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Adaptive Playout Scheduling for Voice over IP: Event-Triggered Control Policy
KTH, School of Electrical Engineering (EES), Sound and Image Processing.
KTH, School of Electrical Engineering (EES), Communication Networks.ORCID iD: 0000-0002-4876-0223
KTH, School of Electrical Engineering (EES), Sound and Image Processing.
(English)In: IEEE Multimedia, ISSN 1070-986X, E-ISSN 1941-0166Article in journal (Other academic) Submitted
Abstract [en]

We study adaptive-playout scheduling for Voice over IP using the frame-work of stochastic impulse control theory. We use the Wiener process tomodel the fluctuation of the buffer length in the absence of control. In thiscontext, the control signal consists of length units that correspond to insert-ing or dropping a pitch cycle. We define an optimality criterion that hasan adjustable trade-off between average buffering delay and average controlsignal (the length of the pitch cycles added plus the length of the pitch cyclesdropped), and show that a band control policy is optimal for this criterion.The band control policy maintains the buffer length within a band regionby imposing impulse control (inserted or dropped pitch cycles) whenever thebounds of the band are reached. One important property of the band controlpolicy is that it incurs no packet-loss through buffering if there are no out-of-order packet-arrivals. Experiments performed on both synthetic and realnetwork-delay traces show that the proposed playout scheduling algorithmoutperforms two recent algorithms in most cases.

Keyword [en]
Playout scheduling, jitter buffer, impulse control
National Category
Telecommunications
Identifiers
URN: urn:nbn:se:kth:diva-24218OAI: oai:DiVA.org:kth-24218DiVA: diva2:345497
Note
QS 2011 QS 20120326Available from: 2010-08-25 Created: 2010-08-25 Last updated: 2012-03-26Bibliographically approved
In thesis
1. Robust Multimedia Communications over Packet Networks
Open this publication in new window or tab >>Robust Multimedia Communications over Packet Networks
2010 (English)Doctoral thesis, comprehensive summary (Other academic)
Abstract [en]

Multimedia communications over packet networks, and in particular the voice over IP (VoIP) application, have become an integral part of society. However, the unreliable and heterogeneous nature of packet networks has led to a best-effort delivery of services. Delay, limitation of bandwidth, and packet-loss rate all affect the quality of service (QoS). In this thesis, we address two important network impairments in the design of robust multimedia communication systems: packet delay-variation and packet-loss.

Paper A considers the mitigation of the effect of packet delay-variation for audio communications by introducing a buffer at the receiver side. A new adaptive playout scheduling approach is proposed to control the buffering length, or, equivalently, the packet playout deadlines, in response to varying network conditions. A Wiener process is used to model the fluctuation of the buffering length without any playout adjustment. The playout scheduling problem is then reformulated as a stochastic impulse control problem by taking the playout adjustment as the control signal. The proposed approach is shown to be the optimal solution to the new control problem. It is demonstrated experimentally that the proposed approach provides improved perceived conversional quality.

Papers B, C and D address the packet-loss issue. Paper B focuses on the design of a low-complexity packet-loss concealment (PLC) method that is compatible with existing speech codecs for VoIP application. The new method is rigorously motivated based on the autoregressive (AR) speech model and the minimum mean squared error (MMSE) criterion. The effect of model estimation error on the prediction of the missing speech segment is also considered and an upper bound for the prediction error is derived. Both the theoretical and experimental results provide insight in the performance of the heuristically designed PLC methods. On the other hand, Paper C and D consider an active packet-loss-resilient coding scheme, namely multiple description coding (MDC). In general, MDC can be used for the transmission of any media data. Paper C derives a simple and accurate approximation of the rate-distortion lower bound of a particular multiple- description scenario and then demonstrates that the performance loss of some practical MD systems can be evaluated easily with the new approximation. Paper D studies the performance limit of a vector Gaussian multiple description scenario. An outer bound to the rate-distortion region is derived, and the outer bound is tight when the problem specializes to the scalar Gaussian case.

 

Place, publisher, year, edition, pages
Stockholm: KTH, 2010. xii, 37 p.
Series
Trita-EE, ISSN 1653-5146 ; 2010:036
National Category
Telecommunications
Identifiers
urn:nbn:se:kth:diva-24223 (URN)
Public defence
2010-08-30, Salongen, KTHB, Osquars backe 25, Stockholm, 10:00 (English)
Opponent
Supervisors
Note
QC20100830Available from: 2010-08-30 Created: 2010-08-25 Last updated: 2010-08-31Bibliographically approved

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Dán, György

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