In this paper , we augment our earlier study for the user equipment 's ( UE ) energy efficiency ( EE ) in the long-term evolution ( LTE ) downlink by looking at real-time heterogeneous traffic QoS requirements . In particular , we utilize the previously proposed cloud radio access network ( C-RAN ) and ray tracing (RT)-based scheduling model to optimize both of the EE and the packet delay jitter for real-time applications with fixed packet delay budget subject to other traffic types requirements . Using the utility-based scheduling approach , we formulate the resource allocation problem as a weighted sum binary integer programming ( BIP ) problem . There are methods , such as communication cost based service selection strategies , developed to cope with the problem . In this paper , we propose a complementing technology , migratable Web services , to overcome the performance problems for real-time applications . A service can be implemented by multiple light weighted Web services and some of them can be migrated to strategic sites to minimize the communication delays so the real-time requirements can be met . Research in Mobile Ad-hoc Networks ( MANETs ) has been in-progress towards the real implementation of ad-hoc routing protocols to be used in the real-time applications . However , existing approaches implement ad-hoc routing protocols as standalone daemons to serve only pure routing functionalities , leading to difficulties on the network configuration once multiple routing protocols are running in the network . Tool support is also discussed e.g. PRISM for probabilistic model checking and Visual Service Composition Studio for service-oriented modelling . A detailed and specific real-time application scenario is included and modelled to allow the various techniques presented in this document to be demonstrated and quantitatively evaluated . The scenario also reveals the level of detail needed in order for meaningful modelling to be achieved in practice . Color analysis also plays a specifically important role in many other different applications for road sign detection , this paper points to many problems regarding stability of color detection due to daylight conditions , so absence of color model can led a better solution . In this project lightweight techniques were mainly used due to limitation of real- time based application and Raspberry Pi capabilities . Raspberry Pi is the main target for the implementation , as it provides an interface between sensors , database , and image processing results , while also performing functions to manipulate peripheral units ( usb dongle , keyboard etc . ) . Nature inspired computing has become one of the necessities of real time application-based methods . While considering the localization aspects of the wireless sensor network , the significance of optimization techniques plays a crucial role in the precise and accurate position estimation of the sensor node . The simulation results show that the parallel algorithm can perform the PCMA-signal blind separation quickly and effectively . Further , it can greatly decrease the time of the separation , without reducing the performance of the algorithm , and improve the real-time application of system . The independent assumption of the SCTM is relaxed by incorporating the covariance structure calibrated from the spatial correlation analysis for probabilistic traffic state evaluation . For real-time application purposes , prediction is conducted in a rolling horizon manner , which is useful for adjusting the predicted traffic state using real-time measurements . The proposed traffic state prediction framework is validated by empirical studies that demonstrate the effectiveness of the proposed method . However , few works have demonstrated the benefits of using ANC systems in active acoustic windows since their implementation depends on several variables , such as the control source configuration ( quantity and position of speakers ) , the adaptive control algorithm and the computational capabilities of the embedded devices . Specifically , the computational complexity of the adaptive control plays an important role in the development of practical real-timeANC applications . Until date , several works have presented theoretical models for ANC systems . Second , when packet loss occurs , our front-end becomes more effective since it is not constrained to the error handling mechanism of the codec . We have considered the ITU G.723.1 standard codec , which is one of the most preponderant coding algorithms in voice over IP ( VoIP ) and compared the proposed front-end with the conventional approach in two automatic speech recognition ( ASR ) tasks , namely , speaker-independent isolated digit recognition and speaker-independent continuous speech recognition . In general , our approach outperforms the conventional procedure , for a variety of simulated packet loss rates . With the increasing deployment of real-time Internet services , evaluating the user perception of quality of service ( QoS ) has gained rapidly increasing importance . In the case of Voice over IP ( VoIP ) , the standard approach of listening-only tests for subjectively assessing a limited number of speech samples , which are supposed to be representative for selected network conditions , does in no way reflect the huge variability of packet loss patterns that may originate from the underlying network . Performing tests by employing objective ( instrumental ) evaluation methods in a live testbed environment is usually extensive and does not deliver reproducible results , moreover the measurement granularity is bounded by the length of the test speech samples . Delay sensitive applications , such as voice over IP and network games , often sacrifice throughput for lower delay to obtain better quality . Unfortunately , the Internet does not allow an application to choose the amount of delay or throughput it receives and instead packets from all applications receive the same best-effort service . There has been a growing concern to provide quality of service ( QoS ) to this protocol suite , the result of which has been the considerable work towards the formulation of the IEEE 802.11e and IEEE 802.11n versions . One important aspect to consider is performance for real time applications like voice over IP ( VoIP ) . In this paper we focus on performance issues of delay and throughput as a function of packet size , initial contention window size settings and the number of active terminals competing for access to the network , when some terminals provide VoIP services , while others transfer data . The Session Initiation Protocol ( SIP ) is used to establish , maintain and terminate multimedia sessions over the Internet . It is the predominant signalling protocol for Voice over IP ( VoIP ) . Hence understanding security vulnerabilities of SIP is crucial to secure VoIP . WLANs based on the IEEE 802.11 standards enjoy an exceptionally high popularity owing to their simplicity of setup , configuration , and management , increased deployment flexibility , operability in unlicensed frequency bands , low cost , and connectivity with minimal infrastructure changes . Lately , the need for real-time multimedia services over WLANs to support applications such as Voice over IP ( VoIP ) , audio/video ( AV ) streaming , Internet videoconferencing , IPTV , distance learning systems , entertainment and gaming programs has dramatically increased . The impressive proliferation of networking scenarios ( e.g. multihop and mesh networks ) and applications ( e.g. real-time multimedia services ) not natively supported by the original standard has boosted the design of several protocol extensions , which have been proposed and evaluated by the research community , the industry and the standardization groups . The first commercial Internet services emerged in 1989 after the interconnection of the Internet to commercial e-mail services . By 1993 , commercial versions of the World Wide Web had appeared , and by 2003 , voice over IP service was growing rapidly , after its first commercial introduction around 1995 ( See Vocaltec : http://www.vocaltec.com/html/about/company.shtml ) . The Internet of the future will be shaped by the tectonic forces of regulation , commercialization , technological change , and a wide range of policy concerns expressed at local , national , regional and international levels . This study uses a gyro sensor attached to around ear in order to detect head movement of the subject during the conversation . The detected signal by the