Voice Codec Research Articles

Voice performance analysis using voice codec by packet fragmentation and contention free periods in wireless networks

The admission control is required to maintain the established route between the source and the destination in the wireless network. To maintain the stability of the route, the wireless channel parameters has to be adopted appropriately. Thus, this study analyzes the wireless access medium parameter through direct coordination (DCF) and point-to-point coordination method. The packets are fragmented in DCF and contention free period interval are adjusted to study the QoS parameters for various VOIP codec using OPNET simulation tool. The result shows that packet fragmentation to 256 bytes and contention free period for 20 ms improves QoS for G.729 for Voice traffic.

Pitch‐based steganography for Speex voice codec

AbstractThis paper presents an improved version of a steganographic algorithm for IP telephony called HideF0. It is based on approximating the F0 parameter, which is responsible for conveying information about the pitch of the speech signal. The bits saved due to simplification of the pitch contour are used for the hidden transmission. In our experiments, the proposed method was applied to the narrowband Speex codec working in five different modes, with bitrates between 5,950 bps and 24,600 bps. We showed that HideF0 was able to create hidden channels with steganographic bandwidths of around 200 bps at the expense of a steganographic cost of between 0.5 and 0.7 MOS, depending on the Speex mode. Because of placing the approximation flag in the voice packet header, the improved version of the proposed algorithm yielded a significantly lower decrease in speech quality, when compared with the original version of HideF0. In addition, for low bitrates of the hidden channel (i.e., below ca. 50 bps) it was able to operate without introducing any steganographic cost. Copyright © 2016 John Wiley & Sons, Ltd.

Intelligent Adaptive Energy Efficiency & Effective Signal Buffer Management Algorithm for VoIP (QoS) over Manet

Energy Efficient in the VoIP transmission is the major criteria and issues to be maintained the packet and data while processing VoIP (QoS) over Manet environment. Mobile Adhoc network (MANETs) presents a good platform for the fast deployment of VoIP services in many application scenarios. The energy efficiency mechanism for the data packet of voice in the queuing system have not been concentrated which actually consumes the more power while in the queue buffer management. Manet provide a considerable complexity that makes the transmission of real-time applications like VoIP a great challenge due to insufficient energy for packet transmission. In this research a new algorithm is proposed which enhance the performance of energy in manet carrying VoIP traffic. Through a simulation and mathematical expression we analyze and evaluate QoS indicators such as Distance, power, Bandwidth. Several voice codec are studied to determine their effect of insufficient energy. In the existing methods queuing energy have been used depend on number of nodes in certain environment which is drawbacks with mobile nodes. Hence a QoS based Intelligent Adaptive Energy Efficiency & Effective signal buffer management algorithm for voip(QoS) over Manet. The main objective of this paper is to segregate the energy altitude as per detachment the bandwidth is assign by using our IAE 3 signal barrier algorithm in manet environment. In great and dynamic networks, central algorithms are not suitable due to the lack of responsiveness. The bandwidth allocation is done by IAE 3 signal buffer algorithm.

Media Access Delay and Throughput Analysis of Voice Codec with Silence Suppression on Wireless Ad Hoc Network

The rapid growth in internet has led to the emergence of different wireless technologies and simultaneously there has been a tough challenge to provide QoS over such wireless technologies.The demand for various real time multimedia application especially voice and video applications to be available on the internet has also increased. Multimedia applications need a lot more bandwidth and have different QoS requirements than the applications that were used in early years of the internet. The main reason for the poor quality of voice over IP is because we are transmitting the real time applications over the IP network which is mainly meant to transfer only data and not for voice transmission. The purpose of this paper is to check the performance of VOIP application under different CODEC's such as G.711, G.729 and G.723.1 over wireless network with and without silence suppression. Another issue addressed in this paper is the effect of increasing nodes on voice transmission for the different codecs. Parameters like media access delay and throughput are taken into consideration. Simulation is performed using OPNET modeler simulator.

Qos Constrained Communication using Voice Codecs over Multicast Ad Hoc Networks

QoS support for group communication over MANETs using real time applications is essential to fulfill the requirements of users. It is very challenging for a multicast ad hoc routing protocol to perform under the QoS constraints due to the unstable network environment, unpredictable traffic loads, mobility, scalability and limited resources etc. Researchers have developed some standards to support real time audio/video traffic over network such as Voice over IP (VoIP), H.323, H.264 etc. Multicast ad hoc routing protocols are designed to work with the traditional traffic streams, so there is need to analyze their performance using real time traffic in order to extend their capabilities. In this paper, we will explore the efficiency of PUMA multicast routing protocol using various voice codecs under the QoS constraints.

