28 research outputs found

    Considering Bluetooth's Subband Codec (SBC) for Wideband Speech and Audio on the Internet

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    The Bluetooth Special Interest Group (SIG) has standardized the subband coding (SBC) audio codec to connect headphones via wireless Bluetooth links. SBC compresses audio at high fidelity while having an ultra-low algorithm delay. To make SBC suitable for the Internet, we extend it by using a time and packet loss concealment (PLC) algorithm that is based on ITU's G.711 Appendix I. The design is novel in the aspect of the interface between codec and speech receiver. We developed a new approach on how to distribute the functionality of a speech receiver between codec and application. Our approach leads to easier implementations of high quality VoIP applications. We conducted subjective and objective listening tests of the audio quality of SBC and PLC in order to determine an optimal coding mode and the trade-off between coding mode and packet loss rate. More precisely, we conducted MUSHRA listening tests for selected sample items. These tests results are then compared with the results of multiple objective assessment algorithms (ITU P.862 PESQ, ITU BS.1387-1 PEAQ, Creusere's algorithm). We found out that a combination of the PEAQ basic and advanced values best matches---after third order linear regression---the subjective MUSHRA results . The linear regression has coefficient of determination of R²=0.907². By comparison, our individual human ratings show a correlation of about R=0.9 compared to our averaged human rating results. Using the combination of both PEAQ algorithms, we calculate hundred thousands of objective audio quality ratings varying audio content and algorithmic parameters of SBC and PLC. The results show which set of parameters value are best suitable for a bandwidth and delay constrained link. The transmission quality of SBC is enhanced significantly by selecting optimal encoding parameters as compared to the default parameter sets given in the standard. Finally, we present preliminary objective tests results on the comparison of the audio codecs SBC, CELT, APT-X and ULD coding speech and audio transmission. They all allow a mono and stereo transmission of music at ultra-low coding delays (<10ms), which is especially useful for distributed ensemble performances over the Internet

    Improved Performance of Secured VoIP Via Enhanced Blowfish Encryption Algorithm

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    Both the development and the integration of efficient network, open source technology, and Voice over Internet Protocol (VoIP) applications have been increasingly important and gained quick popularity due to new rapidly emerging IP-based network technology. Nonetheless, security and privacy concerns have emerged as issues that need to be addressed. The privacy process ensures that encryption and decryption methods protect the data from being alternate and intercept, a privacy VoIP call will contribute to private and confidential conversation purposes such as telebanking, telepsychiatry, health, safety issues and many more. Hence, this study had quantified VoIP performance and voice quality under security implementation with the technique of IPSec and the enhancement of the Blowfish encryption algorithm. In fact, the primary objective of this study is to improve the performance of Blowfish encryption algorithm. The proposed algorithm was tested with varying network topologies and a variety of audio codecs, which contributed to the impact upon VoIP network. A network testbed with seven experiments and network configurations had been set up in two labs to determine its effects on network performance. Besides, an experimental work using OPNET simulations under 54 experiments of network scenarios were compared with the network testbed for validation and verification purposes. Next, an enhanced Blowfish algorithm for VoIP services had been designed and executed throughout this research. From the stance of VoIP session and services performance, the redesign of the Blowfish algorithm displayed several significant effects that improved both the performance of VoIP network and the quality of voice. This finding indicates some available opportunities that could enhance encrypted algorithm, data privacy, and integrity; where the balance between Quality of Services (QoS) and security techniques can be applied to boost network throughput, performance, and voice quality of existing VoIP services. With that, this study had executed and contributed to a threefold aspect, which refers to the redesign of the Blowfish algorithm that could minimize computational resources. In addition, the VoIP network performance was analysed and compared in terms of end-to-end delay, jitter, packet loss, and finally, sought improvement for voice quality in VoIP services, as well as the effect of the designed enhanced Blowfish algorithm upon voice quality, which had been quantified by using a variety of voice codecs

