being elf-organized, auto-configurable and elf-healing make the wirele meh network a very reliable and robut choice [],[3]. WMN typically conit of many bae tation, ome of which are directly connected to the Internet. The uer connect to one of the bae tation, and the bae tation form a multi-hop wirele network to route traffic between the Internet and the uer. Figure 1 how the baic architecture of WMN where all meh point or router are connected with each other. The WMN backbone provide alternative path between each pair of endpoint, increaing communication reliability and eliminating ingle point of failure within the meh [4]. Meh Point Portal with the gateway/bridge functionalitie for meh network connect it to the Internet a well a with other network. In thi paper, a delay model for WMN ha been developed and ubequent imulation have been conducted to analye the performance of VoIP. The major performance metric including Average Delay, R-Factor and Mean Opinion Score (MOS) for VoIP have been evaluated by varying number of meh point in Wirele meh network. In thi work, M/D/1 queue baed model ha been developed and delay incurred by the VOIP traffic ha been calculated. Thi paper i organized into 8 Section. Section II dicue fundamental of VoIP and Section III contain brief introduction to Voice Quality Metric like Delay, Packet Lo, Jitter, R-Factor and MOS. Section IV decribe the related work done in thi area. Section V decribe the Analytical Delay Modeling of the WMN and ection VI preent the imulation reult. Section VII preent comparion between analytical and imulation reult. Finally, Section VIII preent concluion and remark regarding the future work. Fig.1 Architecture of WMN. VOIP BASICS Voice over Internet Protocol (VoIP) [6] i a technology that tranport voice data packet acro packet witched network uing the Internet Protocol. VoIP involve digitization of voice tream and tranmitting the digital voice a packet over conventional IP-baed packet network like the Internet, Local Area Network (LAN) or wirele LAN (WLAN) [5],[6].Although the quality of VoIP doe not yet match the quality of a circuitwitched telephone network. In WLAN, a VoIP technology i till in the early tage of commercial deployment, it i neceary to examine if VoIP over WLAN can provide a Quality of Service (QoS) comparable to that of the exiting PSTN and cellular network. Alo a the 8.11 baed WMN i gaining popularity, the reearch effort are required to invetigate the Quality of Service of VoIP over uch multi-hop network. Figure how the baic VOIP architecture. 7

2 International Journal of Computer Application ( ) Volume 1 No.9, May 11 In a typical VoIP application, a voice ignal i ampled, digitized, and encoded uing a given algorithm/coder. The encoded data i packetized and tranmitted uing RTP/UDP/IP [7]. At the receiver ide, data i de-packetized and forwarded to a jitter buffer, which moothe out the delay incurred in the network. Finally, the data i decoded and the voice ignal i recontructed. In a VoIP ytem, the total mouth-to-ear delay i compoed of three component: codec delay, jitter delay, and network delay. However, WMN are multi-hop in nature and add additional delay when VoIP traffic i paed over them. Alo the delay added by WMN depend on the number of wirele hop over which the traffic ha travelled. Phone Encoding Packetization Streaming Network Fig.. VOIP Architecture Phone Decoding DePacketization Jitter Buffer 3. VOICE QUALITY METRICS IN WIRELESS MESH NETWORK VoIP ha become a killer application and i gradually being teted over emerging area like Wirele meh network. There are variou challenge for VoIP in WMN. To date ome effort have been carried out to invetigate the performance of VoIP over Wirele Meh Network [11],[1],[13],[14], and there are many challenging iue which till remain to be reolved. There are many factor that affect the quality of voice over Wirele meh network. The quality of a VoIP call [1] i impacted by everal parameter uch a delay, delay jitter, packet lo and o on. Thee parameter are determined by the performance of codec, echo control, buffering and type of network [6],[7]. Delay i the time taken by the VoIP packet to travel from mobile node to gateway in a WMN. Delay can be meaured in either oneway or round-trip delay. A typical VoIP traffic can tolerate up to 15 m end to end delay in a ingle direction before the quality of the call become unacceptable. Jitter i the variation in packet meaured over time. If the delay of tranmiion varie too widely in a VoIP call, the call quality i greatly degraded. The amount of jitter tolerable on the network i affected by the depth of the jitter or playout buffer on the network equipment in the voice path. The more jitter buffer available, the more the network can reduce the effect of jitter. Mot VoIP endpoint device have jitter buffer to compenate for network jitter. The acceptable level of jitter in a network hould be le than m. Packet lo i loing packet along the data path, which everely degrade the voice application. Voice i not tolerant of packet lo. Even 1% packet lo can "ignificantly degrade" a VoIP call. Packet lo can occur becaue of everal reaon: a packet can be dicarded in a router becaue of buffer overflow or becaue the arriving packet i corrupted, the packet can be accidentally mirouted or be lot becaue of a link failure. In thi, reearch work invetigate the performance of VoIP traffic a dependent on number of meh node in a multi-hop 8.11baed WMN. 3.1 R-Factor and Mean Opinion Score The mot reliable method for evaluating the voice quality i the E- model[8][9] defined by the International Telecommunication Union-Telecommunication (ITU-T). An ETSI work group developed the E-Model alo known a European telecommunication Standard Intitute (ETSI) Computation Model. The E-Model define an analytical model for prediction of VoIP quality baed on variou network impairment parameter uch a packet lo, delay, codec type, noie etc. E-Model give a ingle, overall meaure of converational voice quality called the rating factor (R-Factor) [9].R Factor include the effect of mouth-to ear delay and loe in packet-witched network and i given by the following relation R R I Where R i voice quality without ditortion. Uually it i equal to 1, I impairment caued due to echo and packet lo, I d correpond to impairment level caued by delay and delay jitter, I e repreent the impairment caued by encoding. A i expectation factor that a uer can tolerate. An R-factor between 5 and 6, 6 and 7, 7 and 8, 8 and 9, or 9 and 1 indicate poor, low, medium, high, or bet voice quality, repectively. Different approache have been ued to tranlate thee rating into an overall ingle meaure from which peech quality can be judged. A popular approach i to calculate the arithmetic mean of core known a a mean opinion core (MOS). A MOS range from a minimum of 1 to a maximum of 5. A MOS of 4. or higher i conidered toll quality, and a lower limit of 3. i uggeted for uable telephony. R and MOS are related a follow MOS 1.35 R 7 *1 I d 6 I e R( R A 6)(1 R) A buffer in the receiving device alway compenate for jitter (delay variation). If the delay variation exceed the ize of the jitter buffer, there will be buffer overrun at the receiving end, cauing packet lo to occur. The VoIP QoS in WMN partly alo depend on the type of voice codec ued [6],[7]. The primary function of a voice codec i to perform analog/digital voice ignal converion and digital compreion. There are many codec available for digitizing peech. Table 1 give ome of the characteritic of a few tandard codec. Among three commonly ued codec [8] in Internet telephony are G.711, G.73.1 and G.79. Thee codec differ in their coding rate (bp), frame rate (frame/), algorithmic latency 8

