Objective Measurement of Speech Quality in VoIP over Wireless LAN during Handoff

Quality of Service is a very important factor to determine the quality of a VoIP call. Different subjective and objective models exist for evaluating the speech quality in VoIP. E-model is one of the objective methods of measuring the speech quality; it considers various factors like packet loss, delay and codec impairments. The calculations of Emodel are not very accurate in case of handovers – when a VoIP call moves from one wireless LAN to another. This project conducted experimental evaluation of performance of E-model during handovers and proposes a new approach to accurately calculate the speech quality of VoIP during handovers. A detailed description of the experimental setup and the comparison of the new approach with E-model is presented in this report

Using Transcoding for Hidden Communication in IP Telephony

The paper presents a new steganographic method for IP telephony called TranSteg (Transcoding Steganography). Typically, in steganographic communication it is advised for covert data to be compressed in order to limit its size. In TranSteg it is the overt data that is compressed to make space for the steganogram. The main innovation of TranSteg is to, for a chosen voice stream, find a codec that will result in a similar voice quality but smaller voice payload size than the originally selected. Then, the voice stream is transcoded. At this step the original voice payload size is intentionally unaltered and the change of the codec is not indicated. Instead, after placing the transcoded voice payload, the remaining free space is filled with hidden data. TranSteg proof of concept implementation was designed and developed. The obtained experimental results are enclosed in this paper. They prove that the proposed method is feasible and offers a high steganographic bandwidth. TranSteg detection is difficult to perform when performing inspection in a single network localisation.Comment: 17 pages, 16 figures, 4 table

Analysing the characteristics of VoIP traffic

In this study, the characteristics of VoIP traffic in a deployed Cisco VoIP phone system and a SIP based soft phone system are analysed. Traffic was captured in a soft phone system, through which elementary understanding about a VoIP system was obtained and experimental setup was validated. An advanced experiment was performed in a deployed Cisco VoIP system in the department of Computer Science at the University of Saskatchewan. Three months of traffic trace was collected beginning October 2006, recording address and protocol information for every packet sent and received on the Cisco VoIP network. The trace was analysed to find out the features of Cisco VoIP system and the findings were presented.This work appears to be one of the first real deployment studies of VoIP that does not rely on artificial traffic. The experimental data provided in this study is useful for design and modeling of such systems, from which more useful predictive models can be generated. The analysis method used in this research can be used for developing synthetic workload models. A clear understanding of usage patterns in a real VoIP network is important for network deployment and potential network activities such as integration, optimizations or expansion. The major factors affecting VoIP quality such as delay, jitter and loss were also measured and simulated in this study, which will be helpful in an advanced VoIP quality study. A traffic generator was developed to generate various simulated VoIP traffic. The data used to provide the traffic model parameters was chosen from peak traffic periods in the captured data from University of Saskatchewan deployment. By utilizing the Traffic Trace function in ns2, the simulated VoIP traffic was fed into ns2, and delay, jitter and packet loss were calculated for different scenarios. Two simulation experiments were performed. The first experiment simulated the traffic of multiple calls running on a backbone link. The second experiment simulated a real network environment with different traffic load patterns. It is significant for network expansion and integration

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

Video Quality Measurement For 3G Handsets

Internet provides many services. VOIP (Voice over IP) is one such service also known as Internet Telephony or IP Telephony. Using VOIP we can make voice telephony calls, participate in video conferences, etc over data networks (WAN'S and LAN'S) or internet. VOIP operates by first converting voice data into digital form, organizing them into packets, transmitting them through the most convenient route to their destination and finally reassembling them at the destination. Protocols like SIP/RTP, H.323, MGCP are designed which perform all the above steps. This project aims to make a video call from a 3G Mobile to an IP phone via Asterisk Gateway. Asterisk to act as bridge for video call between 3G-IP network must capture the audio/video stream from 3G mobile, convert captured stream into an IP compatible stream and send stream to an IP client and vice-versa. Asterisk needs to support AMR codec for audio and MPEG-4 codec for video and H.324M protocol stack for capturing audio/video streams from 3G Mobile. Asterisk currently supports audio codec's like GSM, G.729, A-law, and U-law. It allows H.261, H.263 video streams as pass-through. It supports VOIP protocols like SIP/RTP, MGCP, and H.323 which allows it to interface with other devices. This project aims to implement AMR codec, H.324M protocol stack, MPEG-4, bridging functions between SIP/RTP-ISDN and 3G Mobile in Asterisk which allows a 3G phone to call a SIP client via Asterisk. This thesis discusses the implementation of AMR in asterisk as well as SIP protocol and SIP soft phones

