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A Dynamic Rate Adaptation Algorithm Using WB E-Model for Voice Traffic

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A Dynamic Rate Adaptation Algorithm Using WB E-Model for Voice Traffic A dynamic rate adaptation algorithm using WB E-model for voice traffic over LTE network Duy-Huy Nguyen and Hang Nguyen Department of Wireless Network and Multimedia Services Institut Mines-Telecom, Telecom SudParis Samovar Laboratory, UMR 5157, CNRS, Evry, France fduy huy.nguyen, [email protected] Abstract—This paper presents a dynamic adaptation algorithm When voice traffic is transmitted over LTE network, the of joint source-channel code rate for enhancing voice transmis- voice signal firstly is compressed at Application layer by sion over LTE network. In order to assess the speech quality, we AMR-WB codec, and then it is packetized into RTP payload. 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 When this payload goes through each layer, it is packetized paper is to find out the best suboptimal solution for improving into the corresponding packet and the header is added. In order voice traffic over LTE network with some constraints on allowed to protect the voice packet when it is delivered over a noisy maximum end-to-end delay and allowed maximum packet loss. channel, some error correcting technologies are included. The The best suboptimal choice is channel code rate corresponding to Forward Error Correction (FEC) channel code is widely used each mode of the AMR-WB codec that minimizes redundant bits generated by channel coding with an acceptable MOS reduction. in LTE network for data channels is Turbo code. Channel Besides, this algorithm can be integrated with rate control in coding reduces Bit Error Rate (BER), so that the speech AMR-WB codec to offer the required mode of LTE network. quality will be improved. Channel coding encodes a k-bits Our results show that the MOS degradation is not significant, block into a n-bits codeword, thus, the number of redundant but the percent of reduced redundant bits to be very considerable. bits equal to n−k, and so that, the code rate is k=n. This means This will requires less bandwidth, so that, more mobile users can be served. The algorithm has simple computational operations, that the higher channel code rate, the higher speech quality, it can be applied to real-time voice communications. but this leads to the longer delay and the higher redundancy. Index Terms—AMR-WB, Wideband E-model, VoIP, VoLTE, Therefore, there needs to be a tradeoff between speech quality Source-Channel code rate, Adaptive algorithm and channel code rate. There are several authors who have proposed techniques to I. INTRODUCTION improve the speech quality delivered over a noisy channel. In present market of mobile communication in the world, Examples include the works of [3], [4], [5], [6], [7], and [8]. 3GPP LTE is developing strongly and is deployed by the But the closest works related to our paper are represented most communications operators. LTE network is based on All- in [3], [7], and [8]. The authors in [3] present a dynamic IP network and does not support circuit switching method joint source channel coding rate adaptation algorithm for VoIP which is utilized to provide the voice call service in 3G using AMR codec. The algorithm computes the optimal rates networks. So that, in order to support voice service over LTE allocated to each frame for a set given QoS constraints. The network, additional technology has to be included. Voice over aim of their paper is to find the tradeoff between packet loss LTE network (VoLTE) service was developed to supply voice recovery and end-to-end delay to maximize perceived speech and video communication and Short Message Service (SMS) quality. In [7], the authors propose an optimization issue for on the LTE network [1]. According to [1], there are two supplying unequal error protection of speech frames according types of voice traffic over LTE network, those are VoLTE to their importance. An optimization framework for identifying and VoIP. VoLTE was launched in 2012, and at present, the optimal joint source-channel code rate of each voice frame many mobile network operators in the world provide VoLTE based on the frame perceptual importance is proposed in [8]. service. VoLTE is VoIP (Voice over Internet Protocol) based In that paper, the quality of the received speech signal is multimedia service in which voice call and video conference maximized. services can be supplied. VoLTE is signalling protocol that The aim of this paper is to extend results in [3] in context of enables carrying of voice packets and is guaranteed of given voice traffic over LTE network. Besides, in stead of finding the QoS information by LTE network operator [2]. Otherwise, tradeoff