sensor was used to control the remote robotic arm , which holds the smart phone for Voice over IP communication , and to mimic the head movement of the subject in a remote place . As a result , presence of a remote subject was significantly recognized in the physically active movement of the phone , which was quite different in comparison with the fixed position of the phone . Voice over IP communication will dominate the world . However , given the growing demand for voice and data communication to make any and all communication reliable and secure , several attacks occur frequently in communication networks , so this work is based on verifying security , analyzing risks , vulnerabilities , such as verifying the attacks and proposing a security measure for voice over IP communication on embedded The networking environment allowing for coexistence of voice and data communication has become complex , and voice aspects of telecommunication , particularly in voice-over-IP networks , demand echo cancellers to cover all voice channels , as opposed to only long-haul channels , as it used to be in traditional public switched telephone networks . Echo path coverage requirements for the echo cancellers have become more demanding , which contributed to an increase of computational cost of their implementations . Conditions for coexistence of voice and data communication have become complex , and voice aspects of telecommunication , particularly in Voice-over-IP networks , demand echo cancellers to cover all voice channels , as opposed to only long-haul channels , as it used to be in traditional public switched telephone networks . Echo path coverage requirements for the echo cancellers have become more demanding , causing an increase of computational cost of their implementations . Summary form only given . As the Internet is becoming more robust with the addition of new services such as audio and video streaming , e-commerce , and news casting , the traditional Web server architecture is unable to keep up with the increasing demands and requisites of such new services . This situation has created a need for secure , reliable , highly available , and scalable Web servers that rely on clustering technologies to be able to meet the growth in the user base and offered services . Based on the finding result , the VPN measurement through hardware device is able to contribute low percentage of CPU and memory usage compare to open source application . Key words — VPN , CPU , NMS , Memory , Audio and Video Streaming Real-time applications such as Voice over IP , audio and video streaming require quality of service ( QoS ) . Such applications are being executed over the public Internet . Mobile Worldwide Interoperability for Microwave Access ( WiMAX ) includes a multimedia multicast/broadcast service ( MBS ) , but delay-sensitive applications such as video and audio streaming require the combination of efficient handling of wireless-link bandwidth and reduced handover delays , which remains a challenge . To reduce the handover delay in the MBS , the IEEE 802.16e standard introduces an MBS zone , which is a group of base stations that are broadcasting the same multicast packets . We used rigorous controlled experiments and statistical analysis of the performance of popular online services to identify traffic differentiation . The observed policies include throttling bandwidth available to video and audio streaming , transcoding video , and selectively zero-rating traffic such as video and music streaming . Such policies may violate the “ The different characteristics of each wireless technology with regard to Quality of Service ( QoS ) brought many challenges for provisioning the continuous services ( audio/video streaming ) in a seamless way . The paper proposes Context-Aware Management Scheme ( CAMS ) which facilitates service continuity in real-time application such as video and audio streaming . Unlike previous research efforts which focuses on session continuity only while handover and does not account for the characteristics differences between the networks , this work focuses on improving QoS and user experience by adapting the session parameters after Vertical Handover ( VHO ) according to the new environment and available resources . This is adequate for traditional data applications like e-mail , web browsing , File Transfer Protocol ( FTP ) and Telnet , but inadequate for applications requiring timeliness . For example , multimedia conferencing or audio and video streaming applications , which require high bandwidth capacity and are sensitive to delay and delay variation . For these applications to perform adequately , Quality of Services ( QoS ) must be quantified and managed , and the Internet must be modified to support real-time QoS and controlled end-to-end delays . While TCP-based applications react to network congestion , UDP-based streaming applications usually do not have any type of flow and congestion control mechanisms . UDP , the transport layer protocol used by audio and video streaming applications , does not react to network congestion thus stealing bandwidth from the responsive TCP-based connections . Several solutions have been suggested to combat this TCP-friendliness problem , mostly applied either in the end systems ( end-to-end ) at the transport layer of the protocol stack or inside the network ( routers ) at the network layer . Quality of Experience ( QoE ) has gained enormous attention during the recent years . So far , most of the existing QoE research has focused on audio and video streaming applications , although HTTP traffic carries the majority of traffic in the residential broadband Internet . However , existing QoE models for this domain do not consider temporal dynamics or historical experiences of the user 's satisfaction while consuming a certain service . A method and an apparatus for guaranteeing QoS of data communication are provided to guarantee QoS(Quality of Service ) of real-time data transmission such as audio or video streaming . A packet including network information of the second network is generated . This paper brings forward a plan for realizing audio and video streaming transmission based on IP network , and analyses its correlative technologies . It elaborates the process to realize this plan , and introduces the application of audio and video streaming transmission to network . This paper proposes a technology for telemedicine consultation H.264/AVC system construction , the use of a new generation of streaming media technology to improve the existing poor quality , consumption of bandwidth and large file size disadvantage , the user through the System to complete diagnosis of high quality live video , remote minor surgery and high-definition video and audio guidance to clinical teaching . Therefore , by H.264/AVC video streaming media technology and the technical streaming media server ( stream media server ) to explore , to create a widely used high-definition video and audio streaming system architecture , system uses high-definition camera ( HDV ) , network cameras , digital cameras , digital camera or cell phone cameras and other photographic equipment , establish a comprehensive long-distance medical consultation system , while improving audio and video streaming transmission over the Internet fluency . This system not only develop a wider variety of applications in the future can also come in a wider area of use , such as multi-network video conferencing , Internet video teaching , nursing home care , internet video chat room and High-definition digital home networking applications , can be efficiently used in synchronous and asynchronous networks of VOD ( video on demand ) and data base sharing ( data on demand ) , video quality is not resolved Good , smooth video streaming and image is not recognizable and poor . While traditional live TV is broadcast from a central server deeper in the network , video-on-demand ( VoD ) servers are typically located at the edge of the core network . In order to support interactivity from the end user for live TV or to serve requests for other very popular content , servers in the access network can become beneficial . This approach , however , has important implications for future access network architectures , as discussed further on in this article . This paper deals with AES-CTR mode . AES algorithms are more complex and require larger time for execution which is not suitable for real time application like Live TV , Video chatting or Video conferencing on handheld device . The proposed