For elderly people who live alone , such a fall is a major risk . In this paper we present an approach for fall detection based on multiple cameras supported by a statistical behavior model . The model describes the spatio-temporal unexpectedness of objects in a scene and is used to verify a fall detected by a semantic driven fall detection . The existing optical sensor-based fall monitoring systems have some disadvantages , such as limited monitoring range and inconvenience to carry for users . Furthermore , the fall detection system based only on an accelerometer often mistakenly determines some activities of daily living ( ADL ) as falls , leading to low accuracy in fall detection . We propose a human fall monitoring system consisting of a highly portable sensor unit including a triaxis accelerometer , a triaxis gyroscope , and a triaxis magnetometer , and a mobile phone . This paper aims for avoidance of accidents and develop helmet detection system . We intend to use background subtraction and optical character recognition for fall detection and for helmet detection we use background subtraction and Hough transform descriptor . One of these purposes is to help of the physically disabled people who use a cane as a mobility aid by detecting the fall . This paper surveys the most popular approaches that have been used in fall detection , the challenges related to developing fall detectors , the techniques that have been used with the Kinect in fall detection , best points of interest ( joints ) to be tracked and the well-known Kinect-Based Fall Datasets . Finally , recommendations and future works will be summarized . Applications that perform continuous sensing on mobile phones have the potential to revolutionize everyday life . Examples range from medical and health monitoring applications , such as pedometers and fall detectors , to participatory sensing applications , such as noise pollution , traffic and seismic activity monitoring . Unfortunately , current mobile devices are a poor match for continuous sensing applications as they require the device to remain awake for extended periods of time , resulting in poor battery life . This paper presents a system for fall detecting using off-the-shelf electronic devices to detect the fall . We use a smart phone with an embedded tri-axial accelerometer sensor . The most important part of a fall detection algorithm is to distinguish fall from the other daily fall-like activities such as lie down and squat . This paper studied the algorithm of fall detection based on vision-approach , compared different approaches to detect falls and presents a new algorithm based on vision using multi-parameters to discriminate falls and other activities which turns out have a higher accuracy on fall detecting than some other vision-based fall down detection algorithms and the experimental results reveal this method made the fall detection more reliable . Switching a communication path from one Access Point ( AP ) to another in inter-domain WLANs is a critical challenge for delay-sensitive applications such as Voice over IP ( VoIP ) because communication quality during handoff ( HO ) is more likely to be deteriorated . To maintain VoIP quality during HO , we need to solve many problems . The existing channel allocation methods needs improvement in QoS parameters such as end-to-end delay , ripple factor and time factor in accessing the channel . To improve the QoS further , the admission control drop and block probability , efficient channel reservation approach is the better solution which leads to better QoS even for delay sensitive applications . A performance analysis is made among the existing methods for contention reduced channel allocation methods , energy conservation channel allocation methods and as last method Admission Control Drop_Block probability methods . Handoff latency causes an interruption in data transfer to the MN . There are many ways of improving the handoff procedure so that delay sensitive applications do not suffer due to handoff latency . Some of the mechanisms are early transfer of context to the new Base Station ( BS ) , early channel scanning , efficient inter-BS communication , introducing extra buffers between BS and MN , using a proxy for seamless connectivity , etc . Long-term evolution ( LTE ) narrowband Internet of Things ( NB-IoT ) is a Third Generation Partnership Project 3GPP defined cellular technology that is designed to enable connectivity to many low-cost and low-power/throughput IoT devices running delay-tolerant applications . NB-IoT can coexist within the LTE spectrum either in a standalone mode , in-band with LTE or in guard-band of LTE . Based on the queueing analysis , the impacts of channel and system parameters on the radio link level performance can be determined and hence cross-layer design and engineering can be performed . Also , efficient admission control schemes can be designed for delay-constrained applications . Analytical expressions of the outage probability and symbol error probability of the SU are obtained considering either perfect or statistical knowledge of the PU channel gains . Furthermore , we analyze the SU throughput for delay constrained applications with two hybrid automatic repeat request protocols , namely , repetition time diversity and incremental redundancy . Numerical results are provided to assess the effect of the number of SU-Txs and number of packet retransmissions on the secondary network performance . The proposed transmission scheme is numerically evaluated and shown to perform well , particularly in the low-SNR regime . Furthermore , it requires no encoding or decoding delay , making it suitable for delay-critical applications in wireless sensor networks . Real-time video transmission over the Internet is becoming increasingly desirable for videoconferencing and other interactive applications . The reliable transport protocols used by the Internet were designed