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Fourth International Conference onDigital Telecommunications, 2009. ICDT '09. 20-25 July 2009 A Comparative Study of VoIP Standards with Asterisk Advisor.

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Presentation on theme: "Fourth International Conference onDigital Telecommunications, 2009. ICDT '09. 20-25 July 2009 A Comparative Study of VoIP Standards with Asterisk Advisor."— Presentation transcript:

1 Fourth International Conference onDigital Telecommunications, 2009. ICDT '09. 20-25 July 2009 A Comparative Study of VoIP Standards with Asterisk Advisor : Lian-Jou Tsai Student : PEI-SIOU HUANG Date : 2013/3/07

2 最近研究方向 Asterisk VoIP Standards

3 Outline Abstract INTRODUCTION VOIP PROTOCOLS UNDER ANALYSIS DESCRIPTION OF THE TESTBED ANALYSIS OF THE EXPERIMENTAL RESULTS CONCLUSIONS

4 Abstract Since the apparition of Voice over IP (VoIP), many standards (mainly signaling protocols and codecs) have arisen with the aim of enabling voice calls through data networks.

5 INTRODUCTION Voice over Internet Protocol (VoIP) has become a very popular technology. extend these analysis to compare the performance of different signaling protocols and codecs for voice calls in an actual VoIP testbed based on Asterisk

6 Protocols :H.323, SIP, IAX H.323 and SIP for signaling and RTP for the transport. IAX is a new concept in VoIP as it combines both functions in the same protocol. (signaling and multimedia) VOIP PROTOCOLS UNDER ANALYSIS(1)

7 VOIP PROTOCOLS UNDER ANALYSIS(2)

8 DESCRIPTION OF THE TESTBED(1) Figure 1. Test bed under study

9 DESCRIPTION OF THE TESTBED(2) Figure 2. Connection diagram

10 DESCRIPTION OF THE TESTBED(3)

11 ANALYSIS OF THE EXPERIMENTAL RESULTS(1) Processor As it was previously remarked, when no transcoding is performed, the main processor load is the routing of the calls.

12 ANALYSIS OF THE EXPERIMENTAL RESULTS(2) Memory Utilization All results are in the range of 33 MB to 42 MB

13 ANALYSIS OF THE EXPERIMENTAL RESULTS(3) Bandwidth Consumption one for the reception traffic and one for the transmission traffic. Both of them should be equal

14 ANALYSIS OF THE EXPERIMENTAL RESULTS(4) show the benefits of trunking

15 ANALYSIS OF THE EXPERIMENTAL RESULTS(5)

16 CONCLUSIONS(1)

17 CONCLUSIONS(2) For g.711 (A law), g.726 and gsm: 54 bytes/20 ms = 2700 bytes/s = 21,6 kbps. For lpc10 will be: 54 bytes/22,5 ms = 2400 bytes/s = 19,2 kbps. For g.711: 54 bytes of headers + 160 of data bytes =214bytes/frame (25 % corresponding to headers). For g.726: 54 bytes headers + 80 data bytes = 134bytes/frame (40 % corresponding to headers). For gsm: 54 bytes headers + 33 data bytes = 87bytes/frame (62 % corresponding to headers). For lpc10: 54 bytes headers + 7 data bytes = 61bytes/frame (88 % corresponding to headers).

18 CONCLUSIONS(3) 46bytes/20 ms=2300 bytes/s = 18,4 kbps, While for lpc10 will be: 46 bytes/22,5 ms=2044 bytes/s=16,3 kbps. For g.711 (A law): 46 bytes headers + 160 databytes=206 bytes/frame (22 % corresponding toheaders). For g.726: 46 bytes headers + 80 data bytes = 126 bytes/frame (36 % corresponding to headers). For gsm: 46 bytes headers + 33 data bytes = 79 bytes/frame (58 % corresponding to headers). For lpc10: 46 bytes headers + 7 data bytes = 53 bytes/frame (87 % corresponding to headers).

19 CONCLUSIONS(4) For g.711 (A law): 188 bytes of IAX headers + 168bytes (4 packets) of UDP/IP/Ethernet headers + 4800 data bytes (30 calls x 160 bytes/frame) = 5156bytes (7 % corresponding to headers). g.726: 188 bytes IAX headers + 84 bytes (2 packets)headers of UDP/IP/Ethernet headers + 2400 data bytes (30 calls x 80 bytes/frame)= 2672 bytes (10 %corresponding to headers). gsm: 188 bytes IAX headers + 42 bytes (1 packet) of headers of UDP/IP/Ethernet headers + 990 data bytes (30 calls x 33 bytes/frame)= 1220 bytes (19 % corresponding to headers). lpc10: 188 bytes IAX headers + 42 bytes (1 packet) of headers of UDP/IP/Ethernet headers + 210 data bytes (30 calls x 7 bytes/frame)= 440 bytes (52 % corresponding to headers).

20 CONCLUSIONS(5) Different VoIP codecs and protocols have been compared obtaining empirical results about processor and memory utilization and bandwidth consumption. Results show the clear benefits of employing call trunking (under IAX) as it reduces the overhead introduced by protocol headers, especially when a high number of simultaneous calls are multiplexed.

