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Cognitive Radio Network (CRN) is one of the emerging technologies to improve the spectrum utilization where Secondary Users (SUs) opportunistically uses the spectrum without creating significant interference to Primary Users (PUs). However, in CRN channel characteristics varies widely due to the random arrival of PUs which leads to call dropping/call blocking. Excessive call dropping reduces the quality...
Achieving high-quality voice conference calls over the Internet is a difficult task. Heterogeneous mobile devices and network dynamics must be properly managed by multiparty VoIP systems to ensure a good quality of experience. In this paper, we propose a multiparty VoIP system based on SDN technology that uses both multicast distribution and dynamic stream adaptation to optimize the conference call...
The rapid growth of wireless mobile communications raises the critical issue of meeting user expectations and requirements. This paper deals with the subjective assessment of the perceived quality of video calling services delivered on a real LTE/4G network. The main goal is to determine the list of parameters that are predominantly impacting the Quality of Experience (QoE). The influencing parameters...
As converged services are becoming the standard for corporate networks, concerns have been arising on the best way to ensure efficient use of expensive WAN bandwidth. Achieving excellent quality of multimedia and voice communication requires the allocation of large bandwidth for these services. In most cases, lossy compression codecs are used to reduce bandwidth requirements resulting in lower quality...
We present a non-intrusive data driven method for codec detection and identification in the presence of background noise. The method uses a number of speech features which are then used to train a CART classifier. We demonstrate the performance of the method using several different noise types over a wide range of SNRs. Our results show that we can identify a codec and its bit rate to an accuracy...
The next generation technology of Voice over Internet Protocol (VoIP) allows voice to be delivered over the Internet Protocol based networks. The integration of VoIP and Wireless Local Area Network (WLAN) has become popular. However, WLAN architecture was primarily designed to carry data and not for real-time applications, such as VoIP. Hence, there is a need to enhance the quality of VoIP transmission...
Stringent QoS maintenance in a wireless environment for VoIP communication is a major challenge. VoIP inherently generates constant bit rate traffic and is highly sensitive to network delay. However, unpredictable network congestion makes a VoIP session so degraded that its QoS goes below a tolerable limit. Accordingly, a suitable solution is needed to adapt varying network conditions satisfying minimum...
Application level QoS control can be used to provide more complete end-to-end QoS. This is particularly important when the network is not able to support QoS. One of the representative application level QoS mechanisms is adaptive encoding, i.e., codec-based QoS control, which adjusts encoding parameters, e.g., codec bit rate, according to the wireless channel status. This paper proposes an adaptive...
Network and service performance measurements are essential in IF (Internet Protocol) networks, e.g., for network management, network monitoring, and service quality assurance. In order to measure the service quality received by individual users, a service-specific measurement system is required. Therefore, this paper develops and implements the Network and Service Quality Measurement (NSQM) architecture,...
This paper presents a VoIP packet loss model applicable to wireless access transmission media and usable for determination of admissible call number threshold value. The main contribution is a more realistic approach for VoIP packet loss evaluation. The known bufferless fluid-flow method is applied but a new simple exact formula for ACT voice packets loss is developed. The suggested method is evaluated...
This paper presents a VoIP traffic model applicable to wireless access transmission media, which is usable for determination of admissible call number threshold value. The main observation is the influence of Voice Activity Detection function and Comfort Noise Generation in particular. The CAC threshold value determination is based on a fluid-flow approach to packet loss evaluation. The well known...
Internet television (IPTV) is rapidly gaining popularity and is being widely deployed on the Internet. In order to deliver optimum user quality of experience (QoE), service providers need to understand and balance the trade-offs involved with user (video content, display device), application (codec type, encoding bit rate) and network (network health, router queuing discipline) factors that affect...
This paper presents a method for adaptive rate control in a video conferencing solution over a variable bandwidth channel. The bandwidth is estimated using the round-trip delay of probe packets. Based on the estimated bandwidth, the video rate adaptation is done in two folds; one in the H.264 video encoder based on adaptive basic unit selection and the other by generation of fragments for encoding...
Determining how to transport delay-sensitive voice data has long been a problem in multimedia networking. The difficulty arises because voice and best-effort data are different by nature. It would not be fair to give priority to voice traffic and starve its best-effort counterpart; however, the voice data delivered might not be perceptible if each voice call is limited to the rate of an average TCP...
G.729.1 is a scalable codec for narrowband and wideband conversational applications standardized by ITU-T Study Group 16. The motivation for the standardization work was to meet the new challenges of VoIP in terms of quality of service and efficiency in networks, in particular regarding the strategic rollout of wideband service. G.729.1 was designed to allow smooth transition from narrowband (300-3400...
In this paper, we propose a scalable speech/audio coder control algorithm to improve quality of service (QoS) of voice over IP (VoIP) calls over wireless local area networks (WLANs). There is an upper bound on the number of simultaneous VoIP calls that can be connected in a basic service set (BSS) of 802.11 WLANs which can be composed of an access point (AP) and wireless stations (WSTAs). In the case...
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