무선 IP 네트워크에서 전용선 모뎀 사용가능성 검증

A leased line modem usability was evaluated and investigated in the wireless internet protocol(IP) network. The signal of the modem in the circuit switching network was translated to IP packet by using several voice codecs (PCM, G.711A, G.711μ, and etc.) and transmitted through the wireless IP network. The wireless IP network was simulated by the Tactical information and communication network Modeling and simulation Software(TMS). The performance and usability of the leased line modem are simulated using the system-in-the-loop(SITL) function of TMS with respect to packet delay, jitter, packet discard ratio, codecs, and wireless link BER.

New Admission Control Policy for a Flexible VoLTE Service

LTE is the most recent mobile technology deployed everywhere in the whole wide world. It provides high Downlink and Uplink throughput for all service classes especially for voice class. All the services offered by LTE are based on IMS architecture. The goal of this paper is to present the developed admission control algorithm for VoLTE based on the Load control information and the UE capability in term of the supported voice codec. This developed algorithm has high performance results compared to the existing one that uses only G.711.

On the undetectability of transcoding steganography

AbstractTranscoding Steganography (TranSteg) is a fairly new IP telephony steganographic method that is characterized by a high steganographic bandwidth, low introduced distortions, and high undetectability. TranSteg utilizes compression of the overt data to free space for the secret data bits. In this paper, we focus on evaluating different possibilities for TranSteg detection. Building on the previous works, we perform a wide analysis of different steganalysis methods to assess the possibility of TranSteg detection and identify the most ‘undetectable’ pairs of voice codecs. Copyright © 2015 John Wiley & Sons, Ltd.

Audio Scenarios Detection Technique

proposed research objective is to add to a framework for programmed recognition of sound. In this framework the real errand is to distinguish any information sound stream investigate it & anticipate the likelihood of diverse sounds show up in it. To create and industrially conveyed an adaptable sound web crawler a flexible sound search engine. The calculation is clamor and contortion safe, computationally productive, and hugely adaptable, equipped for rapidly recognizing a short portion of sound stream caught through a phone microphone in the presence of frontal area voices and other predominant commotion, and through voice codec pressure, out of a database of over accessible tracks. The algorithm utilizes a combinatorial hashed time-recurrence group of stars examination of the sound, yielding ordinary properties, for example, transparency, in which numerous tracks combined may each be distinguished.

Design of Wireless Voice Communication System in Underground Coal Mine Based on ZigBee

For disadvantages of the present mine voice communication systems, a kind of wireless voice communication system based on Zig Bee is put forward. The paper provides detailed informations about hardware and software of the wireless voice communication device. In the system, adopt CC2530 as RF sending-receiving unit of voice communication node, convert speech signals to digital or analog signals by CSP1027, encode or decode quantized voice data by AMBE voice codec technology and realize voice message two-way wireless communication over the Zig Bee wireless communication protocol IEEE 802.15.4 between voice communication devices. Experiments have shown that voice communication device can get a clear voice and have a high reliability in the effective distance of communication, meet the requirements of voice communication.

Enhanced Quality of Service in Worldwide Interoperability for Microwave Access Networks

The WiMAX system effectively supports wide variety of broadband wireless access technologies, which including high speed internet and multimedia access with high Quality of service (QoS) requirements. Real time services such as VoIP are becoming popular and are major revenue earners for network service providers. However, there are still many challenges that need to be addressed to provide a steady and good quality voice connection over the best effort WiMAX network .To support flexibility, efficiency and various requirements of QoS over a range of different applications and environments several provisioning and mechanisms are provided .This research paper investigates the performance of VoIP traffic over WiMAX networks, the impact of various voice codec schemes and statistical distribution for VoIP over WiMAX has been investigated in detail. Through various simulation experiments under realistic networking scenarios, this study provides an insight into the VoIP performance in the WIMAX networks. The simulations results indicate that better choice of voice codec's and statistical distribution have significant impact on VoIP performance in the WiMAX networks and Performance of selected parameters will be done using the network simulator OPNET

Quality of Experience of Voice Services in Corporate Network

This paper aims on a quality estimation of voice services in converged corporate networks. Besides the techno-centric Quality of Service (QoS) metrics a research area called Quality of Experience (QoE) provides metrics and methods for quality evaluation from the end-user's perspective. This contribution focuses on a QoE estimation of Voice over IP (VoIP) calls. Existing methods of voice quality estimation are compared on different voice codecs tested on a network topology suffering from distortions of real network. Finally, a regression analysis is employed to provide better understanding of impact of network conditions (delay, jitter, packet loss) on the VoIP service quality. The results can improve the voice quality monitoring systems in corporate networks.