21 References [1] J.V.Meggelen, J.Smith, and L.Madsen, Asterisk - The Future of Telephony, 2nd ed., Sebastopol (USA): Oâ™Reilly Media, Inc.,Aug.2007. [2] V.N.G.J.Soares, P.A.C.Neves, and J.J.P.Rodrigues, “Past,Present and Future of IP Telephony”, Proc.of Communication Theory, Reliability, and Quality of Service (CTRQ’08), Bucharest (Romania), Jul.2008, pp.19-24. [3] H.M.Chong, and H.S.Matthews, “Comparative Analysis of Traditional Telephone and Voice-over-Internet protocol (VoIP) Systems”, Proc.of IEEE International Symposium on Electronics and the Environment, Scottdale (AZ USA), May 2004, pp.106111. [4] B.Chatras, and S.Garcin, “Service drivers for selecting VoIP protocols”, Proc.of Telecommunications Network Strategy and Planning Symposium, Vienna (Austria), Jun.2004, pp.131-136. [5] Basicevic, M.Popovic, D.Kukolj, “Comparison of SIP and H.323 Protocols”, Proc.of The Third International Conference on Digital Telecommunications (ICDT’08), Bucharest (Romania), Jul.2008, pp.162-167. [6] T.Abbasi, S.Prasad, N.Seddigh, and I.Lambadaris, “A comparative study of the SIP and IAX VoIP protocols”, Proc.of Canadian Conference on Electrical and Computer Engineering, Saskatoon (Saskatchewan Canada), May 2005, pp.179-183. [7] M.E.Nasr, and S.A.Napoleon, “On improving voice quality degraded by packet loss in data networks”, Proc.of the TwentySecond National Radio Science Conference (NRSC 2005), Cairo (Egypt), Mar.2005,pp.465-472. [8] L.Deri, “Open Source VoIP Traffic Monitoring”, Proc.of System Administration and Network Engineering (SANE 2006), Delft (The Netherlands), May.2006.

22 [9] G.Zhang, M.Hillenbrand, and P.Muller, “Facilitating the Interoperability among Different VoIP Protocols with VoIP Web Services”, Proc.of First International Conference on Distributed Frameworks for Multimedia Applications (DFMA’05), Besançon (France), Feb.2005, pp.39-44. [10] M.Z.Alam, S.Bose, M.M.Rahman, and M.A.Al-Mumin, “Small Office PBX Using Voice Over Internet Protocol (VOIP)”, Proc.of IEEE The 9th International Conference on Advanced Communication Technology (ICACT 2007), Vol.3, Gangwon-Do (Korea), Feb.2007, pp.1618-1622. [11] Tressel, and J.Keller, “A System for Secure IP Telephone Conferences”, Proc.of the Fifth IEEE International Symposium on Network Computing and Applications, Cambridge (Massachusetts), Jul.2006, pp.231-234. [12] A. Gorti, “A fault tolerant VoIP implementation based on open standards”, Proc.of Sixth European Dependable Computing Conference (EDCC’06), Coimbra (Portugal), Oct.2006, pp.35-38. [13] Telecommunication standardization sector of ITU (ITU-T), “Visual telephone systems and equipment for local area networks which provide a non guaranteed quality of service”, International Telecommunication Union (ITU), ITU-T Recommendation H.323, Nov.1996.Available: http://www.itu.int/rec/T-REC-H.323/e. [14] Telecommunication standardization sector of ITU (ITU-T), “Packet-based multimedia communications systems”, International Telecommunication Union (ITU), ITU-T Recommendation H.323, Jun.2006. Available: http://www.itu.int/rec/T-REC-H.323/e. [15] J.Rosenberg, H.Schulzrinne, G.Camarillo, A.Johnston, J.Peterson, and R.Sparks, M.Handley, E.Schooler, “SIP: Session Initiation Protocol”, Internet Engineering Task Force (IETF), Request for Comments (RFC) 3261, Jun.2002.Available: http://www.ietf.org/rfc/rfc3261.txt. [16] M.Spencer, B.Capouch, E.Guy, F.Miller, and K.Shumard, “IAX: Inter-Asterisk eXchange Version 2, draft- guy-iax-04”, Internet Engineering Task Force (IETF), Internet-Draft, Mar.2008.Available: http://www.ietf.org/internet-drafts/draft-guyiax-04.txt.

23 [17] H.Schulzrinne, S.Casner, R.Frederick, and V.Jacobson,“RTP: A Transport Protocol for Real-Time Applications”, Internet Engineering Task Force (IETF), Request for Comments (RFC) 3550, Jul.2003. Available: http://www.ietf.org/rfc/rfc3550.txt. http://www.ietf.org/rfc/rfc3550.txt [18] E.Casilari, H.Montes, and F.Sandoval, “Modelling of Voice Traffic Over IP Networks”, Proc.of 3rd International symposium on communication systems, networks and digital signal processing (CSNDSP 2002), Stafford (UK), Jun.2002, pp.411-414. [19] P.Pragtong, K.M.Ahmed, and T.J.Erke, “Analysis and Modeling of Voice over IP Traffic in the Real Network”, IEICE Trans Inf Syst, Vol.E89-D, Nº 12, Dec.2006, pp.2886-2896. [20] J.K.Muppala, T.Bancherdvanich, and A.Tyagi, “VoIP Performance on Differentiated Services Enabled Network”, Proc.of IEEE International Conference on Networks (ICON 2000), Singapore (Republic of Singapore), Sep.2000, pp.419â“23. [21] W.Wang, S.C.Liew, and V.O.K.Li, “Solutions to performance problems in VoIP over 802.11 wireless LAN”, IEEE Transactions on Vehicular Technology, Vol.34, Nº 1, Jan.2005,pp.366-384. [22] Zabbix monitoring tools. http://www.zabbix.com/ [23] J.Postel, “User Datagram Protocol”, Internet Engineering Task Force (IETF), Request for Comments (RFC) 768, Aug.1980.Available at: http://www.ietf.org/rfc/rfc0768.txt.http://www.ietf.org/rfc/rfc0768.txt [24] Defense Advanced Research Projects Agency, “Internet Protocol”, Internet Engineering Task Force (IETF), Request for Comments (RFC) 791, Sep.1981. Available at: http://www.ietf.org/rfc/rfc0791.txt.

24 Thank You For Your Attention!


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