In this research, we propose an architectural solution to implement the voice
over IP (VoIP) service in campus environment network. Voice over IP (VoIP)
technology has become a discussion issue for this time being. Today, the
deployment of this technology on an organization truly can give a great
financial benefit over traditional telephony. Therefore, this study is to
analyze the VoIP Codec selection and investigate the Mean Opinion Score (MOS)
performance areas evolved with the quality of service delivered by soft phone
and IP phone. This study focuses on quality of voice prediction such as i)
accuracy of MOS between automated system and human perception and ii) different
types of codec performance measurement via human perception using MOS
technique. In this study, network management system (NMS) is used to monitor
and capture the performance of VoIP in campus environment. In addition, the
most apparent of implementing soft phone and IP phone in campus environment is
to define the best codec selection that can be used in operational environment.
Based on the finding result, the MOS measurement through automated and manual
system is able to predict and evaluate VoIP performance. In addition, based on
manual MOS measurement, VoIP conversations over LAN contribute more reliability
and availability performance compare to WAN.Comment: 14 pages, exposed on 5th International Conference "Actualities and
  Perspectives on Hardware and Software" - APHS2009, Timisoara, Romani

Ismail, Mohd Nazri

English

arXiv

In this research, we propose an architectural solution to implement the voice over IP (VoIP) service in campus environment network. Voice over IP (VoIP) technology has become a discussion issue for this time being. Today, the deployment of this technology on an organization truly can give a great financial benefit over traditional telephony. Therefore, this study is to analyze the VoIP Codec selection and investigate the Mean Opinion Score (MOS) performance areas evolved with the quality of service delivered by soft phone and IP phone. This study focuses on quality of voice prediction such as i) accuracy of MOS between automated system and human perception and ii) different types of codec performance measurement via human perception using MOS technique. In this study, network management system (NMS) is used to monitor and capture the performance of VoIP in campus environment. In addition, the most apparent of implementing soft phone and IP phone in campus environment is to define the best codec selection that can be used in operational environment. Based on the finding result, the MOS measurement through automated and manual system is able to predict and evaluate VoIP performance. In addition, based on manual MOS measurement, VoIP conversations over LAN contribute more reliability and availability performance compare to WAN

Mohd Nazri Ismail

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     Anale. Seria Informatică. Vol. VII fasc. 1 – 2009 Annals. Computer Science Series. 7th Tome 1st Fasc. – 2009    263      A An na al ly yz zi in ng g   o of f   M MO OS S   a an nd d   C Co od de ec c   S Se el le ec ct ti io on n   f fo or r   V Vo oi ic ce e   o ov ve er r   I IP P   T Te ec ch hn no ol lo og gy y     M Mo oh hd d   N Na az zr ri i   I Is sm ma ai il l   Department of MIIT,  University of Kuala Lumpur (UniKL), Malaysia mnazrii@miit.unikl.edu.my   ABSTRACT. In this research, we propose an architectural solution to implement  the  voice  over  IP  (VoIP)  service  in  campus  environment network. Voice over IP (VoIP) technology has become a discussion issue for  this  time  being.  Today,  the  deployment  of  this  technology  on  an organization  truly  can  give  a  great  financial  benefit  over  traditional telephony. Therefore, this study is to analyze the VoIP Codec selection and  investigate  the  Mean  Opinion  Score  (MOS)  performance  areas evolved with the quality of service delivered by soft phone and IP phone. This study focuses on quality of voice prediction such as i) accuracy of MOS between automated system and human perception and ii) different types of codec performance measurement via human perception using MOS technique. In this study, network management system (NMS) is used  to  monitor  and  capture  the  performance  of  VoIP  in  campus environment.  