Quality of Service Improvement for Voice Streaming over WirelessAd-hoc Networks using an Adaptive Playout Adjustment Algorithm

Subject Areas : Journal of Computer & Robotics

Maral Salehi ¹ , Mehdi Dehghan ²

1 - Department of Computer Engineering, Amirkabir University of Technology, Tehran, Iran
2 - Department of Computer Engineering, Amirkabir University of Technology, Tehran, Iran

Received: 2009-08-26 Accepted : 2010-02-09 Published : 2012-06-12

Keywords: Wireless Mobile Ad-hoc Networks, Voice Streaming, Adaptive Playout Algorithm, Quality of Service, Multimedia,

Abstract :

Providing a high-quality service for transmission and playing real-time voice conversations (voice streaming) over wireless ad-hoc networks is no mean feat. Buffering together with adjusting the playout time of the packets is a receiver-side solution to overcome this challenge. In this paper, a new adaptive playout adjustment algorithm is proposed to stream the voice conversations over wireless ad-hoc networks. This algorithm always tries to be aware of the network's conditions, adapts itself with these conditions and adjusts the playout time of the voice packets as efficiently as possible. It is required that not only most of the packets be received before their playout time, as scheduled in the receiver, but also that the playout time not be too long so as to adversely affect the interactivity between the sender and the receiver. The main features of the presented method are: adjusting the threshold adaptively with respect to the varying conditions of the network in order to determine the state of system; calculating the mean network jitter dynamically based on the current conditions of the network in order to calculate the playout delay for the current packet; being optimistic about the future state of the network and not using the delay history in order to calculate the mean network delay. Simulation results show that the proposed algorithm adapts itself with the network's dynamics and adjusts the playout delay for voice packets better than the other algorithms.

References:

[1] R. Ramjee, J. Kurose, D. Towsley, H. Schulzrinne, Adaptive playout mechanisms for packetized audio applications in wide area network, in: Proc. of IEEE INFOCOM, Toronto, Canada, vol. 2, pp. 680–688, 1994.
[2] M. Liu, QOS Improvement for real-time wireless VoIP, PhD. Thesis, Arizona University, 2006.
[3] V. Jacobson , M. J. Karels, Congestion avoidance and control, in: Proc. of ACM SIGCOMM Conf., Stanford, 1988.
[4] M. Salehi, QOS improvement for voice transmission over wireless LAN networks, MSc. Thesis, Amirkabir University of Technology, 2009.
[5] M. Narbutt, L. Murphy, VOIP playout buffer adjustment using adaptive estimation of network delays, in: Proc. of 18th Int. Teletraffic Congress (ITC-18), Berlin, Germany, pp. 1171–1180, September 2003.
[6] S. Moon, J. Kurose, D. Towsley, Packet audio playout delay adjustment: performances bounds and algorithms, ACM/Springer Multimedia Systems, vol. 6, pp. 17–28, 1998.
[7] Y. Liang, N. Farber, B. Girod, Adaptive playout scheduling and loss concealment for voice communication over IP networks, IEEE Trans. on Multimedia, vol. 5, no. 4, pp. 532-543, 2003.
[8] S. Jelassi, H. Youssef, Adaptive playout scheduling algorithm tailored for real-time packet-based voice conversations over wireless ad-hoc networks, in: Proc. of Elsevier Computer Networks, vol. 52, pp. 2473–2488, 2008.
[9] Z. Fu, X. Meng, S. Lu, TCP friendly rate adaptation for multimedia streaming in mobile ad hoc networks, in: Proc. of IEEE Int. Conf. on Management of Multimedia Networks and Services (MMNS’03), Belfast, Northern Ireland, UK, pp. 115–127, September 2003.
[10] M. Narbutt, M. Davis, Predicting the impact of application-layer adaptation mechanisms and MAC-layer parameter tuning on speech transmission quality in VoIP systems, Available on-line at http:// www.cnri.dit.ie/publications/, visited on January 2008.
[11] R.G. Cole, J.H. Rosenbluth, Voice over IP performance monitoring, computer communication review, ACM SIGCOMM, vol. 31, No. 2, 2001.
[12] L. Sun, E. Ifeachor, New models for perceived voice quality prediction and their applications in playout buffer optimization for VOIP networks, in: Proc. of IEEE Int. Conf. on Communication (ICC’04), vol. 27, No. 1, June 2004.
[13] N. Shacham, P. McKenney, Packet recovery in high-speed networks using coding and buffer management, In Proc. of the IEEE 1990 INFOCOM, vol.1, pp.124-131, June 1990.
[14] R.A Valenzuela, C.N. Animalu: a new voice-packet reconstruction technique, In Proc. of the IEEE 1989 Int. Conf. on Acoustic, Speech, and Signal Processing, vol.2, pp. 1334- 1336, 1989.
[15] A.K. Anandakumar, A.V. McCree, V.Viswanathan, Efficient CELP-based diversity schemes for VoIP, In Proc. of IEEE 2000 Int. Conf. on Acoustic, Speech, and Signal Processing, vol. 6, pp. 3682-3685, 2000.
[16] R.C.F. Tucker, J.E. Flood, Optimizing the performance of packet-switch speech, In Proc. of IEEE 1985 Conf. on Digital Processing of Signals in Communications, pp. 227-234, Apr. 1985.
[17] D.L. Stone, K. Jeffay, An empirical study of delay jitter management policies, Multimedia Systems 2 (1) (1995) 3–27.
[18] H. Schulzrinne, Voice communication across the Internet: a network voice terminal, Technical Report TR 92-50, Department of Computer Science, University of Massachusetts, Amherst, Massachusetts, July 1992.