A-LNT: A Wireless Sensor Network Platform for Low-Power Real-Time Voice Communications

Combining wireless sensor networks and voice communication for multidata hybrid wireless network suggests possible applications in numerous fields. However, voice communication and sensor data transmissions have significant differences, Meanwhile, high-speed massive real-time voice data processing poses challenges for hardware design, protocol design, and especially power management. In this paper, we present a wireless audio sensor network platform A-LNT and study and discuss key elements for systematic design and implementation: node hardware design, low-power voice codec and processing, wireless network topology, hybrid MAC protocol design based on superframe, radio channel allocation, and clock synchronization. Furthermore, we discuss energy management methods such as address filtering and efficient power management in detail. The experimental and simulation results show that A-LNT is a lightweight, low-power, low-speed, and high-performance wireless sensor network platform for multichannel real-time voice communications.

Transform Coding Based on Source Filter Model in the MDCT Domain

State-of-the-art voice codecs have been developed to extend the input bandwidth to enhance quality while maintaining interoperability with a legacy codec. Most of them employ a modified discrete cosine transform (MDCT) for coding their extended band. We propose a source filter model-based coding algorithm of MDCT spectral coefficients, apply it to the ITU-T G.711.1 super wideband (SWB) extension codec, and subjectively test it to validate the model. A subjective test shows a better quality over the standardized SWB codec.

Efficient speech encryption using chaotic cat map for code-excited linear prediction based coders in packet networks

The increasing importance of multimedia applications is placing a great insist on content protection and customer privacy. Communications can be intercepted, especially over wireless links. Since encryption can effectively prevent eavesdropping, its use is widely advocated. The codec G. 729 based CS-ACELP algorithm is standardized as voice codec by ITU-T for multimedia and Voice over Internet Protocol (VoIP) applications. In this paper, we introduce a speech encryption method based chaotic cat map algorithm. Cat map extended to two-dimensional NxN matrix. It takes concepts from linear algebra and uses them to change the positions of the values of the matrix. The result after applying the Cat Map will be shuffled signals that contain the same values of the original signals. We applied our encryption scheme to the standard ITU-T G.729 standard speech coder to evaluate its performance. Simulation results show that G.729 based cat map encryption is very efficient since the encrypted speech is similar to a white noise. The perceptual evaluation of speech quality (PESQ) and enhanced modified bark spectral distortion (EMBSD) tests for speech extracted from TIMIT database confirm the efficiency of our proposed scheme.

Performance of VoIP in Wired-Cum-Wireless Ethernet Network

The Voice over Internet Protocol (VoIP) is a rapidly growing technology that enables transport of voice over data networks such as Ethernet Local Area Networks (LANs). This growth is due to the integration of voice and data traffic over the existing network infrastructure, low cost, and improved network management offered by the technology. This paper reports on the performance of VoIP traffic characteristics in a wired-cum-wireless Ethernet LAN. The effect of increasing the number of VoIP wireless clients, different voice codec schemes, and packet arrival distributions on system performance is investigated. Through various simulation experiments under realistic network scenarios, such as Small Office Home Office (SOHO) and campus networks, this paper provides an insight into the performance of VoIP over Ethernet LANs. Simulation results show that VoIP clients and voice codec schemes have significant effect on system performance. The authors preformed OPNET-based simulations to validate their experiments.

