Bandwidth Estimation Algorithms for the Dynamic Adaptation of Voice Codec
In the recent years multimedia traffic and in particular
VoIP services are growing dramatically. We present a new algorithm
to control the resource utilization and to optimize the voice codec
selection during SIP call setup on behalf of the traffic condition
estimated on the network path.
The most suitable methodologies and the tools that perform realtime
evaluation of the available bandwidth on a network path have
been integrated with our proposed algorithm: this selects the best
codec for a VoIP call in function of the instantaneous available
bandwidth on the path. The algorithm does not require any explicit
feedback from the network, and this makes it easily deployable over
the Internet. We have also performed intensive tests on real network
scenarios with a software prototype, verifying the algorithm
efficiency with different network topologies and traffic patterns
between two SIP PBXs.
The promising results obtained during the experimental validation
of the algorithm are now the basis for the extension towards a larger
set of multimedia services and the integration of our methodology
with existing PBX appliances.
[1] T.J. Kostas, M.S. Borella, I. Sidhu, G.M. Schuster, J. Grabiec, and J.
Mahler, "Real-Time Voice Over Packet-Switched Networks". IEEE
Network, 12(1):18-27, January-February 1998.
[2] H.M. Chong and H.S. Matthews, "Comparative Analysis of Traditional
Telephone and Voice-over-Internet Protocol (VoIP) Systems". In IEEE
International Symposium on Electronics and the Environment, pp. 106-
111, 2004.
[3] S. Floyd and K. Fall, "Promoting the Use of End-to-End Congestion
Control in the Internet". IEEE/ACM Trans. on Networking, 7(6), August
1999.
[4] A.B. Johnston, SIP, Understanding the Session Initiation Protocol.
Artech House, Norwood, MA, USA, 2004.
[5] H. Sinnreich and A.B. Johnston, Internet Communications Using SIP.
Wiley Publishing, Indianapolis, IN, USA, 2006.
[6] J. Rosenberg, H. Schulzrinne, G. Camarillo, A. Johnstone, J. Peterson,
R. Sparks, M. Handley, and E. Schooler, "SIP: Session Initiation
Protocol". RFC3261, June 2002.
[7] R.S. Prasad, M. Murray, C. Dovrolis, and K. Claffy, "Bandwidth
estimation: metrics, measurement techniques, and tools". IEEE Network,
17(6):27-35, November-December 2003.
[8] A. Shriram, M. Murray, Y. Hyun, N. Brownlee, A. Broido, and M.
Fomenkov, "Comparison of Public End-to-End Bandwidth Estimation
Tools on High-Speed Links". In Passive and Active Measurement
Workshop (PAM2005), 2005.
[9] C. Dovrolis, P. Ramanathan, and D. Moore, "Packet dispersion
techniques and a capacity estimation methodology". IEEE/ACM Trans.
on Networking, 12(6):963-977, December 2004.
[10] C. Dovrolis, P. Ramanathan, and D. Moore, "What do packet dispersion
techniques measure?". In 20th Annual Joint Conference of the IEEE
Computer and Comunications Societies (INFOCOM), pp. 905-914,
2001.
[11] V.J. Ribeiro, R.H. Riedi, R.G. Baraniuk, J. Navratil, and L. Cottrell,
"pathChirp: Efficient Available Bandwidth Estimation for Network
Paths". In Passive and Active Measurements Workshop (PAM2003),
2003.
[12] V.J. Ribeiro, R.H. Riedi, and R.G. Baraniuk, "Locating Available
Bandwidth Bottlenecks". IEEE Internet Computing, 8(5):34-41, October
2004.
[13] J. Van Meggelen, J. Smith, and L. Madsen, Asterisk, the future of
Telephony. O'Reilly Media, Sebastopol, CA, USA, 2005.
[14] V. J. Ribeiro, R. King and N. Hoven, Poisson traffic generator.
Available on http://www.spin.rice.edu/Software/poisson_gen/
[1] T.J. Kostas, M.S. Borella, I. Sidhu, G.M. Schuster, J. Grabiec, and J.