system aims to gain a deep understanding of video data security on multimedia technologies and to provide security for real time video applications using selective encryption for H.264/AVC . A general concept for providing physical separation of data streams together with a threat model is presented . The main target is delay-sensitive applications such as telephony signaling , live TV , and radio broadcasts that require only lightweight security . The threat considered is malicious interception of network transfers through so-called eavesdropping attacks . The closed captioning was prepared for CT Sport , the sport channel of the public service broadcaster in the Czech Republic . We will briefly discuss our solution for distributed captioning architecture on live TV programs using re-speaking approach as well as several modifications of existing live captioning application ( especially LVCSR system ) , but also the way of re-speaking of a real TV commentary for individual sports . We will show that a re-speaker after hard training can achieve such accuracy ( more than 98 % ) and readability of captions which clearly outperform accuracy of captions created by automatic recognition of TV soundtrack . In order to satisfy the individual needs of consumers for recording TV programs in some different ways , an improved scheme is presented to implement Personal Video Recorder ( PVR ) function in Android set-top box ( STB ) . As for live TV programs , there are three alternative ways which are continuous recording , timer recording and time-shifting recording being designed . As for TV programs in the future , the way of scheduled recording is designed . In current Chinese TV market , Live TV programs are buffeted by internet contents from smart TV/over-the-top ( OTT ) box and other screens like mobile phone and pad . The user experience of Live TV watching ca n't satisfy user needs . Live TV production , due to its distributed nature , requires broadcasters to deploy equipments and human resources to several different places . This increases production costs . Cloud computing is a new infrastructure environment that delivers on the promise of supporting on-demand services in a flexible manner by scheduling bandwidth , storage and compute resources on the fly . IPTV services like Video On Demand ( VoD ) and Live broadcast TV requires substantial bandwidth and compute resources to meet the real time requirements and to handle the very bursty resource requirements for each of these services . To meet the needs of the bursts of requests , each with a deadline constraint for both VoD and LiveTV channel changes , we propose a resource provisioning framework that allows these services to co-exist on a common infrastructure by taking advantage of virtualization . Cloud computing is a new infrastructure environment that delivers on the promise of supporting ondemand services in a flexible manner by scheduling bandwidth , storage and compute resources on the fly . IPTelevision services like Video On Demand ( VoD ) and Live broadcast TV requires substantial bandwidth and compute resources to meet the real time requirements and to handle the very bursty resource requirements for each of these services and need to allow these IPTV services to co-exist on a common infrastructure by taking advantage of virtualization . One of the biggest challenge that service providers face today is IPbased video delivery has dramatically increased the demands placed on service provider resources , service provider provision their resources for peak demands of each service across the subscriber population . Cloud computing is a new infrastructure environment that delivers on the promise of supporting on-demand services in a flexible manner by scheduling bandwidth , storage and compute resources on the fly . IPTV services like Video on Demand ( VoD ) and Live broadcast TV require substantial bandwidth and compute resources to meet the real time requirements and to handle the very bursty resource requirements for each of these services . To meet the needs of the bursts of requests , each with a deadline constraint for both VoD and Live TV channel changes , we propose a resource provisioning framework that allows these services to co-exist on a common infrastructure by taking advantage of virtualization . The approximate algorithm was able to compute the burstiness curve of a 2-h long elementary video stream in approximately 10 s , as compared to over 6 h for the exact algorithm , with virtually no loss of accuracy in the computation . The efficiency of the proposed algorithms makes them suitable for quality-of-service ( QoS ) provisioning not only in off-line environments such as in video-on-demand ( VoD ) servers , but also in real-time applications such as in live TV distribution systems . A hierarchical distributed telecom cloud architecture for live-TV distribution exploiting flexgrid networking and SBVTs is proposed . Its scalability is compared to that of a centralized architecture . In this paper we investigate the role of social media as the facilitator of second screen for TV , through the analysis of tweets for weekend primetime UK TV show the X-Factor . The results highlight the rich source of information that can be extracted in real-time and its enormous potential for broadcasters and producers both in terms of reinvigorating live TV viewing and creating new forms of audience interaction . The results highlight the rich source of information that can be extracted in real-time and how audiences are creating their own parallel narrative of the show through Twitter . This interaction has enormous potential for broadcasters and producers both in terms of reinvigorating live TV viewing and creating new forms of audience interaction . In this paper we investigate the role of social media as the facilitator of second screen for TV , through the analysis of tweets for weekend primetime UK TV show the X-Factor . The results highlight the rich source of information that can be extracted in real-time and its enormous potential for broadcasters and producers both in terms of reinvigorating live TV viewing and creating new forms of audience interaction . Distribution over the internet is destined to become a stand ard approach for live broadcasting of TV or events of nation-wide interest . The demand f or high-quality live video with personal requirements is destined to grow exponential ly over the next few years . The resulting model is used in the framework of a programmable and active network based management platform associated with a DVPN service . The particular DVPN service is used in the context of live TV broadcast to the subscribed residential users . iTVP is a system built for delivery of live TV programming , video-on-demand and audio-on-demand with interactive access over IP networks . It has a nationwide range and is designed to provide service to a high number of concurrent users . Wide deployment of USN/RFID will generate an unprecedented volume of primitive data in a short time . Duplication and redundancy of primitive data will affect real time performance of application . Thus , security applications must filter primitive data and correlate them for complex pattern detection and transform them to events that provide meaningful , actionable information to end application . The implemented design includes filters of different functions and character generator with keyboard interface . The real-time of application like stock ticker and time display are also demonstrated . The VHDL codes for the architecture were synthesized using Xilinx ISE 10.1 and targeted for Xilinx Spartan3E FPGA . The problem to be considered is specification and synthesis of real-time inter-task communication interface among software-to-software , software-to-hardware , and hardware-to-hardware tasks . Third , we also consider the functionality required for real-time processing of application tasks and protocol functions . What it is need to be studied is the question of how to design a data frame unit with real-time features , to provide the inter-task communication channel which concurrently carries messages between software and hardware elements , and to explain how both fine-grained and coarse-grained real-time processing . At the same time , it is easier to realize than the LSE . It is quite potential for real-time requirement of application . One of the paper objectives is to compare the experimental results of MOS values with calculated values of parameter R. Therefore , the measured values are compared and incorporated