3 International Journal of Computer Application ( ) Volume 1 No.9, May 11 that influence the peech quality or Mean Opinion Source (MOS) in a VoIP network. Payload ize for each codec depend on the codec peed or data rate. The G.711 ha peed of 64 Kbp and if each peech packet ize i of m, then the payload ize for G.711 will be of 16 byte. Payload Size( byte) peech packetize( m)]/[8( bit/ byte)*1 ( m / ec)] [ codec peed ( bit/ ec)* Payload ize of 16 byte for G.711 codec mean that the codec produce 16 byte chunk of VoIP traffic every m interval. The G.711 codec give the bet voice quality, ince it perform no compreion, introduce the leat delay, and i le enitive than other codec to packet lo. Other codec, like G.79 and the G.73 conume le bandwidth by compreing the ignal. In thi reearch work, G.711 codec ha been ued becaue of it good voice quality and leat delay. Table 1. Different Attribute of Codec Codec Data Rate Speech Packet (kbp) Size (m) G m G m G.73m m G.73a m 3. Call Signaling Protocol There are everal VoIP call ignaling protocol [7] like H.33, SIP, MGCP, and megaco/h.48, which can be ued for etablihment and management of VoIP traffic in WMN. The ignaling refer to etablihing a connection or an aociation between a pair of participant in VoIP call. Signaling protocol mut alo perform addre tranlation, bandwidth management, authorization, and in ome cae make routing deciion. H.33 and SIP are peer-to-peer control-ignaling protocol, while MGCP and Megaco are mater lave control-ignaling protocol.h.33 and Megaco are deigned to accommodate video conferencing a well a baic telephony, but they are till baed on a connectionoriented paradigm, depite their ue for packet communication ytem. The Voice-over-IP Activity Group of the International Multimedia telecommunication Conortium (IMTC) recommend H.33, which had been developed for multimedia communication over packet data network. Thee packet network might include LAN or WAN. H.33 wa originally developed for videoconferencing over a packet baed network, but wa quickly adopted for Voice over IP.The main function of the protocol i to perform call control and management on an IP network. In thi reearch work we have ued H.33 a a ignaling protocol for VoIP in wirele meh network. 4. RELATED WORK VoIP over WMN ha witneed ignificant reearch activity in the recent pat. Literature review reveal coniderable reearch work conducted in thi area. Akyildiz et. al. in [1] preent an excellent urvey on wirele meh network. Introductory tudie on Wirele Meh Network including it type, application, implementation, problem and perpective have been preented in [],[3],[4]. Work pecifically focuing on VoIP ytem including the choice of codec and call ignaling protocol have been preented in [5],[6],[7]. Author in [8],[9] preent a voice quality meaurement tool baed on ITU-T E-Model in which quality metric like R-Factor and Mean Opinion Score have been defined. Analye conducted in [1] reveal that even ingle additional VoIP call in the cell could degrade the quality of all the ongoing VoIP call. In [11],[1],[13], the author propoe variou voice quality meaure uch a delay, jitter, lo rate and R-Score and alo propoe an Interference capacity model for a wirele meh network. Several performance optimization cheme like packet Aggregation, Header Compreion and Label Baed Forwarding have been propoed to enhance the performance of voice in wirele meh network in [14]. Author in [15] have performed analyi of Delay and throughput uing M/D/1 queuing theory by modeling the gateway node a independent M/D/1 queue tation, and derived cloed-form olution for the bottleneck delay and throughput with linear and grid topologie of wirele meh network. In [16] author have analyzed the throughput, packet lo, and delay and alo conider the impact of interference on modeling the meh hop node a M\M\1. Sunny et. al. in [17] have olved the problem of modeling the average delay in an IEEE 8.11 DCF wirele meh network with a ingle root node under light aggregate traffic in Wirele Meh Network uing M\M\1 Queue. In [18], author have given variou delay component to be aggregated to calculate the overall end to end delay for voice traffic. Literature review reveal that WMN ha emerged a a ueful technology for variou real time application. Thi provide motivation for conducting invetigation into VoIP performance of WMN. In thi paper, the performance of VoIP over WMN i dependent on number of node or meh router modeled a linear meh linear topology. In WMN, the backbone i mainly fixed a the route followed by the traffic i fixed or tatic. So, under thee aumption, the WMN can be conidered a a linear multi-hop network that could be modeled a M/D/1 Queue [19]. 