    Modelling and Simulation of SIP and IAX Sessions

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    Import 03/11/2016My thesis is focused on simulating a functioning model of SIP and IAX and compare these two VoIP protocols. This is done by implementing an Asterisk server onto two virtual machines with Ubuntu operating system where I build a trunking system for each protocol, tested it by calling the peers in both directions, captured the traffic passing through and analysed it with Wireshark. The acquired data is then implemented and presented on a chart form for a better view and comparison of the two parallel protocols.Moje práce je zaměřena na simulaci funkčnosti modelu SIP a IAX a porovnání těchto dvou VoIP protokolů. To je provedeno zavedením Asteriskem serveru na dva virtuální počítaček s operačním systémem Ubuntu, kde je vybudován trunking systém pro každý protokol a to tak, že spojuje volající v obou směrech, zachycuje průchod, a analyzuje pomocí Wireshark. Získaná data jsou pak použita a prezentována ve formě grafů pro lepší přehlednost a srovnání obou paralelních protokolů.440 - Katedra telekomunikační technikydobř

    Měření Triple play služeb v hybridní síti

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    The master's thesis deals with a project regarding the implementation, design and the quality of IPTV, VoIP and Data services within the Triple Play services. In heterostructural networks made up of GEPON and xDSL technologies. Different lengths of the optical and metallic paths were used for the measurements. The first part of the thesis is theoretically analyzed the development and trend of optical and metallic networks. The second part deals with the measurement of typical optical and metallic parameters on the constructed experimental network, where its integrity was tested. Another part of the thesis is the evaluation of Triple play results, regarding the test where the network was variously tasked/burdened with data traffic and evaluated according to defined standards. The last part is concerned with the Optiwave Software simulation environment.Diplomová práce se zabývá návrhem, realizací a kvalitou služeb IPTV, VoIP a Data v rámci Triple play služeb v heterostrukturní sítí tvořené GEPON a xDSL technologiemi. Pro měření byli využity různé délky optické a metalické trasy. První části diplomové práce je teoreticky rozebrán vývoj a trend optických a metalických sítí. Druhá část se zaměřuje na měření typických optických a metalických parametrů na vybudované experimentální síti, kde byla následně testována její integrita. Dalším bodem práce je vyhodnocení výsledků Triple play, kde síť je různě zatěžována datovým provozem a následně vyhodnocována podle definovaných norem. Závěr práce je věnovaný simulačnímu prostředí Optiwave.440 - Katedra telekomunikační technikyvýborn

    An improved medium access control protocol for real-time applications in WLANs and its firmware development

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    The IEEE 802.11 Wireless Local Area Network (WLAN), commonly known as Wi-Fi, has emerged as a popular internet access technology and researchers are continuously working on improvement of the quality of service (QoS) in WLAN by proposing new and efficient schemes. Voice and video over Internet Protocol (VVoIP) applications are becoming very popular in Wi-Fi enabled portable/handheld devices because of recent technological advancements and lower service costs. Different from normal voice and video streaming, these applications demand symmetric throughput for the upstream and downstream. Existing Wi-Fi standards are optimised for generic internet applications and fail to provide symmetric throughput due to traffic bottleneck at access points. Performance analysis and benchmarking is an integral part of WLAN research, and in the majority of the cases, this is done through computer simulation using popular network simulators such as Network Simulator ff 2 (NS-2) or OPNET. While computer simulation is an excellent approach for saving time and money, results generated from computer simulations do not always match practical observations. This is why, for proper assessment of the merits of a proposed system in WLAN, a trial on a practical hardware platform is highly recommended and is often a requirement. In this thesis work, with a view to address the abovementioned challenges for facilitating VoIP and VVoIP services over Wi-Fi, two key contributions are made: i) formulating a suitable medium access control (MAC) protocol to address symmetric traffic scenario and ii) firmware development of this newly devised MAC protocol for real WLAN hardware. The proposed solution shows signifocant improvements over existing standards by supporting higher number of stations with strict QoS criteria. The proposed hardware platform is available off-the-shelf in the market and is a cost effective way of generating and evaluating performance results on a hardware system

    RTP Payload Format for Standard apt-X and Enhanced apt-X Codecs

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    Protocols and Algorithms for Adaptive Multimedia Systems