in adjusted Emodel . This simplified model is presented in framework of constructing the model for VoIP communication optimisation based on simple measurement information as a base for the proposal of agent architecture for optimisation of VoIP communication quality . VoIP is an example of an application which is subject to performance impairments due to these characteristics . VoIP trials over existing broadband satellite offers reveal that the application can be carried out with an acceptable quality . However results also reveal the need for further improvement of the bandwidth on demand scheme in order to reduce delay or improve bandwidth utilization . This is important , as satellites can play a role in broadband service provision by addressing the cases of limited terrestrial access such as in rural or underserved areas . The first part of this paper presents the results of our VoIP trials with different commercial DVB-S/RCS satellite offers and popular internet telephony and videoconferencing applications ( Skype , MSN , etc . ) . These results reveal that packet delay and jitter are strongly affected by the satellite network component as well as the type of speech codec used . One of the most parts of VOIP management is fault management and , in having a good fault management , finding good mechanisms to detect faults in the network have to be considered . The main focus of this project is to implement different types of fast fault recovery protocols in networks , especially networks that carry IP telephony . On the other hand , complete messages do not provide information about packet level behavior , which is necessary for internetworking layer monitoring . Paradoxically , the Network Management community has not paid much attention to this fact , although it compromises several VoIP management tasks . To fill in this gap , we provide an empirical evaluation of its impact for the most popular VoIP signaling protocols using traces from enterprise networks , and present the architecture and heuristic that we are currently developing to partially solve the effect of the segmentation . Finding digital evidence in VoIP malicious attacks is the most difficult task , due to its associated features with converged network . In this paper , a Model of investigating VoIP malicious attacks is proposed for forensic analysis . The model formalizes hypotheses through information gathering and adopt a Secure Temporal Logic of Action(S-TLA+ ) in the process of reconstructing potential attack scenario . The first is based on the traditional centralized IN service architecture . The second proposes a new decentralized architecture based on direct control of VoIP call control equipment by open service architecture interfaces . VOIP technical Provides regular phone services , as well as other services such as the ability to create a multilateral conversation , voice mail , and the ability to customize ringtones and add details of people their caller and saved , was also able to send and receive data such as images and documents during a voice conversation . The research aims to assess the quality of the sound VOIP transmission over a private UMTS virtual network made up of two networks subgroups in two different locations are within about each other a distance of 80 Km , by examining VoIP scenarios using IP within the UMTS network used to transfer data through the delay and loss using simulation OPNET measurement program which It provides an environment perfectly mimic " the reality of the studied network , has enabled the results that have been reached in addition to evaluating the quality of voice transmission determine the width required service package , the type of audio encoding to be used . In this paper , VoIP ( All-IP type , and for Gateway type ) for system and network scanning , and , IP Phone to get the information and analysis of the vulnerability . All-IP type and Gateway type discovered about the vulnerability of VoIP hacking attacks ( Denial of Service attacks , VoIP spam attacks ) is carried out . And that is a real VoIP system installed and operated in the field of security measures through research and analysis is proposed . Because virtually everyone has access to the Internet , threats due to insecurity and anonymity reach critical levels and have to be tackled by both carriers and Internet Service Providers . Regaining trust-by-wire , comparable to classic fixed line telephones , could mitigate or even solve problems like Spam , Phishing , and the localization of VoIP emergency calls . This paper presents the hardware implementation of a new and highly flexible solution-Internet protocol-calling line identification presentation-which provides additional support for new services to restore peoplepsilas confidence into the Internet . During the past 5 years , domestic VoIP service has grown rapidly absorbing more than 8 million subscribers . More recently , the increase of enterprise or institutional subscribers become more dominant than individual subscribers . Thus it is effective method to identify IP and MAC spoofed attacks . I will construct the secured VOIP communication over MANET area using TCP / IP and storage Data in GSM modem as SMS ( inbox ) . Finally using IP and MAC get fail to prevent spoofing attack then DoS attack prevent using BIOS serial number of clients .
In this paper , we augment our earlier study for the user equipment 's ( UE ) energy efficiency ( EE ) in the long-term evolution ( LTE ) downlink by looking at real-time heterogeneous traffic QoS requirements . In particular , we utilize the previously proposed cloud radio access network ( C-RAN ) and ray tracing (RT)-based scheduling model to optimize both of the EE and the packet delay jitter for real-time applications with fixed packet delay budget subject to other traffic types requirements . Using the utility-based scheduling approach , we formulate the resource allocation problem as a weighted sum binary integer programming ( BIP ) problem .
There are methods , such as communication cost based service selection strategies , developed to cope with the problem . In this paper , we propose a complementing technology , migratable Web services , to overcome the performance problems for real-time applications . A service can be implemented by multiple light weighted Web services and some of them can be migrated to strategic sites to minimize the communication delays so the real-time requirements can be met .
Research in Mobile Ad-hoc Networks ( MANETs ) has been in-progress towards the real implementation of ad-hoc routing protocols to be used in the real-time applications . However , existing approaches implement ad-hoc routing protocols as standalone daemons to serve only pure routing functionalities , leading to difficulties on the network configuration once multiple routing protocols are running in the network .
Tool support is also discussed e.g. PRISM for probabilistic model checking and Visual Service Composition Studio for service-oriented modelling . A detailed and specific real-time application scenario is included and modelled to allow the various techniques presented in this document to be demonstrated and quantitatively evaluated . The scenario also reveals the level of detail needed in order for meaningful modelling to be achieved in practice .
Color analysis also plays a specifically important role in many other different applications for road sign detection , this paper points to many problems regarding stability of color detection due to daylight conditions , so absence of color model can led a better solution . In this project lightweight techniques were mainly used due to limitation of real- time based application and Raspberry Pi capabilities . Raspberry Pi is the main target for the implementation , as it provides an interface between sensors , database , and image processing results , while also performing functions to manipulate peripheral units ( usb dongle , keyboard etc . ) .
Nature inspired computing has become one of the necessities of real time application-based methods . While considering the localization aspects of the wireless sensor network , the significance of optimization techniques plays a crucial role in the precise and accurate position estimation of the sensor node .
The simulation results show that the parallel algorithm can perform the PCMA-signal blind separation quickly and effectively . Further , it can greatly decrease the time of the separation , without reducing the performance of the algorithm , and improve the real-time application of system .