In addition, the most apparent of implementing soft phone and IP phone in campus environment is to define the best codec selection that can be used in operational environment.  Based on the finding result, the MOS measurement through automated and manual system is able to predict and evaluate VoIP performance. In addition, based  on  manual MOS  measurement,  VoIP  conversations  over  LAN  contribute  more reliability and availability performance compare to WAN.   KEYWORDS:  VoIP,  MOS,  NMS,  Packet  loss,  Codec,  Campus Environment   Introduction  As with most new technologies, Voice over Internet Protocol (VoIP) brings new challenges along with the benefits. The main challenge is VoIP’s extreme      Anale. Seria Informatică. Vol. VII fasc. 1 – 2009 Annals. Computer Science Series. 7th Tome 1st Fasc. – 2009    264 sensitivity to delay and packet loss compared with other network applications such as web and e-mail services. A basic understanding of VoIP traffic and of the quality metrics provided by VoIP monitoring tools will help to keep VoIP network running smoothly.  Existing hardware video phones have integrated video camera and microphone and can be used independently of any other equipment [VCC06].  Traditionally, set top boxes are video broadcast receivers and are equipped with video/audio decoder (MPEG2) and usually based on H.263 codec for video and G.711, G.723 or G.729 codecs for voice [VCC06]. The  actual  multimedia  content  (voice  and  video)  is  usually  transmitted  by means of the Real-time Transport Protocol (RTP) [S+03]. The objectives of this study  as  follows:  i)  design  and  implement  VoIP  technology  for  ease interactions and communications between UniKL branches; ii) provide toll free communication  platform  using  real  time  voice  communication  in  campus environment; iii) implement and test the accuracy of VoIP management system with manual system using MOS technique; and iv) use and test different types of codec for VoIP conversation. This study focuses on quality of voice and video performance using MOS  technique  such  as  i)  automated  MOS  measurement;  ii)  physical (manual) MOS by human perception measurement; and iii) different codec selection measurement.  Today, many people are talked and communicated face to face using VoIP Phone system. When VoIP is implemented using the public Internet, users may experience quality degradations due to dynamic delays and losses in the LAN and Internet. Packets may be lost, either in isolation or in batches, and may experience sudden delay increases [SW07]. Figure 1 and Figure 2 show USB Phone system and soft phone application will  use  in  real  network  environment  for  the  experimental.    In  addition, Figure 3 shows the sample of VoIP network architecture.                                       Fig.  1: USB Video Phone System                     Fig.  2: Soft Phone Application      Anale. Seria Informatică. Vol. VII fasc. 1 – 2009 Annals. Computer Science Series. 7th Tome 1st Fasc. – 2009    265    Fig.  3: Sample of VoIP Architecture   1. Related Works  Recently, VoIP (Voice over IP) [Col02] is rapidly growing and becoming a mainstream telecommunication service, not only because of the lower cost compared with traditional PSTN (Public Switched Telephone Network), but also its convergence technologies of data and voice communication. VoIP applications like Skype [BS06] have also achieved great success. However, due to the complexity of the  Internet, it is unpractical to calculate VoIP performance metrics only through the mathematical modeling, as what was done  in  the  telephone  networks,  so  the  performance  evaluation  of  VoIP requires actual measurement activities.  There  have  been  numerous  studies  on  VoIP  measurement.  A. Markopoulou [MTK02] measured loss and delay characteristics of American backbone  networks,  and  analyzed  how  these  characteristics  impact  VoIP quality.  