Hang Nguyen

WE-MQS-VoIP Priority: An enhanced LTE Downlink Scheduler for voice services with the integration of VoIP priority mode

The Long Term Evolution (LTE) is a high data rates and fully All-IP network. It is developed to support well to multimedia services such as Video, VoIP, Gaming, etc. So that, the real-time services such as VoIP, video, etc. need to be optimized. Nevertheless, the deployment of such live stream services having many challenges. Scheduling and allocating radio resource are very important in LTE network, especially with multimedia services such as VoIP. When voice service transmitted over LTE network, it is affected by many network impairments where there are three main factors including packet loss, delay, and jitter. This study proposes a new scheduler which is based on VoIP priority mode,Wideband (WB) E-model, QoS- and Channel-Aware (called WE-MQS-VoIP Priority scheduler) for voice services in LTE downlink direction. The proposed scheduling scheme is built based on the WB E-model and Maximum Queue Size (MQS). In addition, we integrate the VoIP priority mode into our scheduling scheme. Since the proposed scheduler considers the VoIP priority mode and user perception, thus, it improves significantly the system performance. The results demonstrate that the proposed scheduler not only meets QoS demands of voice calls but also outperforms Modified Largest Weighted Delay First (M-LWDF) in terms of delay, Packet Loss Rate (PLR) for all number of user (NU) and excepting NU equals 30, respectively. For Fairness Index (FI), cell throughput, and Spectral Efficiency (SE), the difference among the packet schedulers is not significant. The performance evaluation is compared in terms of Delay, PLR, Throughput, FI, and SE.

A dynamic rate adaptation algorithm using WB E-model for voice traffic over LTE network

This paper presents a dynamic adaptation algorithm of joint source-channel code rate for enhancing voice transmission over LTE network. In order to assess the speech quality, we use the Wideband (WB) E-model. In this model, both end-to-end delay and packet loss are taken into account. The goal of this paper is to find out the best suboptimal solution for improving voice traffic over LTE network with some constraints on allowed maximum end-to-end delay, allowed maximum packet loss and minimum required bandwidth. The best suboptimal choice is channel code rate corresponding to each mode of the AMR-WB codec that minimizes redundant bits generated by channel coding with an acceptable MOS reduction. Besides, this algorithm can be integrated with rate control in AMR-WB codec to offer the required mode of LTE network. Our results show that the MOS degradation is not significant, but the percent of reduced redundant bits to be very considerable. This will requires less bandwidth, thus, more mobile users can be served. The algorithm has simple computational operations, it can be applied to realtime voice communications.

E-MQS - A New Downlink Scheduler for Real-Time Flows in LTE Network

This paper proposes a new scheduling scheme which based on the extended E-model, Channel- and QoS- Aware (known as E-MQS scheduler) for real-time traffics in LTE downlink direction. The real-time services (VoIP, Video, etc.) are very sensitive to network impairments such as delay, packet loss, jitter, etc. The proposed scheduling scheme is based on the extension of the E-model and the consideration of Maximum Queue Size (MQS) as a factor for the metric. Since this scheduling scheme considers Mean Opinion Score (MOS) values, thus, it gets higher user perception. The simulation results show that the proposed scheme has the performance which not only satisfies QoS requirements of real- time services but also outperforms the Frame Level Scheduler (FLS), Modified Largest Weighted Delay First (M-LWDF) and Exponential/Proportional Fair (EXP/PF) schedulers in terms of delay, cell throughput, Fairness Index (FI) and Spectral Efficiency (SE), especially for Video flow. Our proposed scheduler also significantly improves the Packet Loss Rate (PLR) in comparison with the M-LWDF and EXP/PF schedulers for both VoIP and Video flows. The performance evaluation is compared in terms of Delay, PLR, Throughput, FI and SE for FLS, M-LWDF, EXP/PF schedulers and our proposed one.