The independent assumption of the SCTM is relaxed by incorporating the covariance structure calibrated from the spatial correlation analysis for probabilistic traffic state evaluation . For real-time application purposes , prediction is conducted in a rolling horizon manner , which is useful for adjusting the predicted traffic state using real-time measurements . The proposed traffic state prediction framework is validated by empirical studies that demonstrate the effectiveness of the proposed method .
However , few works have demonstrated the benefits of using ANC systems in active acoustic windows since their implementation depends on several variables , such as the control source configuration ( quantity and position of speakers ) , the adaptive control algorithm and the computational capabilities of the embedded devices . Specifically , the computational complexity of the adaptive control plays an important role in the development of practical real-timeANC applications . Until date , several works have presented theoretical models for ANC systems .
Second , when packet loss occurs , our front-end becomes more effective since it is not constrained to the error handling mechanism of the codec . We have considered the ITU G.723.1 standard codec , which is one of the most preponderant coding algorithms in voice over IP ( VoIP ) and compared the proposed front-end with the conventional approach in two automatic speech recognition ( ASR ) tasks , namely , speaker-independent isolated digit recognition and speaker-independent continuous speech recognition . In general , our approach outperforms the conventional procedure , for a variety of simulated packet loss rates .
With the increasing deployment of real-time Internet services , evaluating the user perception of quality of service ( QoS ) has gained rapidly increasing importance . In the case of Voice over IP ( VoIP ) , the standard approach of listening-only tests for subjectively assessing a limited number of speech samples , which are supposed to be representative for selected network conditions , does in no way reflect the huge variability of packet loss patterns that may originate from the underlying network . Performing tests by employing objective ( instrumental ) evaluation methods in a live testbed environment is usually extensive and does not deliver reproducible results , moreover the measurement granularity is bounded by the length of the test speech samples .
Delay sensitive applications , such as voice over IP and network games , often sacrifice throughput for lower delay to obtain better quality . Unfortunately , the Internet does not allow an application to choose the amount of delay or throughput it receives and instead packets from all applications receive the same best-effort service .
There has been a growing concern to provide quality of service ( QoS ) to this protocol suite , the result of which has been the considerable work towards the formulation of the IEEE 802.11e and IEEE 802.11n versions . One important aspect to consider is performance for real time applications like voice over IP ( VoIP ) . In this paper we focus on performance issues of delay and throughput as a function of packet size , initial contention window size settings and the number of active terminals competing for access to the network , when some terminals provide VoIP services , while others transfer data .
The Session Initiation Protocol ( SIP ) is used to establish , maintain and terminate multimedia sessions over the Internet . It is the predominant signalling protocol for Voice over IP ( VoIP ) . Hence understanding security vulnerabilities of SIP is crucial to secure VoIP .
WLANs based on the IEEE 802.11 standards enjoy an exceptionally high popularity owing to their simplicity of setup , configuration , and management , increased deployment flexibility , operability in unlicensed frequency bands , low cost , and connectivity with minimal infrastructure changes . Lately , the need for real-time multimedia services over WLANs to support applications such as Voice over IP ( VoIP ) , audio/video ( AV ) streaming , Internet videoconferencing , IPTV , distance learning systems , entertainment and gaming programs has dramatically increased . The impressive proliferation of networking scenarios ( e.g. multihop and mesh networks ) and applications ( e.g. real-time multimedia services ) not natively supported by the original standard has boosted the design of several protocol extensions , which have been proposed and evaluated by the research community , the industry and the standardization groups .
The first commercial Internet services emerged in 1989 after the interconnection of the Internet to commercial e-mail services . By 1993 , commercial versions of the World Wide Web had appeared , and by 2003 , voice over IP service was growing rapidly , after its first commercial introduction around 1995 ( See Vocaltec : http://www.vocaltec.com/html/about/company.shtml ) . The Internet of the future will be shaped by the tectonic forces of regulation , commercialization , technological change , and a wide range of policy concerns expressed at local , national , regional and international levels .
This study uses a gyro sensor attached to around ear in order to detect head movement of the subject during the conversation . The detected signal by the sensor was used to control the remote robotic arm , which holds the smart phone for Voice over IP communication , and to mimic the head movement of the subject in a remote place . As a result , presence of a remote subject was significantly recognized in the physically active movement of the phone , which was quite different in comparison with the fixed position of the phone .
Voice over IP communication will dominate the world . However , given the growing demand for voice and data communication to make any and all communication reliable and secure , several attacks occur frequently in communication networks , so this work is based on verifying security , analyzing risks , vulnerabilities , such as verifying the attacks and proposing a security measure for voice over IP communication on embedded
The networking environment allowing for coexistence of voice and data communication has become complex , and voice aspects of telecommunication , particularly in voice-over-IP networks , demand echo cancellers to cover all voice channels , as opposed to only long-haul channels , as it used to be in traditional public switched telephone networks . Echo path coverage requirements for the echo cancellers have become more demanding , which contributed to an increase of computational cost of their implementations .
Conditions for coexistence of voice and data communication have become complex , and voice aspects of telecommunication , particularly in Voice-over-IP networks , demand echo cancellers to cover all voice channels , as opposed to only long-haul channels , as it used to be in traditional public switched telephone networks . Echo path coverage requirements for the echo cancellers have become more demanding , causing an increase of computational cost of their implementations .
Summary form only given . As the Internet is becoming more robust with the addition of new services such as audio and video streaming , e-commerce , and news casting , the traditional Web server architecture is unable to keep up with the increasing demands and requisites of such new services . This situation has created a need for secure , reliable , highly available , and scalable Web servers that rely on clustering technologies to be able to meet the growth in the user base and offered services .
Based on the finding result , the VPN measurement through hardware device is able to contribute low percentage of CPU and memory usage compare to open source application . Key words — VPN , CPU , NMS , Memory , Audio and Video Streaming
Real-time applications such as Voice over IP , audio and video streaming require quality of service ( QoS ) . Such applications are being executed over the public Internet .
Mobile Worldwide Interoperability for Microwave Access ( WiMAX ) includes a multimedia multicast/broadcast service ( MBS ) , but delay-sensitive applications such as video and audio streaming require the combination of efficient handling of wireless-link bandwidth and reduced handover delays , which remains a challenge . To reduce the handover delay in the MBS , the IEEE 802.16e standard introduces an MBS zone , which is a group of base stations that are broadcasting the same multicast packets .