For  examples,  most  work  focused  on  monitoring  and  analyzing performance of actual applications, like MSN and Skype [C+06a] [C+06b]. Skype using more to video codec like H.263 and H.261. Skype also using the audio codec likes G.711, G.729, G.723, G.728 and GSM. In contrast to other works in the literature, we implemented VoIP network environment to measure the performance of VoIP using MOS technique at campus environment.      Anale. Seria Informatică. Vol. VII fasc. 1 – 2009 Annals. Computer Science Series. 7th Tome 1st Fasc. – 2009    266  2. Methodology  Figure  4  shows  the  overall  framework  of  the  VoIP  service  in  campus environment.  There  are  five  phases  development  process  such  as:  i) planning and research; ii) development; iii) implementation; iv) testing and v) documentations.    Fig.  4: Framework of VoIP Service in Campus Environment Development  Figure  5  shows  the  overall  framework  of  the  VoIP  performance analysis using MOS technique. In the experiment, the performance analysis will  focus  on  automated  MOS  system,  manual  MOS  measurement  and different  codec  selection  over  LAN  and  WAN.  Network  management system such as VQ manager is used to analyze VoIP service via automated MOS measurement in campus environment. Manual MOS measurement is based on human perception.          Anale. Seria Informatică. Vol. VII fasc. 1 – 2009 Annals. Computer Science Series. 7th Tome 1st Fasc. – 2009    267   Fig.  5: Framework of VoIP Performance Analysis using MOS Technique   3. Proposed VoIP Network Architecture in Campus Environment  We  have  setup  a  real  network  environment  to  analyze  and  measure implementation of VoIP service at University of Kuala Lumpur (UniKL) in Malaysia.  This  study  posits  several  research  questions:  i)  what  is  the performance level of the VoIP over LAN and WAN using MOS technique; and  ii)  Is  the  analysis  for  evaluating  and  measuring  VoIP  performance effective using MOS technique?  Figure  6  shows  the  VoIP  architecture  in  real  network  campus environment.  VoIP  quality  can  be  monitored  periodically  through  the measurement  using  VQnet  (VoIP)  management  tools  to  gather  quality variation information, avoiding the ignorance of unacceptable VoIP quality caused  by  the  network  failure  or  bandwidth  bottleneck  (see  Figure  7). Figure  8  shows  MOS  technique  implementation  in  campus  environment using IP phone and multiple soft phones performance measurement. Several soft phones have used to evaluate and measurement of VoIP performance using MOS technique such as i) Wengo Phone; ii) 3CX; iii) PhoneLite; iv) Kapanga; v) SJPhone and others [BHW07].        Anale. Seria Informatică. Vol. VII fasc. 1 – 2009 Annals. Computer Science Series. 7th Tome 1st Fasc. – 2009    268   Fig.  6: Development of V2oIP Architecture in Real Network    Fig.   7:  Installation of Network Monitoring System for VoIP        Anale. Seria Informatică. Vol. VII fasc. 1 – 2009 Annals. Computer Science Series. 7th Tome 1st Fasc. – 2009    269   Fig.   8: Implementation of MOS Technique   4. Experimental and Analysis Results  This section analyses, measures and compares VoIP performance in campus environment. The MOS is expressed as a single number in the range 1 to 5, where 1 is lowest perceived quality, and 5 is the highest perceived quality. The  MOS  is  generated  by  averaging  the  results  of  a  set  of  standard, subjective tests where a number of listeners rate the heard audio quality of test  sentences  read  aloud  by  both  male  and  female  speakers  over  the communications medium being tested. In the VoIP world, voice quality is measured by a ‘Mean Opinion Score’, which is a number between 1 and 5 used  to  quantitatively  express  the  subjective  quality  of  speech  in communications systems, especially digital networks that carry VoIP traffic. Anything above a 4.0 is considered toll grade (see Figure 9 and Table 1).  