between packet loss recovery and end-to-end delay to VoIP relies on the internet which is done on a “best effort” maximize the perceived speech quality, our proposal focuses basis to deliver voice packets. It is said that VoLTE is basically on finding out the compromise between source code rate a subset of VoIP. It is enhanced VoIP over specific access and channel code rate to minimize the number of redundant technology (LTE) with given QoS information, but it is still bits generated by channel coding with an acceptable Mean VoIP. Opinion Score (MOS) degradation. We would like to offer 1 an other point of view of choosing the channel code rate LTE network for voice compression and decompression. It is corresponding to each mode of AMR-WB codec for voice detailed described in [11]. AMR-WB codec uses a sampling traffic over LTE network. In order to assess speech quality, we rate of 16 kHz, which covers 50-7000 Hz audio bandwidth. use the Wideband E-model [9]. In this model, the transmission It has 9 different codec modes (from mode 0 to mode 8) rating factor (Rwb) is used as a measure of subjective quality. corresponding to 9 source bit rates in range from 6.6 Kb/s This factor is then mapped to the corresponding MOS. In this to 23.85 Kb/s. Each of them generates encoded 20 ms speech paper, a suboptimal solution for joint source-channel code rate frame and switches among them every 20 ms. The bits in adaptation is proposed. The aim of this solution is to find the encoded speech frame to be ordered according to their out the most suitable channel code rate for each mode of subjective importance. These bits are divided into three classes AMR-WB codec and for entire modes of AMR-WB codec. with reducing perceptual importance: Class A, Class B and It takes into account some constraints on maximum allowed Class C. Total bits of each class depends on codec mode. In end-to-end delay and maximum permitted packet loss rate this study, we consider the same level of error protection for for voice traffic over LTE network. We do not consider the these three classes. Thus, the bits of these classes are equally perceived speech quality as the first target to optimize. We protected by channel coding. want to present a suboptimal solution for the tradeoff between In LTE network, AMR-WB codec is configured into 3 speech quality and redundancy caused by channel coding. This configurations [12] as follows: means that our algorithm find out the suboptimal solution for • Configuration A (Config-WB-Code 0): 6.6, 8.85, and minimizing the redundant bits generated by channel coding 12.65 Kb/s (Mandatory multi-rate configuration). with an acceptable MOS reduction. The rest of this paper is • Configuration B (Config-WB-Code 2): 6.6, 8.85, 12.65, organized as follows: Overview of voice transmission over and 15.85 Kb/s. LTE network is described in section II. In section III, we • Configuration C (Config-WB-Code 4): 6.6, 8.85, 12.65, present the proposed algorithm. The simulation results and and 23.85 Kb/s. performance evaluation of the proposed algorithm are analysed These configurations are used to simplify the negotiation of in section IV. The conclusion and future work is represented bit rate between the user equipment and the base station, thus in section V. will simplify the implementation and testing. The remaining II. VOICE TRAFFIC OVER LTE NETWORK:OVERVIEW bit rates can still be used for other purposes in mobile networks. In order to choose a bit rate, the receiver measures A. Voice traffic protocol stack layers quality of radio channel. The quality indicator (QI) is used In LTE network, the speech frame is packetized sequen- for this purpose. It is defined as an equivalent carrier-to- tially with network protocols, including Real-time Transport interference (C/I) ratio. The C/I ratio then compared to a set Protocol (RTP), User Datagram Protocol (UDP) and Internet of predefined thresholds to decide which mode to be used. Protocol (IP). And then, it will be encapsulated with other Switching among modes in a configuration depend on the radio protocols as Packet Data Convergence Protocol (PDCP), rate control algorithm in AMR-WB codec. The criterion for Radio Control Link (RLC) and Mac Access Control (MAC). mode switching is threshold value of C/I ratio. These threshold All of these protocols will add their headers into the packetized values depend on the channel condition, frequency hopping speech packet. The sizes of these protocols headers as follows: scheme, network configuration and other factors. Furthermore, RTP - 12 bytes, UDP - 8 bytes, IP - 20 bytes (with IPv4) and network conditions change over time, so that, even well- 40 bytes (with IPv6), PDCP - 1 byte, RLC - 1 byte, and MAC selected adaption thresholds will not be best.
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