We used rigorous controlled experiments and statistical analysis of the performance of popular online services to identify traffic differentiation . The observed policies include throttling bandwidth available to video and audio streaming , transcoding video , and selectively zero-rating traffic such as video and music streaming . Such policies may violate the “
The different characteristics of each wireless technology with regard to Quality of Service ( QoS ) brought many challenges for provisioning the continuous services ( audio/video streaming ) in a seamless way . The paper proposes Context-Aware Management Scheme ( CAMS ) which facilitates service continuity in real-time application such as video and audio streaming . Unlike previous research efforts which focuses on session continuity only while handover and does not account for the characteristics differences between the networks , this work focuses on improving QoS and user experience by adapting the session parameters after Vertical Handover ( VHO ) according to the new environment and available resources .
This is adequate for traditional data applications like e-mail , web browsing , File Transfer Protocol ( FTP ) and Telnet , but inadequate for applications requiring timeliness . For example , multimedia conferencing or audio and video streaming applications , which require high bandwidth capacity and are sensitive to delay and delay variation . For these applications to perform adequately , Quality of Services ( QoS ) must be quantified and managed , and the Internet must be modified to support real-time QoS and controlled end-to-end delays .
While TCP-based applications react to network congestion , UDP-based streaming applications usually do not have any type of flow and congestion control mechanisms . UDP , the transport layer protocol used by audio and video streaming applications , does not react to network congestion thus stealing bandwidth from the responsive TCP-based connections . Several solutions have been suggested to combat this TCP-friendliness problem , mostly applied either in the end systems ( end-to-end ) at the transport layer of the protocol stack or inside the network ( routers ) at the network layer .
Quality of Experience ( QoE ) has gained enormous attention during the recent years . So far , most of the existing QoE research has focused on audio and video streaming applications , although HTTP traffic carries the majority of traffic in the residential broadband Internet . However , existing QoE models for this domain do not consider temporal dynamics or historical experiences of the user 's satisfaction while consuming a certain service .
A method and an apparatus for guaranteeing QoS of data communication are provided to guarantee QoS(Quality of Service ) of real-time data transmission such as audio or video streaming . A packet including network information of the second network is generated .
This paper brings forward a plan for realizing audio and video streaming transmission based on IP network , and analyses its correlative technologies . It elaborates the process to realize this plan , and introduces the application of audio and video streaming transmission to network .
This paper proposes a technology for telemedicine consultation H.264/AVC system construction , the use of a new generation of streaming media technology to improve the existing poor quality , consumption of bandwidth and large file size disadvantage , the user through the System to complete diagnosis of high quality live video , remote minor surgery and high-definition video and audio guidance to clinical teaching . Therefore , by H.264/AVC video streaming media technology and the technical streaming media server ( stream media server ) to explore , to create a widely used high-definition video and audio streaming system architecture , system uses high-definition camera ( HDV ) , network cameras , digital cameras , digital camera or cell phone cameras and other photographic equipment , establish a comprehensive long-distance medical consultation system , while improving audio and video streaming transmission over the Internet fluency . This system not only develop a wider variety of applications in the future can also come in a wider area of use , such as multi-network video conferencing , Internet video teaching , nursing home care , internet video chat room and High-definition digital home networking applications , can be efficiently used in synchronous and asynchronous networks of VOD ( video on demand ) and data base sharing ( data on demand ) , video quality is not resolved Good , smooth video streaming and image is not recognizable and poor .
While traditional live TV is broadcast from a central server deeper in the network , video-on-demand ( VoD ) servers are typically located at the edge of the core network . In order to support interactivity from the end user for live TV or to serve requests for other very popular content , servers in the access network can become beneficial . This approach , however , has important implications for future access network architectures , as discussed further on in this article .
This paper deals with AES-CTR mode . AES algorithms are more complex and require larger time for execution which is not suitable for real time application like Live TV , Video chatting or Video conferencing on handheld device . The proposed system aims to gain a deep understanding of video data security on multimedia technologies and to provide security for real time video applications using selective encryption for H.264/AVC .
A general concept for providing physical separation of data streams together with a threat model is presented . The main target is delay-sensitive applications such as telephony signaling , live TV , and radio broadcasts that require only lightweight security . The threat considered is malicious interception of network transfers through so-called eavesdropping attacks .
The closed captioning was prepared for CT Sport , the sport channel of the public service broadcaster in the Czech Republic . We will briefly discuss our solution for distributed captioning architecture on live TV programs using re-speaking approach as well as several modifications of existing live captioning application ( especially LVCSR system ) , but also the way of re-speaking of a real TV commentary for individual sports . We will show that a re-speaker after hard training can achieve such accuracy ( more than 98 % ) and readability of captions which clearly outperform accuracy of captions created by automatic recognition of TV soundtrack .
In order to satisfy the individual needs of consumers for recording TV programs in some different ways , an improved scheme is presented to implement Personal Video Recorder ( PVR ) function in Android set-top box ( STB ) . As for live TV programs , there are three alternative ways which are continuous recording , timer recording and time-shifting recording being designed . As for TV programs in the future , the way of scheduled recording is designed .
In current Chinese TV market , Live TV programs are buffeted by internet contents from smart TV/over-the-top ( OTT ) box and other screens like mobile phone and pad . The user experience of Live TV watching ca n't satisfy user needs .
Live TV production , due to its distributed nature , requires broadcasters to deploy equipments and human resources to several different places . This increases production costs .
Cloud computing is a new infrastructure environment that delivers on the promise of supporting on-demand services in a flexible manner by scheduling bandwidth , storage and compute resources on the fly . IPTV services like Video On Demand ( VoD ) and Live broadcast TV requires substantial bandwidth and compute resources to meet the real time requirements and to handle the very bursty resource requirements for each of these services . To meet the needs of the bursts of requests , each with a deadline constraint for both VoD and LiveTV channel changes , we propose a resource provisioning framework that allows these services to co-exist on a common infrastructure by taking advantage of virtualization .