An unified VoIP model for workload generation

This paper presents a new model for VoIP workload generation. The novelty of our proposal consists in modeling the sessions by characterizing both the user behavior (session level) and the packet generation for an active call (intra-session level) with easily measured parameters and low computational complexity. This approach also facilitates systematic study of changes in user behavior and voice codec. The session level was modeled by analysis of call-holding time and time interval between successive calls. The model for call-holding time, characterizing the individual user behavior, uses the Pareto type 2 probability distribution. The time interval between calls is obtained from aggregate traffic and can be modeled by exponential probability distribution. Aggregate traffic is obtained by superposition of simultaneous sessions. The data used to characterize the session level were collected at the backbone of two Brazilian telecommunication carriers. The model for intra-session level comprises the characterization of the packet size and the packet inter-arrival time. The intra-session model was based on data generated in a laboratory environment, in order to properly characterize the codec influence on packet generation and to avoid the effects of delay, jitter and loss commonly present in an operational network. Models for constant bit rate and variable bit rate codecs were considered. A simulator was implemented and the results indicate that our model properly mimics the characteristics observed in real traffic and can be used for VoIP modeling and workload generation. Additionally, an application to automate the performance analysis was developed.

무음 압축을 이용하는 음성 통신 시스템을 위한 동적 버퍼 관리 알고리즘

본 논문은 무음 압축(silence suppression)을 이용하는 음성 코덱 이용 시 음성 통신 용량(voice capacity)을 늘리기 위한 동적 버퍼 관리 알고리즘인 silence drop first(SDF) 알고리즘을 제안한다. SDF 알고리즘은 음성 통신을 위한 버퍼에서 오버플로우가 발생할 경우 무음 패킷을 음성 패킷보다 먼저 폐기하는 기법을 이용하여 상대적으로 더 많은 음성 패킷을 종단간 전송한다. 알고리즘의 성능 평가를 위하여 G.729A 및 G.711 음성 코덱을 사용하였다. 성능 분석 결과 제안하는 기법을 이용할 경우 MOS가 3.0인 수준의 품질을 유지하면서 G.729A 코덱에 대해 약 84.21%, G.711 코덱에 대해 약 38.46%의 사용자가 추가로 네트워크 이용이 가능함을 확인할 수 있었다. 또한 제안하는 기법 이용 시 동일 사용자 수에 대해 음성 통화 품질을 보장하기 위해 필요한 무음 패킷생성 주기 및 링크 용량의 제한을 완화할 수 있음을 확인하였다. This paper proposes silence drop first(SDF) active buffer management algorithm to increase the voice capacity when silence suppression is used. This algorithm finds and drops silence packet rather than voice packet in the queue for resolving buffer overflow of queue. Simulations with voice codec of G.729A and G.711 are performed. By using proposed SDF algorithm, the voice capacity is increased by 84.21% with G.729A and 38.46% with G.711. Further more, SDF algorithm reduces the required link capacity and loosens the silence packet inter-arrival time limit to provide target voice quality compared with that of conventional algorithms.

On the quality of VoIP with DCCP for satellite communications

SUMMARYWe present experimental results for the performance of selected voice codecs using Datagram Congestion Control Protocol (DCCP) with CCID4 congestion control over a satellite link. We evaluate the performance of both constant and variable data rate speech codecs for a number of simultaneous calls using the ITU E‐model. We analyse the sources of packet losses and additionally analyse the effect of jitter, which is one of the crucial parameters contributing to Voice over IP (VoIP) quality and has, to the best of our knowledge, not been considered previously in the published DCCP performance results. We propose modifications to the CCID4 algorithm and demonstrate how these improve the VoIP performance, without the need for additional link information other than what is already monitored by CCID4. We also demonstrate the fairness of the proposed modifications to other flows. Although the recently adopted changes to TCP‐Friendly Rate Control specification alleviate some of the performance issues for VoIP on satellite links, we argue that the characteristics of commercial satellite links necessitate consideration of further improvements. We identify the additional benefit of DCCP when used in VoIP admission control mechanisms and draw conclusions about the advantages and disadvantages of the proposed DCCP/CCID4 congestion control mechanism for use with VoIP applications. Copyright © 2012 John Wiley & Sons, Ltd.

Speech codecs for high-quality voice over ZigBee applications: Evaluation and implementation challenges

The ZigBee standard is designed to enable the deployment of low-cost, low-power wireless sensor and control networks based on the IEEE 802.15.4 physical radio standard. Despite the low data rates of ZigBee, its use for transmission of voice has been proven to be feasible. This article presents the evaluation and implementation of a voice codec performed in the framework of the Z-Phone project, which targeted the development of a low-cost, ultra-low-power, long-range, ergonomic wireless headset using ZigBee technology to transmit voice in full-duplex mode for use with leading PC VoIP programs. In particular, it presents all the criteria taken into account in selecting the most suitable codec, and also the details and performance of its implementation.