Performance study of FMIPv6-based cross-layer WiMAX handover scheme for supporting VoIP service

This report presents performance study of the FMIPv6-based cross-layer handover scheme for VoIP supports over mobile WiMAX network. For this performance validation and evaluation, the handover delays for four different handover mechanisms are formulated and ns2 based simulation module has been developed. The handover delay, the total delay, and the R factor representing VoIP quality are measured to evaluate the VoIP support characteristics of the FMIPv6-based cross-layer scheme. Simulation results verified that the proposed FMIPv6-based cross-layer handover scheme, compared to the non-cross-layer scheme, successfully reduces total handover delay by almost 50% for the case of layer-3 handover. Further, simulation was also evaluated in terms of R factor indicating voice quality level, of which 70 is a minimum value of a traditional PSTN call to be considered as the lower limit of a VoIP call quality [6]. Through the simulation in this study, the result revealed that the proposed scheme effectively improves VoIP call quality from unacceptable quality to acceptable quality (R factor of 75). Based on these simulation results, it was found that the proposed FMIPv6-based cross-layer handover scheme is an adequate protocol for supporting VoIP services in mobile WiMAX environment

Covert Voice over Internet Protocol communications based on spatial model

This paper presents a new spatial steganography model for covert communications over Voice over Internet Protocol (VoIP), providing a solution to the issue of increasing the capacity of covert VoIP channels without compromising the imperceptibility of the channels. Drawing from Orthogonal Modulation Theory in communications, the model introduced two concepts, orthogonal data hiding features and data hiding vectors, to covert VoIP communications. By taking into account the variation characteristics of VoIP audio streams in the time domain, a hiding vector negotiation mechanism was suggested to achieve dynamic self-adaptive ste-ganography in media streams. Experimental results on VoIP steganography show that the pro-posed steganographic method effectively depicted the spatial and temporal characteristics of VoIP audio streams, and enhanced robustness against detection of steganalysis tools, thereby improving the security of covert VoIP communications

A configurable vector processor for accelerating speech coding algorithms

The growing demand for voice-over-packer (VoIP) services and multimedia-rich applications has made increasingly important the efficient, real-time implementation of low-bit rates speech coders on embedded VLSI platforms. Such speech coders are designed to substantially reduce the bandwidth requirements thus enabling dense multichannel gateways in small form factor. This however comes at a high computational cost which mandates the use of very high performance embedded processors. This thesis investigates the potential acceleration of two major ITU-T speech coding algorithms, namely G.729A and G.723.1, through their efficient implementation on a configurable extensible vector embedded CPU architecture. New scalar and vector ISAs were introduced which resulted in up to 80% reduction in the dynamic instruction count of both workloads. These instructions were subsequently encapsulated into a parametric, hybrid SISD (scalar processor)–SIMD (vector) processor. This work presents the research and implementation of the vector datapath of this vector coprocessor which is tightly-coupled to a Sparc-V8 compliant CPU, the optimization and simulation methodologies employed and the use of Electronic System Level (ESL) techniques to rapidly design SIMD datapaths

An Analysis of the MOS under Conditions of Delay, Jitter and Packet Loss and an Analysis of the Impact of Introducing Piggybacking and Reed Solomon FEC for VOIP

Voice over IP (VoIP) is a real time application that allows transmitting voice through the Internet network. Recently there has been amazing progress in this field, mainly due to the development of voice codecs that react appropriately under conditions of packet loss, and the improvement of intelligent jitter buffers that perform better under conditions of variable inter packet delay. In addition, there are other factors that indirectly benefited VoIP. Today, computer networks are faster due to the advances in hardware and breakthrough algorithms. As a result, the quality of VoIP calls has improved considerably. However, the quality of VoIP calls under extreme conditions of packet loss still remains a major problem that needs to be addressed for the next generation of VoIP services. This thesis concentrates in making an analysis of the effects that network impairments, such as: delay, jitter, and packet loss have in the quality of VoIP calls and approaches to solve this problem. Finally, we analyze the impact of introducing forward error correction (FEC) Piggybacking and Reed Solomon codes for VoIP. To measure the mean opinion score of VoIP calls we develop an application based on the E-Model, and utilize perceptual evaluation of speech quality (PESQ)

A Survey of Bandwidth Optimization Techniques and Patterns in VoIP Services and Applications