Table 1: MOS Rating Measurement  MOS  QUALITY  IMPAIRMENT 5  Excellent  Imperceptible 4  Good  Perceptible but not annoying 3  Fair  Slightly annoying 2  Poor  Annoying 1  Bad  Very Annoying       Anale. Seria Informatică. Vol. VII fasc. 1 – 2009 Annals. Computer Science Series. 7th Tome 1st Fasc. – 2009    270                                 Fig.  9: MOS Technique Rating  A.  Comparison of MOS Automated System and Manual Human Perception  Figure 10 shows the sample of MOS automated system that generate by ‘Advenet Manage Engine VQ Manager’ VoIP management tool. Soft and IP phones  are  configured  using  Codec  G.711.  We  used  several  soft  phones such as SJPhone and Wengo Phone to measure the accuracy of automated system and human perception.        Figure 10: MOS Generate by Automated System  Based on the results, Figure 11 and Figure 12 show the MOS value between automated system and manual system (physical).  Both automated and  manual  system  measurement  behave  approximately  the  same  graph pattern during the evaluation of VoIP performance. We conclude that base      Anale. Seria Informatică. Vol. VII fasc. 1 – 2009 Annals. Computer Science Series. 7th Tome 1st Fasc. – 2009    271 on  our  findings,  the  MOS  measurement  through  automated  and  manual system is able to predict and evaluate VoIP performance.     Figure 11: Comparison of MOS Automated and Manual System (Physical)  via Wengo Phone Application    Figure 12: Comparison of MOS Automated and Manual System (Physical)  via SJPhone Application  Different  Codec  Selection  Performance  Measurement  by  MOS Technique:  Codecs  generally  provide  a  compression  capability  to  save network bandwidth. Some Codecs also support silence suppression, where silence is not encoded or transmitted. . The most common voice CODECs includes  G.711,  G.723,  G.726,  G.728,  and  G.729.  Table  2  shows  VoIP Codec protocols and algorithms. G.711 has two versions called U-law (US,      Anale. Seria Informatică. Vol. VII fasc. 1 – 2009 Annals. Computer Science Series. 7th Tome 1st Fasc. – 2009    272 Japan) and A-law (Europe). U-law is indigenous to the T1 standard used in North America and Japan. In our experiments, we used four different types of Codec such as i) G.711 U-law; ii) G.711 A-law; iii) ILBC; and iv) speex. We  selected  five  users  to  measure  VoIP  performance  through  MOS technique. The users need to make conversation between them.  Table 2: VoIP Codec Protocols and Algorithms  Codec  Algorithm  Kbps G.711  PCM (Pulse Code    Modulation)  64 G.722  SBADPCM (Sub-Band Adaptive Differential Pulse Code Modulation) 48, 56 and 64 G.723  Multi-rate Coder  5.3 and 6.4 G.726  ADPCM (Adaptive Differential Pulse Code Modulation) 16, 24, 32, and 40 G.727  Variable-Rate ADPCM  16-40 G.728  LD-CELP (Low-Delay Code Excited Linear Prediction) 16 G.729  CS-ACELP (Conjugate Structure Algebraic-Code Excited Linear Prediction) 8 ILBC  Internet Low Bitrate Codec  13.33 and 15.20 Speex  CELP (Code Excited Linear Prediction)  2.15-44.2 GSM  RPE-LTP (Regular Pulse Excitation Long-Term Prediction) 13  We  conclude  that  base  on  our  findings,  the  MOS  measurement through LAN and WAN using different types of Codec during conversation VoIP  (see  Table  3).  Codec  G.711  contributes  high  reliability  VoIP conversation compare to iLBC and Speex. Both medium (LAN and WAN) contribute different performance and characteristics. If the PC (install soft phone) load is very high, it can achieve higher CPU usage and make the throughput  low  that  can  cause  inconsistent  conversation  between  two parties.  Therefore, it is important to understand how delay, jitter and CPU usage can affect speech quality through soft phone. In addition, poor link quality network between two parties can also increase the delay and jitter for VoIP service. The efficiency of VoIP service in campus environment can quickly deteriorates when the jitter and delay increases in WAN (see Figure 13). The results of this study show that voice service is able to contribute and achieve reliability and scalability in LAN (see Figure 13). Therefore, proper  plan  can  mitigate  most  of  the  negative  effects  that  are  related  to V2oIP characteristics.       