Anti-run-dry algorithm for optimal control of playoutbuffers
ISICT '03 Proceedings of the 1st international symposium on Information and communication technologies
Adaptive VoIP Playout Scheduling: Assessing User Satisfaction
IEEE Internet Computing
Adaptive Delay and Synchronization Control for Wi-Fi Based Mobile AV Conferencing
Wireless Personal Communications: An International Journal
Adaptive playback algorithm for interactive audio streaming over wireless ad-hoc networks
Proceedings of the 2006 international conference on Wireless communications and mobile computing
Managing mobility and adaptation in upcoming 802.21 enabled devices
WMASH '06 Proceedings of the 4th international workshop on Wireless mobile applications and services on WLAN hotspots
adaMOS: MOS-adaptive VoIP sources
WebMedia '06 Proceedings of the 12th Brazilian Symposium on Multimedia and the web
Voice transmission over IEEE 802.11 networks: main issues and restrictions
WebMedia '06 Proceedings of the 12th Brazilian Symposium on Multimedia and the web
The effect of packet dispersion on voice applications in IP networks
IEEE/ACM Transactions on Networking (TON)
Accurate and efficient SLA compliance monitoring
Proceedings of the 2007 conference on Applications, technologies, architectures, and protocols for computer communications
Proceedings of the 15th international conference on Multimedia
Media streaming via TFRC: An analytical study of the impact of TFRC on user-perceived media quality
Computer Networks: The International Journal of Computer and Telecommunications Networking
A New Buffer Algorithm for Speech Quality Improvement in VoIP Systems
Wireless Personal Communications: An International Journal
Cheating detection through game time modeling: A better way to avoid time cheats in P2P MOGs?
Multimedia Tools and Applications
Data & Knowledge Engineering
Enhancing VoIP service for ubiquitous communication in a campus WLAN with partial coverage
Computer Networks: The International Journal of Computer and Telecommunications Networking
Dynamic load balancing through real-time transport feedback control for VoIP QoS
International Journal of Computer Applications in Technology
An empirical evaluation of VoIP playout buffer dimensioning in Skype, Google talk, and MSN Messenger
Proceedings of the 18th international workshop on Network and operating systems support for digital audio and video
A moving average predictor for playout delay control in VoIP
IWQoS'03 Proceedings of the 11th international conference on Quality of service
Buffer-based low-delay playout control methods for IPTV terminals
GLOBECOM'09 Proceedings of the 28th IEEE conference on Global telecommunications
DSP subsystem for multiparty conferencing in VoIP
IMSAA'09 Proceedings of the 3rd IEEE international conference on Internet multimedia services architecture and applications
Multiobjective monitoring for SLA compliance
IEEE/ACM Transactions on Networking (TON)
Playout delay calculations based on truncated Gaussians for internet telephony
ICICS'09 Proceedings of the 7th international conference on Information, communications and signal processing
An evaluation of the performance advantage of Diffserv with respect to voice over IP flows
MILCOM'09 Proceedings of the 28th IEEE conference on Military communications
A Garch-based adaptive playout delay algorithm for VoIP
Computer Networks: The International Journal of Computer and Telecommunications Networking
An adaptive jitter buffer play-out scheme to improve VoIP quality in wireless networks
MILCOM'06 Proceedings of the 2006 IEEE conference on Military communications
QoS and resource management in distributed interactive multimedia environments
Multimedia Tools and Applications
Quantifying QoS requirements of network services: a cheat-proof framework
MMSys '11 Proceedings of the second annual ACM conference on Multimedia systems
VoIP Performance over HSPA with Different VoIP Clients
Wireless Personal Communications: An International Journal
What are suspicious VoIP delays?
Multimedia Tools and Applications
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The quality of service limitation of today's Internet is a major challenge for real-time voice communications. Excessive delay, packet loss, and high delay jitter all impair the communication quality. A new receiver-based playout scheduling scheme is proposed to improve the tradeoff between buffering delay and late loss for real-time voice communication over IP networks. In this scheme the network delay is estimated from past statistics and the playout time of the voice packets is adaptively adjusted. In contrast to previous work, the adjustment is not only performed between talkspurts, but also within talkspurts in a highly dynamic way. Proper reconstruction of continuous playout speech is achieved by scaling individual voice packets using a time-scale modification technique based on the Waveform Similarity Overlap-Add (WSOLA) algorithm. Results of subjective listening tests show that this operation does not impair audio quality, since the adaptation process requires infrequent scaling of the voice packets and low playout jitter is perceptually tolerable. The same time-scale modification technique is also used to conceal packet loss at very low delay, i.e., one packet time. Simulation results based on Internet measurements show that the tradeoff between buffering delay and late loss can be improved significantly. The overall audio quality is investigated based on subjective listening tests, showing typical gains of 1 on a 5-point scale of the Mean Opinion Score.