A Comparative Study of VoIP Standards with Asterisk Advisor : Lian-Jou Tsai

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A Comparative Study of VoIP
Standards with Asterisk
Fourth International Conference onDigital Telecommunications, 2009. ICDT '09.
20-25 July 2009
Advisor : Lian-Jou Tsai
Student : PEI-SIOU HUANG
Date : 2013/3/07
最近研究方向
 Asterisk
 VoIP Standards
Outline
 Abstract
 INTRODUCTION
 VOIP PROTOCOLS UNDER ANALYSIS
 DESCRIPTION OF THE TESTBED
 ANALYSIS OF THE EXPERIMENTAL RESULTS
 CONCLUSIONS
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.
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
VOIP PROTOCOLS UNDER
ANALYSIS(1)
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(2)
DESCRIPTION OF THE TESTBED(1)
Figure 1. Test bed under study
DESCRIPTION OF THE TESTBED(2)
Figure 2. Connection diagram
DESCRIPTION OF THE TESTBED(3)
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.
ANALYSIS OF THE
EXPERIMENTAL RESULTS(2)
Memory Utilization
All results are in the range of 33 MB to 42 MB
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
ANALYSIS OF THE
EXPERIMENTAL RESULTS(4)
show the benefits of trunking
ANALYSIS OF THE
EXPERIMENTAL RESULTS(5)
CONCLUSIONS(1)
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).
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).
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).
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.
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.
[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, draftguy-iax-04”, Internet Engineering Task Force (IETF), Internet-Draft, Mar.2008.Available:
http://www.ietf.org/internet-drafts/draft-guyiax-04.txt.
[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.
[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.
[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.
Thank You For Your Attention!
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