This article surveys the various techniques adopted for optimising bandwidth for VoIP services over the period 1999-2014. The improvement of bandwidth can be realized through; silence suppression measure of repressing the silent portions (packets) in a voice conversation using Voice Activity Detection algorithm; by so doing, the transmission rate during the inactive periods of speech is reduced, and thus, the mean transmission rate can be reduced. A second measure is packet header reduction which defines a process of multiplexing and de-multiplexing packet headers to curb excesses. Voice/ Packet Header compression is considered the most productive of all the techniques, offering a scheme where VoIP packets are compressed from the 40 bytes of size to a smaller byte size of 2 bytes. When combined with aggregation, compression potentially yields a compressed size of up to 1 byte. In either case, bandwidth save is reached using compression and decompression codecs of varying data and bit rates. It is envisaged that an improvement in the performance of codecs would yield a better result in terms of enhancing results favourably in Voice over broadband networksComment: 8 pages, 7 figures. ISSN (Print): 1694-0814 | ISSN (Online): 1694-078

E-model implementation for VoIP QoS across a hybrid UMTS network

Voice over Internet Protocol (VoIP) provides a new telephony approach where the voice traffic passes over Internet Protocol shared traffic networks. VoIP is a significant application of the converged network principle. The research aim is to model VoIP over a hybrid Universal Mobile Telecommunications System (UMTS) network and to identify an improved approach to applying the ITU-T Recommendation G.107 (E-Model) to understand possible Quality of Service (QoS) outcomes for the hybrid UMTS network. This research included Modeling the hybrid UMTS network and carrying out simulations of different traffic types transmitted over the network. The traffic characteristics were analysed and compared with results from the literature. VoIP traffic was modelled over the hybrid UMTS network and the VoIP traffic was generated to represent different loads on the network from light to medium and heavy VoIP traffic. The VoIP over hybrid UMTS network traffic results were characterized and used in conjunction with the E-Model to identify VoIP QoS outcomes. The E-Model technique was implemented and results achieved were compared with results for other network types highlighted in the literature. The research identified an approach that permits accurate Modeling of VoIP QoS over a hybrid UMTS network. Accurate results should allow network design to facilitate new approaches to achieving an optimal network implementation for VoIP

Enabling a Low-delay Internet Service via Built-in Performance Incentives

The single best-effort service of the Internet struggles to accommodate divergent needs of different distributed applications. Numerous alternative network architectures have been proposed to offer diversified network services. These innovative solutions failed to gain wide deployment primarily due to economic and legacy issues rather than technical shortcomings. Our paper presents a new simple paradigm for network service differentiation that accounts explicitly for the multiplicity of Internet service providers and users as well as their economic interests in environments with partly deployed new services. Our key idea is to base the service differentiation on performance itself, rather than price. We design RD (Rate-Delay) network services that give a user an opportunity to choose between a higher transmission rate or low queuing delay at a congested network link. To support the two services, an RD router maintains two queues per output link and achieves the intended ratedelay differentiation through simple link scheduling and dynamic buffer sizing. Our extensive evaluation of the RD network services reports their performance, deployability, and security properties

Scalable Speech Coding for IP Networks

The emergence of Voice over Internet Protocol (VoIP) has posed new challenges to the development of speech codecs. The key issue of transporting real-time voice packet over IP networks is the lack of guarantee for reasonable speech quality due to packet delay or loss. Most of the widely used narrowband codecs depend on the Code Excited Linear Prediction (CELP) coding technique. The CELP technique utilizes the long-term prediction across the frame boundaries and therefore causes error propagation in the case of packet loss and need to transmit redundant information in order to mitigate the problem. The internet Low Bit-rate Codec (iLBC) employs the frame-independent coding and therefore inherently possesses high robustness to packet loss. However, the original iLBC lacks in some of the key features of speech codecs for IP networks: Rate flexibility, Scalability, and Wideband support. This dissertation presents novel scalable narrowband and wideband speech codecs for IP networks using the frame independent coding scheme based on the iLBC. The rate flexibility is added to the iLBC by employing the discrete cosine transform (DCT) and iii the scalable algebraic vector quantization (AVQ) and by allocating different number of bits to the AVQ. The bit-rate scalability is obtained by adding the enhancement layer to the core layer of the multi-rate iLBC. The enhancement layer encodes the weighted iLBC coding error in the modified DCT (MDCT) domain. The proposed wideband codec employs the bandwidth extension technique to extend the capabilities of existing narrowband codecs to provide wideband coding functionality. The wavelet transform is also used to further enhance the performance of the proposed codec. The performance evaluation results show that the proposed codec provides high robustness to packet loss and achieves equivalent or higher speech quality than state-of-the-art codecs under the clean channel condition