Speech quality prediction for voice over Internet protocol networks

Merged with duplicate record 10026.1/878 on 03.01.2017 by CS (TIS). Merged with duplicate record 10026.1/1657 on 15.03.2017 by CS (TIS)This is a digitised version of a thesis that was deposited in the University Library. If you are the author please contact PEARL Admin ([email protected]) to discuss options.IP networks are on a steep slope of innovation that will make them the long-term carrier of all types of traffic, including voice. However, such networks are not designed to support real-time voice communication because their variable characteristics (e.g. due to delay, delay variation and packet loss) lead to a deterioration in voice quality. A major challenge in such networks is how to measure or predict voice quality accurately and efficiently for QoS monitoring and/or control purposes to ensure that technical and commercial requirements are met. Voice quality can be measured using either subjective or objective methods. Subjective measurement (e.g. MOS) is the benchmark for objective methods, but it is slow, time consuming and expensive. Objective measurement can be intrusive or non-intrusive. Intrusive methods (e.g. ITU PESQ) are more accurate, but normally are unsuitable for monitoring live traffic because of the need for a reference data and to utilise the network. This makes non-intrusive methods(e.g. ITU E-model) more attractive for monitoring voice quality from IP network impairments. However, current non-intrusive methods rely on subjective tests to derive model parameters and as a result are limited and do not meet new and emerging applications. The main goal of the project is to develop novel and efficient models for non-intrusive speech quality prediction to overcome the disadvantages of current subjective-based methods and to demonstrate their usefulness in new and emerging VoIP applications. The main contributions of the thesis are fourfold: (1) a detailed understanding of the relationships between voice quality, IP network impairments (e.g. packet loss, jitter and delay) and relevant parameters associated with speech (e.g. codec type, gender and language) is provided. An understanding of the perceptual effects of these key parameters on voice quality is important as it provides a basis for the development of non-intrusive voice quality prediction models. A fundamental investigation of the impact of the parameters on perceived voice quality was carried out using the latest ITU algorithm for perceptual evaluation of speech quality, PESQ, and by exploiting the ITU E-model to obtain an objective measure of voice quality. (2) a new methodology to predict voice quality non-intrusively was developed. The method exploits the intrusive algorithm, PESQ, and a combined PESQ/E-model structure to provide a perceptually accurate prediction of both listening and conversational voice quality non-intrusively. This avoids time-consuming subjective tests and so removes one of the major obstacles in the development of models for voice quality prediction. The method is generic and as such has wide applicability in multimedia applications. Efficient regression-based models and robust artificial neural network-based learning models were developed for predicting voice quality non-intrusively for VoIP applications. (3) three applications of the new models were investigated: voice quality monitoring/prediction for real Internet VoIP traces, perceived quality driven playout buffer optimization and perceived quality driven QoS control. The neural network and regression models were both used to predict voice quality for real Internet VoIP traces based on international links. A new adaptive playout buffer and a perceptual optimization playout buffer algorithms are presented. A QoS control scheme that combines the strengths of rate-adaptive and priority marking control schemes to provide a superior QoS control in terms of measured perceived voice quality is also provided. (4) a new methodology for Internet-based subjective speech quality measurement which allows rapid assessment of voice quality for VoIP applications is proposed and assessed using both objective and traditional MOS test methods

Analysis of Real Time Video Communication Systems

The most of the existing real time video communication systems mainly focus their work on providing better video quality throughout session. In quest of maintaining video quality they succeed in doing so at the cost of broken sessions, blocky video or sound disturbances when network bandwidth drops below required rate. The system described in this paper mainly concentrates on analysis of input parameters to audio and video encoder which affects the quality of communication. The input parameters to video encoder are altered such that a balance is maintained between video quality and continuity in communication. The input parameters to video encoderused for analysis are video frame size, and frames per second and target encode bitrate used for encoding video frame. The input parameters for audio encoder used for analysis are sampling frequency, bits per sample and no of audio channels used for recording sound. The input parameters to video encoderare changed frequently depending upon various factors such as bandwidth variations, and encodetime required on hardware used. In extreme low bandwidth situation the video is stopped. The communication should always keep alive throughout the session by keeping audio session connected always, so that users should not feel disconnected. The other important factors required for real time video communication to work smoothly are transport protocols used to carry media data and control data across peers. The protocols discussed in this paper are Real Time Protocol (RTP) and Real Time Control Protocol (RTCP). The media data generated at peers is transported using RTP and the control data describing the media data is transported using RTCP. DOI: 10.17762/ijritcc2321-8169.15083

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

Secure VoIP Performance Measurement

This project presents a mechanism for instrumentation of secure VoIP calls. The experiments were run under different network conditions and security systems. VoIP services such as Google Talk, Express Talk and Skype were under test. The project allowed analysis of the voice quality of the VoIP services based on the Mean Opinion Score (MOS) values generated by Perceptual valuation of Speech Quality (PESQ). The quality of the audio streams produced were subjected to end-to-end delay, jitter, packet loss and extra processing in the networking hardware and end devices due to Internetworking Layer security or Transport Layer security implementations. The MOS values were mapped to Perceptual Evaluation of Speech Quality for wideband (PESQ-WB) scores. From these PESQ-WB scores, the graphs of the mean of 10 runs and box and whisker plots for each parameter were drawn. Analysis on the graphs was performed in order to deduce the quality of each VoIP service. The E-model was used to predict the network readiness and Common vulnerability Scoring System (CVSS) was used to predict the network vulnerabilities. The project also provided the mechanism to measure the throughput for each test case. The overall performance of each VoIP service was determined by PESQ-WB scores, CVSS scores and the throughput. The experiment demonstrated the relationship among VoIP performance, VoIP security and VoIP service type. The experiment also suggested that, when compared to an unsecure IPIP tunnel, Internetworking Layer security like IPSec ESP or Transport Layer security like OpenVPN TLS would improve a VoIP security by reducing the vulnerabilities of the media part of the VoIP signal. Morever, adding a security layer has little impact on the VoIP voice quality