Anale. Seria Informatică. Vol. VII fasc. 1 – 2009 Annals. Computer Science Series. 7th Tome 1st Fasc. – 2009    273 Table 3: Manual MOS Codec Measurement  USERS  CODECS  MOS in LAN  MOS in WAN G711 aLaw  4  3 G711 uLaw  4  3 iLBC  3  2 User 1 Speex  3  2 G711 aLaw  4  3 G711 uLaw  3  2 iLBC  2  1 User 2 Speex  3  2 G711 aLaw  3  2 G711 uLaw  4  3 iLBC  2  1 User 3 Speex  3  2 G711 aLaw  3  2 G711 uLaw  4  3 iLBC  3  2 User 4 Speex  3  2 G711 aLaw  4  3 G711 uLaw  3  2 iLBC  2  1 User 5 Speex  2  1   Conclusion  Convergence  of  video,  voice  and  data  over  IP  with  current  network infrastructure can contribute many revenues to campus users.  This paper discussed which  •  MOS system (automated and manual measurement) could produce good performance evaluation;  •  different types of VoIP Codec selection during conversation.  We conclude that base on our findings, WAN can contribute higher delay, packet loss and CPU usage compare to LAN in campus environment. While,  automated  management  and  manual  system  measurement  will generate  approximately  similar  results.  If  VoIP  want  to  implement  in campus environment, it is recommended to enable QoS function in order that to achieve a good quality conversation between two parties over LAN and WAN.       Anale. Seria Informatică. Vol. VII fasc. 1 – 2009 Annals. Computer Science Series. 7th Tome 1st Fasc. – 2009    274   Figure 13: Comparison of VoIP over LAN and WAN Measurement   In future work, it should consider implementing techniques to improve quality  of  VoIP  over  WAN.  Therefore,  efficient  and  effective  QoS provisioning techniques are important to achieve a robust service. There are several  techniques  should  be  studied  and  analyzed  that  can  be  used  to increase performance of VoIP in campus environment as follows:  •  Dejitter buffer;  •  Type of Service (ToS);  •  Weighted Fair Queuing (WFQ);  •  Random Early Detection (RED).  Implementing  quality  of  service  mechanisms  on  peak  hours  is  a method  to  improve  V2oIP  network  service  performance  in  campus environment.    References  [BS06]   S. A. Baset, H. Schulzrinne - An analysis of the Skype peer-to-peer internet telephony protocol, In Proc. of IEEE INFOCOM, Barcelona, Spain, Apr. 2006       Anale. Seria Informatică. Vol. VII fasc. 1 – 2009 Annals. Computer Science Series. 7th Tome 1st Fasc. – 2009    275 [BHW07]   Batu Sat, Zixia Huang, Benjamin W. Wah, - The Design of a Multi-Party VoIP Conferencing System over the Internet, Ninth IEEE International Symposium on Multimedia, IEEE Computer Society  Washington, DC, USA, 2007  [Col02]   Daniel  Collins  -  Carrier  Grade  Voice  over  IP,  2nd  Ed., McGraw-Hill,September 2002  [C+06a]   W. Chiang, et al. - A performance study of voip applications: Msn vs. skype, In Proc. of MULTICOMM, 2006  [C+06b]   K. Chen, C. Huang, P. Huang, C. Lei - Quantifying Skype user satisfaction, In Proc. of ACM SIGCOMM, Pisa, Italy, 2006  [MTK02]   A. Markopoulou, F. Tobagi, M. Karam - Assessment of VoIP quality over Internet backbones, in Proc. of IEEE INFOCOM, New York, NY, Jun. 2002  [SW06]   B. Sat, B. W. Wah - Analysis and evaluation of the Skype and Google-Talk  VoIP  systems,  In  Proc.  IEEE  Int’l  Conf.  on Multimedia and Expo, July 2006  [SW07]   B. Sat, B. W. Wah - Evaluation of conversational voice quality of the Skype, Google-Talk, Windows Live, and Yahoo Messenger VoIP systems,  In  IEEE  Int’l Workshop on Multimedia Signal Processing, Oct. 2007  [S+03]   H. Schulzrinne, S. Casner, V. Jacobson, R. Frederick - RTP: A Transport Protocol for Real-Time Applications, RFC 3550, June 2003  [VCC06]   Antonio Vilei, Gabriella Convertino, Fabrizio Crudo - A New UPnP  Architecture  for  Distributed  Video  Voice  over  IP, Proceedings of the 5th international conference on Mobile and ubiquitous multimedia, Stanford, California, 2006             Anale. Seria Informatică. Vol. VII fasc. 1 – 2009 Annals. Computer Science Series. 7th Tome 1st Fasc. – 2009    276   

Analyzing of MOS and Codec Selection for Voice over IP Technology

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