Hybrid Source-Channel Coding with Bandwidth Expansion for Speech Data

Hybrid digital-analog (HDA) architectures have been widely developed for efficient digital transmission of analog speech, audio or video data. By considering the advantage of both digital and analog components, HDA systems gain better performances than purely analog and digital schemes in a wide range of channel conditions. However, HDA systems described in previous works are mostly designed for continuous-valued sources. In this paper, we address the problem of transmission of discrete sources over noisy channels. In particular, our work focuses on digital speech data in PCM format. We proposed two analog schemes, linear mapping, and non-linear mappings. The linear analog mapping employs an equal error protection scheme while the non-linear mapping takes into account the heterogeneous nature of error values to provide better protection to important values. The experiment results show that our HDA systems provide a better performance on a wide range of channel qualities in comparison with traditional purely digital systems.

Performance evaluation of E-MQS scheduler with Mobility in LTE heterogeneous network

This paper proposes a new scheduling scheme which based on user perception, Channel- and QoS-Aware (known as E-MQS scheduler) for real-time traffics in LTE downlink direction. The proposed scheduling scheme is based on the extension of the E-model and the consideration of Maximum Queue Size (MQS) as a an essential factor for the metric. The proposed scheduling scheme is evaluated in LTE heterogeneous traffic with mobility. The simulation results show that the proposed scheduler not only satisfies QoS requirements of real-time services but also outperforms the Frame Level Scheduler (FLS), Modified Largest Weighted Delay First (M-LWDF) and Exponential/Proportional Fair (EXP/PF) schedulers in terms of delay, cell throughput, Fairness Index (FI) and Spectral Efficiency (SE), especially for Video flow. The proposed scheduler also significantly improves the Packet Loss Rate (PLR) in comparison with the M-LWDF and EXP/PF schedulers for both VoIP and Video flows.

Hybrid digital-analog source and channel coding with adaptation

Hybrid analog digital (HDA) architectures have been widely used in communication systems for analog source over discrete-time memoryless Gaussian channels. They employ a linear coding scheme in the analog parts, while considering separately the design of source and channel codes in the digital parts. To the best of our knowledge, none of the previous HDA schemes exploit the advantages of maintaining a joint source and channel coding design in the digital segment. In this paper, we investigate the effect of the analog parts on various outputs of the digital modules in a HDA communication system, and introduce a novel HDA architecture with adaptation for the digital parts. Such adaptation allows our system to exploit the joint effect of the analog components and the channel noise on outputs of the digital components, while simultaneously taking into consideration the unequal distribution of source code outputs. Our simulations illustrate that the proposed HDA system provides robust and graceful performance (on both bandwidth compression and expansion modes) for a wide range of channel conditions.

Joint source-channel coding with adaptation

According to Shannons source-channel separation theorem, the source coding and the channel coding operations should be performed independently. However, such a method does not exploit the nature of unequiprobable source output in a finite block regime so that it can not achieve good qualities in several scenarios in practical systems. In this study, we propose a JSCC strategy which takes into account the unequiprobable source output to attain better data quality than traditional methods. An adaptation is performed in order to map source code outputs to appropriate inputs of channel code. We also propose a greedy algorithm to construct parameters for the adaptation in polynomial time. Our simulation is performed on AMR compression to estimate the effect of our approach as compared to the traditional strategy. Using this approach we find significantly improved speech quality after transmission in both cases: transfer of only one AMR codec mode and transfer of all AMR codec modes.

A New Non-intrusive Assessment Method for VoLTE Quality Based on Extended E-Model

Voice over Long Term Evolution (VoLTE) is always a main service that brings a big benefit for mobile operators. However, the deployment of VoLTE is very complex, specially for guaranteeing of Quality of Service (QoS) to meet quality of experience of mobile users. The key purpose of this paper is to present an object non-intrusive prediction model for VoLTE quality based on LTE-Sim framework and the extended E-model. This combination allows overcoming the lack of the E-model is that how to determine exactly its inputs. Besides, we propose to complement the jitter buffer as an essential input factor to the E-model. The simulation results show that the proposed model can predict voice quality in LTE network via both the E-model and the extended E-model. The proposed model does not refer to the original signal, thus, it is very suitable for predicting voice quality in LTE network for many different scenarios which are configured in the LTE-Sim framework. The simulation results also show that the effect of jitter buffer on user perception is very significant.