5. MODELLING THE WMN Wirele Meh Network can be modeled uing M/D/1 Queue [19]. It i conidered that the WMN conit of n backbone meh point. The Sytem ha been aumed to conit of only one erver i.e. the gateway through which the traffic i forwarded onto the Internet. The arrival proce ha been aumed to be Poion ditributed where a the ervice time i determinitic. Under uch aumption the ytem can be modeled by uing M/D/1 queue. Thi model ha been ued to find the average delay taken by the VoIP packet from ource meh point to detination meh point. Each meh router i a tation in the equivalent queuing network repreentation. The average delay i the expectation of packet delay over all packet. Thi work ha been retricted to intra meh communication cenario i.e. cae where meh client communicate with each other uing the meh router backbone. 9

4 Queuing Delay International Journal of Computer Application ( ) Volume 1 No.9, May 11 With Aumption of M/D/1 Queuing theory, the voice packet arrive according to a Poion proce with mean arrival rate λ, and packet are proceed in a determinitic manner with ervice time. Suppoe that packet are proceed in the order they arrive and that i i the ervice time of the ith arrival. We aume that each random variable (1, ) i identically ditributed and identical for all voice packet. Let ] 1/ E [ ] AverageServiceTime Second moment of ervicetime According to Pollaczek-Khinchin (P-K) formula: For a General Ditribution, W ]/ (*(1 (1) Where, W i the expected wait time in queue and ρ i the utilization factor. / E [ ] Applying little formula to W, we get the expected number of packet in the queue a NQ and the expected number in the ytem a N: NQ NQ N W ]/ (*(1 ]/(*(1 When ervice time i identical for all voice packet i.e. if M/D/1 theory i applied, E [ ] Now, the total average packet Queuing delay = Service time +Waiting time in Queue. Therefore, total time taken by VoIP packet in queue and ervice in a wirele network can be repreented a: Q Q d d ] /(*(1 1/ ]/ (*(1 () where, Q d i the average queuing delay taken by VoIP packet to reach from ource client to detination via a meh router in a network. Now for a Wirele Meh Network having n meh router, the Eq.() can be changed to Eq. (3) by changing λ by nλ. In cae of n meh router, the net arrival rate will become nλ. So, the average queuing delay incurred by VoIP packet from ource meh client to detination (Internet gateway in thi cae) in a Wirele Meh Network having n meh router [15] can be expreed a equation: Q nd n /(*(1 n (3) 5.1 Packet Aggregation A, we are uing G.711 encoder, the default packetization interval for 16 byte payload i m i.e. voice packet are tranmitted at the rate of λ = 5 packet per econd. Taking ervice time equal to.1 ec. By increaing the number of meh point i.e. value of n in Eq. (3), we get negative value for queuing delay for uing more than 1 meh node. It i due to high value of arrival rate λ of voice packet. So, we can decreae thi arrival rate by aggregating the packet to a uper packet to tranmit them at higher packetization interval. Packet aggregation [14] multiplexe VoIP packet from different connection into one large packet. It aggregate variou data packet into a frame to reduce protocol header overhead. If we conider a VoIP packet of payload 16 byte emitted once in m, then we can merge 5 packet into one giving 8 byte emitted once in 1 m. With a 1 m ample period correponding to a rate of 1 packet per econd, the payload ize i (64*1)/ (8 *1) = 8 byte. So, more the value of ample period, leer will be the arrival rate. Figure 3 how the increae of Analytical Queuing Delay for voice packet with increae in number of meh point Queuing Delay with Packet Aggregation Fig.3. Analytical Average Queuing Delay with varying meh router Total delay in VoIP application i compried of packetization delay, queuing delay, propagation delay, encoding/decoding delay, tranmiion delay and jitter buffer delay. The end to end delay D for VoIP traffic in wirele meh network in one direction from ender to receiver can be repreented by the expreion [18]: D