Mahler, "Real-Time Voice Over Packet-Switched Networks". IEEE
Network, 12(1):18-27, January-February 1998.
[2] H.M. Chong and H.S. Matthews, "Comparative Analysis of Traditional
Telephone and Voice-over-Internet Protocol (VoIP) Systems". In IEEE
International Symposium on Electronics and the Environment, pp. 106-
111, 2004.
[3] S. Floyd and K. Fall, "Promoting the Use of End-to-End Congestion
Control in the Internet". IEEE/ACM Trans. on Networking, 7(6), August
1999.
[4] A.B. Johnston, SIP, Understanding the Session Initiation Protocol.
Artech House, Norwood, MA, USA, 2004.
[5] H. Sinnreich and A.B. Johnston, Internet Communications Using SIP.
Wiley Publishing, Indianapolis, IN, USA, 2006.
[6] J. Rosenberg, H. Schulzrinne, G. Camarillo, A. Johnstone, J. Peterson,
R. Sparks, M. Handley, and E. Schooler, "SIP: Session Initiation
Protocol". RFC3261, June 2002.
[7] R.S. Prasad, M. Murray, C. Dovrolis, and K. Claffy, "Bandwidth
estimation: metrics, measurement techniques, and tools". IEEE Network,
17(6):27-35, November-December 2003.
[8] A. Shriram, M. Murray, Y. Hyun, N. Brownlee, A. Broido, and M.
Fomenkov, "Comparison of Public End-to-End Bandwidth Estimation
Tools on High-Speed Links". In Passive and Active Measurement
Workshop (PAM2005), 2005.
[9] C. Dovrolis, P. Ramanathan, and D. Moore, "Packet dispersion
techniques and a capacity estimation methodology". IEEE/ACM Trans.
on Networking, 12(6):963-977, December 2004.
[10] C. Dovrolis, P. Ramanathan, and D. Moore, "What do packet dispersion
techniques measure?". In 20th Annual Joint Conference of the IEEE
Computer and Comunications Societies (INFOCOM), pp. 905-914,
2001.
[11] V.J. Ribeiro, R.H. Riedi, R.G. Baraniuk, J. Navratil, and L. Cottrell,
"pathChirp: Efficient Available Bandwidth Estimation for Network
Paths". In Passive and Active Measurements Workshop (PAM2003),
2003.
[12] V.J. Ribeiro, R.H. Riedi, and R.G. Baraniuk, "Locating Available
Bandwidth Bottlenecks". IEEE Internet Computing, 8(5):34-41, October
2004.
[13] J. Van Meggelen, J. Smith, and L. Madsen, Asterisk, the future of
Telephony. O'Reilly Media, Sebastopol, CA, USA, 2005.
[14] V. J. Ribeiro, R. King and N. Hoven, Poisson traffic generator.
Available on http://www.spin.rice.edu/Software/poisson_gen/
@article{"International Journal of Electrical, Electronic and Communication Sciences:52642", author = "Davide Pierattoni and Ivan Macor and Pier Luca Montessoro", title = "Bandwidth Estimation Algorithms for the Dynamic Adaptation of Voice Codec", abstract = "In the recent years multimedia traffic and in particular
VoIP services are growing dramatically. We present a new algorithm
to control the resource utilization and to optimize the voice codec
selection during SIP call setup on behalf of the traffic condition
estimated on the network path.
The most suitable methodologies and the tools that perform realtime
evaluation of the available bandwidth on a network path have
been integrated with our proposed algorithm: this selects the best
codec for a VoIP call in function of the instantaneous available
bandwidth on the path. The algorithm does not require any explicit
feedback from the network, and this makes it easily deployable over
the Internet. We have also performed intensive tests on real network
scenarios with a software prototype, verifying the algorithm
efficiency with different network topologies and traffic patterns
between two SIP PBXs.
The promising results obtained during the experimental validation
of the algorithm are now the basis for the extension towards a larger
set of multimedia services and the integration of our methodology
with existing PBX appliances.", keywords = "Integrated voice-data communication, computernetwork performance, resource optimization.", volume = "3", number = "11", pages = "1971-5", }