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ACM SIGCOMM Computer Communication Review
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MobiCom '98 Proceedings of the 4th annual ACM/IEEE international conference on Mobile computing and networking
Loss concealment for multi-channel streaming audio
NOSSDAV '03 Proceedings of the 13th international workshop on Network and operating systems support for digital audio and video
Achieving moderate fairness for UDP flows by path-status classification
LCN '00 Proceedings of the 25th Annual IEEE Conference on Local Computer Networks
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ICON '00 Proceedings of the 8th IEEE International Conference on Networks
WTCP: An Efficient Mechanism for Improving TCP Performance over Wireless Links
ISCC '98 Proceedings of the Third IEEE Symposium on Computers & Communications
Selecting an Audio Redundancy Codec Combination for Error Control in Internet Telephony
ICPADS '01 Proceedings of the Eighth International Conference on Parallel and Distributed Systems
Adaptive Resource Allocation For Multimedia Services In Wireless Communication Networks
ICDCSW '01 Proceedings of the 21st International Conference on Distributed Computing Systems
MULTIMEDIA '03 Proceedings of the eleventh ACM international conference on Multimedia
Content-based UEP: a new scheme for packet loss recovery in music streaming
MULTIMEDIA '03 Proceedings of the eleventh ACM international conference on Multimedia
Impromptu: managing networked audio applications for mobile users
Proceedings of the 2nd international conference on Mobile systems, applications, and services
Control mechanisms for packet audio in the internet
INFOCOM'96 Proceedings of the Fifteenth annual joint conference of the IEEE computer and communications societies conference on The conference on computer communications - Volume 1
IEEE Transactions on Consumer Electronics
A survey of packet loss recovery techniques for streaming audio
IEEE Network: The Magazine of Global Internetworking
Content based packet loss recovery for classical music transmissions over the internet
PCM'10 Proceedings of the Advances in multimedia information processing, and 11th Pacific Rim conference on Multimedia: Part II
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Due to the characteristics of (1) smaller bandwidth and (2) unreliable transmission media, real-time audio streaming over wireless networks is not trivial. To have smooth audio streaming over wireless networks, we propose a scheme called REDUP in this paper. Two sending modes, in which redundant information is embedded in each packet, that the REDUP scheme contains are (1) the ''redundant'' mode and (2) the ''duplicated'' mode. Let a packet i can contain three audio frames i, i-1, and i-2. In the redundant mode, frame i uses a codec of better quality than that for frames i-1 and i-2. In the duplicated mode, frames i, i-1, and i-2 use the same codec, which has lower quality than that for frame i used in the redundant mode. The ''redundant'' mode may give better quality of sound but consumes more bandwidth, while the ''duplicated'' mode gives lower quality of sound but consumes less bandwidth. The two modes are selected depending on the networking situation. Round trip time (RTT) between the mobile host and the mobile gateway is used to determine the networking situation. Two thresholds named ''upper-ratio'' and ''lower-ratio'' are set. When the average RTT exceeds the upper-ratio multiplies the maximum RTT, the networking situation is set to congested; when the average RTT is under the lower-ratio multiplies the maximum RTT, the networking situation is set to unloaded. The transmission mode is switched from ''redundant'' to ''duplicated'' when the networking situation is determined to be congested. The transmission mode is switched from ''duplicated'' to ''redundant'' when the networking situation is determined to be unloaded. In this way, the redundant information of packets is adapted to the networking situation and the bandwidth can be utilized more effectively.