P Q J ( PR T ) (4) d nd d i i For eachmehnodei 1 i n where P d i the packetization delay introduced by encoder G.711 at the ource. A, we are uing G.711 encoder, the default packetization delay for 8 byte payload will be 1 m. Q nd i the Queuing Delay for meh network having n meh node which can be calculated from Eq. (3). J d i the Jitter delay incurred by the jitter buffer. In thi cae, we are taking no jitter buffer delay. PR i and T i are the propagation and tranmiion delay incurred at each meh node i in the path from the ender to the receiver. Finally, we can aggregate all the delay to find the total average one way delay of VoIP traffic from ource meh client to detination. 1

5 MOS Average Delay() International Journal of Computer Application ( ) Volume 1 No.9, May Analytical Average Delay Fig.4. Analytical Average one way Delay from VoIP ource to detination with packetization delay of 1 m Figure 4 repreent increae in Analytical Average one way Delay of voice in Wirele Meh Network computed uing Eq. (4). Voice i paed from ource to detination with varying meh node with packetization delay of 1 m. 6. SIMULATION AND RESULTS To verify the correctne of the model, a WMN cenario ha been imulated uing Qualnet imulator. Qualnet[] i a dicrete event imulation ytem having a robut graphical uer interface which cover all apect of the imulation from cenario creation, topology etup, integration of cutom protocol, through real-time execution of network model from within the GUI, animation, to pot-imulation tatitical analyi. Qualnet ha been ued to imulate high-fidelity model of wirele network with a many a 5, mobile node. Six cenario of wirele meh network have been conidered in the imulation. The number of meh node or meh point ha been varied in the network to ee the influence of the network ize on the performance of VoIP traffic in WMN. In the firt cenario, the ource and detination ha been eparated by only one meh acce point or hop and then in further cenario number of meh acce point or hop have been increaed to 3, 5, 7, 9 and 11 hop repectively. In each cae 5 imulation intance have been run with different eed. After that average i taken for different cae of a cenario to produce the reult. Figure 5 how a Scenario Deigner Window in Qualnet having five Meh Router or Meh Point numbered 1,, 3, 4 and 5 repectively. All node are interlinked with each other through wirele ubnet and VoIP Traffic i paed between mobile client numbered with 6 and 7 with interval of m. Fig.5. Scenario Deigner Window of Qualnet Figure 6 how the graph which repreent the degradation of voice quality in term of Mean Opinion Score (MOS) with increae in number of hop (Meh Router) in wirele meh network Fig.6. MOS v Mean Opinion Score A we can ee in Fig. 6, up to 7 meh router voice quality in term of MOS degrade in almot linear manner but voice quality greatly reduce when more than 7 meh router have been introduced in a wirele meh network. From the graph, it can be een that voice quality decreae in an exponential manner with increae in number of meh router. Fig. 7 how the degradation of R-Factor with increae in number of hop. 11

6 Average Delay R-Factor Average Delay International Journal of Computer Application ( ) Volume 1 No.9, May Fig.7. R-Factor v. R-Factor Fig. 8 how the increae in Average one way Delay with increae in number of meh router in a wirele meh network when VoIP traffic i tranmitted with packetization interval of m Average Delay Fig. 8: Average one way delay with varying meh router with packetization interval of m 7. VALIDATION Finally the comparion between the theoretical and imulation reult ha been preented. For the analytical computation of mean delay from Eq (4), the arrival rate λ i taken a 1 pp. The value of n i.e. the number of hop ha been varied. Validation tet were run multiple time to determine the accuracy of the analyi. Extenive imulation have been conducted to verify the analytical reult.fig. 9 how the comparion between Analytical and Simulation Reult obtained for average delay incurred by VoIP packet from ource to detination with varying number of hop (upto 9) for packetization