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    The deployment of WebRTC and telepresence systems is going to start a wide-scale adoption of high quality real-time communication. Delivering high quality video usually corresponds to an increase in required network capacity and also requires an assurance of network stability. A real-time multimedia application that uses the Real-time Transport Protocol (RTP) over UDP needs to implement congestion control since UDP does not implement any such mechanism. This thesis is about enabling congestion control for real-time communication, and deploying it on the public Internet containing a mixture of wired and wireless links. A congestion control algorithm relies on congestion cues, such as RTT and loss. Hence, in this thesis, we first propose a framework for classifying congestion cues. We classify the congestion cues as a combination of: where they are measured or observed? And, how is the sending endpoint notified? For each there are two options, i.e., the cues are either observed and reported by an in-path or by an off-path source, and, the cue is either reported in-band or out-of-band, which results in four combinations. Hence, the framework provides options to look at congestion cues beyond those reported by the receiver. We propose a sender-driven, a receiver-driven and a hybrid congestion control algorithm. The hybrid algorithm relies on both the sender and receiver co-operating to perform congestion control. Lastly, we compare the performance of these different algorithms. We also explore the idea of using capacity notifications from middleboxes (e.g., 3G/LTE base stations) along the path as cues for a congestion control algorithm. Further, we look at the interaction between error-resilience mechanisms and show that FEC can be used in a congestion control algorithm for probing for additional capacity. We propose Multipath RTP (MPRTP), an extension to RTP, which uses multiple paths for either aggregating capacity or for increasing error-resilience. We show that our proposed scheduling algorithm works in diverse scenarios (e.g., 3G and WLAN, 3G and 3G, etc.) with paths with varying latencies. Lastly, we propose a network coverage map service (NCMS), which aggregates throughput measurements from mobile users consuming multimedia services. The NCMS sends notifications to its subscribers about the upcoming network conditions, which take these notifications into account when performing congestion control. In order to test and refine the ideas presented in this thesis, we have implemented most of them in proof-of-concept prototypes, and conducted experiments and simulations to validate our assumptions and gain new insights.

    Enterprise network convergence: path to cost optimization

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    During the past two decades, telecommunications has evolved a great deal. In the eighties, people were using television, radio and telephone as their communication systems. Eventually, the introduction of the Internet and the WWW immensely transformed the telecommunications industry. This internet revolution brought about a huge change in the way businesses communicated and operated. Enterprise networks now had an increasing demand for more bandwidth as they started to embrace newer technologies. The requirements of the enterprise networks grew as the applications and services that were used in the network expanded. This stipulation for fast and high performance communication systems has now led to the emergence of converged network solutions. Enterprises across the globe are investigating new ways to implement voice, video, and data over a single network for various reasons – to optimize network costs, to restructure their communication system, to extend next generation networking abilities, or to bridge the gap between their corporate network and the existing technological progress. To date, organizations had multiple network services to support a range of communication needs. Investing in this type of multiple communication infrastructures limits the networks ability to provide resourceful bandwidth optimization services throughout the system. Thus, as the requirements for the corporate networks to handle dynamic traffic grow day by day, the need for a more effective and efficient network arises. A converged network is the solution for enterprises aspiring to employ advanced applications and innovative services. This thesis will emphasize the importance of converging network infrastructure and prove that it leads to cost savings. It discusses the characteristics, architecture, and relevant protocols of the voice, data and video traffic over both traditional infrastructure and converged architecture. While IP-based networks present excellent quality for non real-time data networking, the network by itself is not capable of providing reliable, quality and secure services for real-time traffic. In order for IP networks to perform reliable and timely transmission of real-time data, additional mechanisms to reduce delay, jitter and packet loss are required. Therefore, this thesis will also discuss the important mechanisms for running real-time traffic like voice and video over an IP network. Lastly, it will also provide an example of an enterprise network specifications (voice, video and data), and present an in depth cost analysis of a typical network vs. a converged network to prove that converged infrastructures provide significant savings
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