Cloud computing is a new infrastructure environment that delivers on the promise of supporting ondemand services in a flexible manner by scheduling bandwidth , storage and compute resources on the fly . IPTelevision services like Video On Demand ( VoD ) and Live broadcast TV requires substantial bandwidth and compute resources to meet the real time requirements and to handle the very bursty resource requirements for each of these services and need to allow these IPTV services to co-exist on a common infrastructure by taking advantage of virtualization . One of the biggest challenge that service providers face today is IPbased video delivery has dramatically increased the demands placed on service provider resources , service provider provision their resources for peak demands of each service across the subscriber population .
Cloud computing is a new infrastructure environment that delivers on the promise of supporting on-demand services in a flexible manner by scheduling bandwidth , storage and compute resources on the fly . IPTV services like Video on Demand ( VoD ) and Live broadcast TV require substantial bandwidth and compute resources to meet the real time requirements and to handle the very bursty resource requirements for each of these services . To meet the needs of the bursts of requests , each with a deadline constraint for both VoD and Live TV channel changes , we propose a resource provisioning framework that allows these services to co-exist on a common infrastructure by taking advantage of virtualization .
The approximate algorithm was able to compute the burstiness curve of a 2-h long elementary video stream in approximately 10 s , as compared to over 6 h for the exact algorithm , with virtually no loss of accuracy in the computation . The efficiency of the proposed algorithms makes them suitable for quality-of-service ( QoS ) provisioning not only in off-line environments such as in video-on-demand ( VoD ) servers , but also in real-time applications such as in live TV distribution systems .
A hierarchical distributed telecom cloud architecture for live-TV distribution exploiting flexgrid networking and SBVTs is proposed . Its scalability is compared to that of a centralized architecture .
In this paper we investigate the role of social media as the facilitator of second screen for TV , through the analysis of tweets for weekend primetime UK TV show the X-Factor . The results highlight the rich source of information that can be extracted in real-time and its enormous potential for broadcasters and producers both in terms of reinvigorating live TV viewing and creating new forms of audience interaction .
The results highlight the rich source of information that can be extracted in real-time and how audiences are creating their own parallel narrative of the show through Twitter . This interaction has enormous potential for broadcasters and producers both in terms of reinvigorating live TV viewing and creating new forms of audience interaction .
In this paper we investigate the role of social media as the facilitator of second screen for TV , through the analysis of tweets for weekend primetime UK TV show the X-Factor . The results highlight the rich source of information that can be extracted in real-time and its enormous potential for broadcasters and producers both in terms of reinvigorating live TV viewing and creating new forms of audience interaction .
Distribution over the internet is destined to become a stand ard approach for live broadcasting of TV or events of nation-wide interest . The demand f or high-quality live video with personal requirements is destined to grow exponential ly over the next few years .
The resulting model is used in the framework of a programmable and active network based management platform associated with a DVPN service . The particular DVPN service is used in the context of live TV broadcast to the subscribed residential users .
iTVP is a system built for delivery of live TV programming , video-on-demand and audio-on-demand with interactive access over IP networks . It has a nationwide range and is designed to provide service to a high number of concurrent users .
Wide deployment of USN/RFID will generate an unprecedented volume of primitive data in a short time . Duplication and redundancy of primitive data will affect real time performance of application . Thus , security applications must filter primitive data and correlate them for complex pattern detection and transform them to events that provide meaningful , actionable information to end application .
The implemented design includes filters of different functions and character generator with keyboard interface . The real-time of application like stock ticker and time display are also demonstrated . The VHDL codes for the architecture were synthesized using Xilinx ISE 10.1 and targeted for Xilinx Spartan3E FPGA .
The problem to be considered is specification and synthesis of real-time inter-task communication interface among software-to-software , software-to-hardware , and hardware-to-hardware tasks . Third , we also consider the functionality required for real-time processing of application tasks and protocol functions . What it is need to be studied is the question of how to design a data frame unit with real-time features , to provide the inter-task communication channel which concurrently carries messages between software and hardware elements , and to explain how both fine-grained and coarse-grained real-time processing .
At the same time , it is easier to realize than the LSE . It is quite potential for real-time requirement of application .
One of the paper objectives is to compare the experimental results of MOS values with calculated values of parameter R. Therefore , the measured values are compared and incorporated in adjusted Emodel . This simplified model is presented in framework of constructing the model for VoIP communication optimisation based on simple measurement information as a base for the proposal of agent architecture for optimisation of VoIP communication quality .
VoIP is an example of an application which is subject to performance impairments due to these characteristics . VoIP trials over existing broadband satellite offers reveal that the application can be carried out with an acceptable quality . However results also reveal the need for further improvement of the bandwidth on demand scheme in order to reduce delay or improve bandwidth utilization .
This is important , as satellites can play a role in broadband service provision by addressing the cases of limited terrestrial access such as in rural or underserved areas . The first part of this paper presents the results of our VoIP trials with different commercial DVB-S/RCS satellite offers and popular internet telephony and videoconferencing applications ( Skype , MSN , etc . ) . These results reveal that packet delay and jitter are strongly affected by the satellite network component as well as the type of speech codec used .
One of the most parts of VOIP management is fault management and , in having a good fault management , finding good mechanisms to detect faults in the network have to be considered . The main focus of this project is to implement different types of fast fault recovery protocols in networks , especially networks that carry IP telephony .
On the other hand , complete messages do not provide information about packet level behavior , which is necessary for internetworking layer monitoring . Paradoxically , the Network Management community has not paid much attention to this fact , although it compromises several VoIP management tasks . To fill in this gap , we provide an empirical evaluation of its impact for the most popular VoIP signaling protocols using traces from enterprise networks , and present the architecture and heuristic that we are currently developing to partially solve the effect of the segmentation .
Finding digital evidence in VoIP malicious attacks is the most difficult task , due to its associated features with converged network . In this paper , a Model of investigating VoIP malicious attacks is proposed for forensic analysis . The model formalizes hypotheses through information gathering and adopt a Secure Temporal Logic of Action(S-TLA+ ) in the process of reconstructing potential attack scenario .
The first is based on the traditional centralized IN service architecture . The second proposes a new decentralized architecture based on direct control of VoIP call control equipment by open service architecture interfaces .
VOIP technical Provides regular phone services , as well as other services such as the ability to create a multilateral conversation , voice mail , and the ability to customize ringtones and add details of people their caller and saved , was also able to send and receive data such as images and documents during a voice conversation . The research aims to assess the quality of the sound VOIP transmission over a private UMTS virtual network made up of two networks subgroups in two different locations are within about each other a distance of 80 Km , by examining VoIP scenarios using IP within the UMTS network used to transfer data through the delay and loss using simulation OPNET measurement program which It provides an environment perfectly mimic " the reality of the studied network , has enabled the results that have been reached in addition to evaluating the quality of voice transmission determine the width required service package , the type of audio encoding to be used .