interval of 1 m. The imulation reult how that the analyi reult almot matche the imulation reult i.e. both imulation and theoretical delay increae in an exponential manner a the number of meh hop increae. Thu the M\D\1 model uccefully predict the average delay in wirele meh network Analytical Simulation Fig. 9: Comparion of Simulation and Analytical Average Delay with packetization interval of 1 m. 8. CONCLUSIONS & FUTURE WORK Thi paper preent analytical model and imulation reult of VoIP traffic over 8.11 baed WMN in term of parameter uch a average delay, R-Factor and Mean Opinion Score. Etimation of Average delay in VoIP over WMN with varying number of node ha been carried out uing theoretical analyi and compared with imulation reult. The developed model can be added to the exiting R-factor for predicting R core when WMN i preent in the network. Alo the upper limit on number of hop in WMN for carrying the VoIP application can be etimated. In the future work the cenario involving multi-channel multi-radio WMN can be explored. Alo performance of voice in a WMN can be meaured and compared for variou proactive and reactive routing protocol. Subequently, the framework could be extended to model the cenario for non-poion arrival and alo for different ervice ditribution. 9. REFERENCES [1] I.F.Akyildiz,.Wang, A Survey on Wirele Meh Network, IEEE Comm Mag. 5, pp.3-3, vol 43(9). [] R. Bruno, M. Conti, and E. Gregori, Meh Network: Commodity Multihop AdHoc Network, IEEE Comm. Mag., vol. 37, pp , March 5. [3] Tzu-Jane and Ju-Wei Chen, IEEE 8.11 MAC Protocol over Wirele Meh Network: Problem and Perpective, March 5, [4] Mahmud, S.A., Khan, S., Khan, SH., Al-Rawehidy, H., A comparion of MANET and WMN: commercial feaibility of community wirele network and MANET, Proceeding of the 1t international conference on Acce network, 6, pp: 36:54, 6. [5] Schulzrinne, H. and Roenberg, J., "Internet telephony: architecture and protocol - an IETF perpective", Computer Network and ISDN Sytem, Vol. 31, pp , Feb [6] Bur Goode, Senior Member, Voice over Internet Protocol (VoIP), Proceeding of the IEEE, Vol. 9. [7] Trad, F. Munir, H. Afifi, Capacity Evaluation of VoIP in IEEE 8.11e WLAN Environment, IEEE CCNC 6 Proceeding, September. 1

7 International Journal of Computer Application ( ) Volume 1 No.9, May 11 [8] Leandro Carvalho, Edjair Mota, Regeane Aguiar Anderon Barreto, An E-Model Implementation for Speech Quality Evaluation in VoIP Sytem, IEEE Sympoium Computer and Communication (ISCC 5), pp , IEEE, 5. [9] L. Ding and R.A. Goubran, Speech quality prediction in VoIP uing the extended E- model, IEEE GLOBECOM 3, pp [1] Sachin Garg, Martin Kappe, Can I add a VoIP call? Proc. of IEEE ICC 3, vol., May 3. [11] udong Wang, Abhihek Patil and Weilin Wang., Voip over wirele meh network: challenge and approache, WICON 6: Proceeding of the nd annual international workhop on Wirele internet, page 6, New York, USA, 6. [1] H.Y. Wei, K. Kim, A. Kahyap and S. Ganguly, On Admiion of VoIP Call over Wirele Meh Network, Proceeding of IEEE ICC, 6. [13] K. Kyungtae and H. Sangjin, "VoMESH: voice over wirele meh network, Proc of IEEE Wirele Communication and Networking Conference, LaVega, USA, 6, pp [14] S.Ganguly et al., Performance Optimization for Deploying VoIP Service in Meh Network, IEEE Journal on Selected Area in Communication (JSAC), 6 pp [15]. Wu, J. Liu, G. Chen, Analyi of Bottleneck Delay and Throughput in Wirele Meh Network, IEEE International Conference on Mobile Ad-hoc and Senor Sytem, 6. [16] Yunxia Feng, ingfashen, Zhigang Gao, Guojun Dai, Queuing Baed Traffic Model for Wirele Meh Network, Proc of 15 th International Conference on Parallel and Ditributed Sytem, 9. [17] Albert Sunny, Joy Kuri, Saurabh Aggarwal, Delay modeling for a ingle-hop wirele meh network under light aggregate raffic, online verion available at [18] M. Karam and F. Tobagi, Analyi of the Delay and Delay Jitter of Voice Traffic in the Internet, Computer Network Magazine, vol 4, no 6, December, pp [19] Queuing web link available: - wiki/m/d/1_model [] Scalable Network Technologie, Inc. QualNet Uer Manual, verion 3.6.,