In this paper , VoIP ( All-IP type , and for Gateway type ) for system and network scanning , and , IP Phone to get the information and analysis of the vulnerability . All-IP type and Gateway type discovered about the vulnerability of VoIP hacking attacks ( Denial of Service attacks , VoIP spam attacks ) is carried out . And that is a real VoIP system installed and operated in the field of security measures through research and analysis is proposed .
Because virtually everyone has access to the Internet , threats due to insecurity and anonymity reach critical levels and have to be tackled by both carriers and Internet Service Providers . Regaining trust-by-wire , comparable to classic fixed line telephones , could mitigate or even solve problems like Spam , Phishing , and the localization of VoIP emergency calls . This paper presents the hardware implementation of a new and highly flexible solution-Internet protocol-calling line identification presentation-which provides additional support for new services to restore peoplepsilas confidence into the Internet .
During the past 5 years , domestic VoIP service has grown rapidly absorbing more than 8 million subscribers . More recently , the increase of enterprise or institutional subscribers become more dominant than individual subscribers .
Thus it is effective method to identify IP and MAC spoofed attacks . I will construct the secured VOIP communication over MANET area using TCP / IP and storage Data in GSM modem as SMS ( inbox ) . Finally using IP and MAC get fail to prevent spoofing attack then DoS attack prevent using BIOS serial number of clients .
QoS-Aware Energy and Jitter-Efficient Downlink Predictive Scheduler for Heterogeneous Traffic LTE Networks An Infrastructure for Web Services Migration for Real-Time Applications Integration of re-active ad-hoc routing daemon into Quagga routing suite for wireless ad-hoc routers D5.1.1 Models of Real-time Applications on Service Oriented Infrastructures Road sign recognition system on Raspberry Pi Performance Evaluation of the Optimized Error Correction Based Hop Localization Approach in a Wireless Sensor Network High-speed parallel particle filter for PCMA signal blind separation Short-Term Traffic State Prediction Based on Temporal–Spatial Correlation FXECLMS Algorithm for Active Acoustic Windows: an Experimental Report Recognizing voice over IP: a robust front-end for speech recognition on the world wide web Where packet traces meet speech samples: an instrumental approach to perceptual QoS evaluation of VoIP Traffic Sensitive Active Queue Management IEEE 802.11 Throughput and Delay Analysis for mixed real time and normal data traffic Uncovering SIP Vulnerabilities to DoS Attacks Using Coloured Petri Nets ACM/Springer Mobile Networks and Applications (MONET) Special Issue on “Recent Advances in IEEE 802.11 WLANs: Protocols, Solutions and Future Directions” On the evolution of Internet technologies An Approach of Body Movement-based Interaction Towards Remote Collaboration A Security Approach using SIP Protocol in Imbedded Systems Parallel structures for fast estimation of echo path pure delay and their applications to sparse echo cancellers Identification of Multi-Reflective Echo Paths using Sub-Rate Adaptive Filtering in Presence of Moderate Aliasing Experimental studies of scalability in clustered Web systems Study the Best Approach for Virtual Private Network Implementation: CPU and Memory Usage Performance An OSPF based load sensitive QoS routing algorithm using alternate paths Reducing Handover Delay by Location Management in Mobile WiMAX Multicast and Broadcast Services An Empirical Evaluation of Deployed DPI Middleboxes and Their Implications for Policymakers Context-Aware Management Scheme for service continuity IP Quality of Service Models A survey of TCP-friendly router-based AQM schemes The memory effect and its implications on Web QoE modeling Method and apparatus for securing a quality of service in data communication Realization of Audio and Video Streaming Transmission Based on IP Network H.264/AVC技術應用於遠距醫療會診系統之建構 IPTV deployment - Trigger for advanced network services! Enhancing selective ISMACryp video encryption for real time applications in handheld devices Data Protection Based on Physical Separation: Concepts and Application Scenarios Captioning of Live TV Commentaries from the Olympic Games in Sochi: Some Interesting Insights A Design Approach for Recording Programs in Android Set-top Box User Research and Design for Live TV UX in China Migrating Live Streaming Applications onto Clouds: Challenges and a CloudStorm Solution Exploiting virtualization for delivering cloud-based IPTV services Cloud Resources Optimization for Delivering IPTelevision Services through Virtualization DYNAMIC BASED IPTV SERVICES THROUGH VIRTUAL ENVIRONMENT IN CLOUD RESOURCES Efficient algorithms for computation of the burstiness curve of video sources Scalability of telecom cloud architectures for live-TV distribution Mobile phones as second screen for TV, enabling inter-audience interaction Sharing the viewing experience through second screens Tweeting with the telly on! Scalable video streaming with prioritised network coding on end-system overlays Managing highly dynamic services using extended temporal network information models Large-scale multimedia content delivery over optical networks for interactive TV services Exhibition Monitoring System using USN/RFID based on ECA Real-Time Video Filtering and Overlay Character Generation on FPGA A Novel Hardware-Software Codesign of Network Protocol Stacks on a SoC Platform Real-time orthogonal vector projection algorithm based on GPU Application of Program Agents for Optimisation of VoIP Communication VoIP over DVB-RCS Satellite Systems: A Novel Capacity Request Mechanism for Improved Voice Activity Detection VoIP over DVB-RCS satellite systems: Trial results and the impact of adaptive speech coding using cross-layer design Fast Fault Recovery in Switched Networks for Carrying IP Telephony Traffic On the impact of TCP segmentation: Experience in VoIP monitoring VoIP evidence model: A new forensic method for investigating VoIP malicious attacks Toward an all-IP-based UMTS system architecture تقييم جودة نقل الصوت عبر شبكات UMTS A Study of Wired and wireless VoIP vulnerability analysis and hacking attacks and security IPclip: An architecture to restore Trust-by-Wire in packet-switched networks An Enhanced Location Offering Mechanism for Supporting Emergency Services in Ethernet-based Enterprise VoIP Networks VOIP and data storage in wireless GSM modem over MANET area
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