mainly for non-real time data , and can not be used for delay critical applications , which must therefore be able to cope with packet loss . Motion compensated video coding is especially sensitive to loss because of temporal error propagation . In this article we describe how we integrated the WavPack audio coding scheme into our NMP system for interactive distributed network centric music performances . Our evaluation shows that WavPack performs very well in such types of delay-critical applications . In NMP it provides high compression ratios and good audio quality without additional buffering delays . The quality of service ( QoS ) limitation of Internet is major challenge for Voice over IP ( VoIP ) . Delay jitter and packet loss are two essential problems which impair the communication quality . It is essential to establish a relationship between QoE and QoS reacting to the network performance . The voice service delivered by the LTE system is a kind of Voice over IP ( VoIP ) service with no QoS-guaranteed mechanism . LTE Standard specifies the QoS , but does n't define the QoS/QoE relationship . These two different technolgies will co-exist while complementing each other in some regions , hence , a mobile node ( MN ) with dual interfaces traverses and executes many handovers ( HOs ) between 802.11 g and 802.16e . Meanwhile , there is a huge demand for Voice over IP ( VoIP ) service over wireless networks . However , VoIP is a delay and loss sensitive application , hence , VoIP session is more likely to be deteriorated during HO between 802.11 g and 802.16e . In this research , we propose an architectural solution to implement the voice over IP ( VoIP ) service in campus environment network . Voice over IP ( VoIP ) technology has become a discussion issue for this time being . Thus , a single communication may experience several handovers between WLANs divided into different IP subnets by Internet Service Providers ( ISPs ) , during which the communication quality may be degraded due to packet loss . In this paper , we propose an effective handover management scheme for Voice over IP ( VoIP ) communication in which the quality is sensitive to packet loss . For this purpose , our handover management scheme employs a transport protocol supporting multiple connections for VoIP communication and makes handover decisions based on a new criterion : the number of retransmissions experienced by a data frame on layer 2 . Accordingly , the cooperation among geographically close fog nodes and the cloud center is desired in fog computing with the ever-increasing computational demands from IoT applications . This paper investigates a workload allocation scheme in an IoT – fog – cloud cooperation system for reducing task service delay , aiming at satisfying as many as possible delay-sensitive IoT applications ’ quality of service ( QoS ) requirements . To this end , we first formulate the workload allocation problem in an IoT-edge-cloud cooperation system , which suggests optimal workload allocation among local fog node , neighboring fog node , and the cloud center to minimize task service delay . Voice over IP applications in wireless networks have gained increasing popularity in recent years . As a delay-sensitive real-time application , a VoIP flow is usually given higher priority in accessing the shared wireless channel than delay-insensitive non-real-time flows . In contention-based wireless networks two widely used prioritizing MAC mechanisms are class-dependent arbitration interframe space and class-dependent contention window . In order to improve estimation performance for wireless communications , statistical models of noise-suppressed speech are also incorporated . Next , a hybrid signal-and-link-parametric quality estimation paradigm is proposed for emerging wireless-VoIP communications . The algorithm uses VoIP connection parameters to estimate a base quality representative of the packet switching network . This quick reference summarizes all commands , keywords , command arguments , and associated prompts relevant for Exam 640 - 461 , and offers tips and examples for applying these commands in live Cisco Unified Communications environments . Topics covered include Fundamentals : Unified Communications components , functions , call signaling , media flows , and VoIP quality Preparation : Configure Cisco switches , routers , and other devices to support Unified Voice Provisioning : Create/modify users , accounts , endpoints , directory numbers , user and mobility features , and calling privileges Messaging and Presence : Configure voice messaging , Cisco Unity Connection , and Cisco Unified Presence Management/Monitoring : Generate reports , monitor voicemail usage , perform backups , and more Support : Verify PSTN connectivity and define fault domains ; troubleshoot endpoints , call quality , and voicemail Access all CCNA Voice commandsuse as a quick , offline resource for research and solutions Logical how-to topic groupings provide one-stop research Great for review before CCNA Voice 640 - 461 certification exams Includes configuration examples for Cisco Unified Communications Manager , Cisco Unified Communications Manager Express , Cisco Unity Connection , and Cisco Unified Presence Compact size makes it easy to carry with you , wherever you go Create Your Own Journal section with blank , lined pages allows you to personalize the book for your needs What Do You Want to Do ? chart inside cover helps you quickly reference specific tasks This book is part of the Cisco Press Certification Self-Study Product Family , which offers readers a self-paced study routine for Cisco certification exams . Titles in the Cisco Press Certification Self-Study Product Family are part of a recommended learning program from Cisco that includes simulation and hands-on training from authorized Cisco Learning Partners and self-study products from Cisco Press . Overall , for the evaluation with scientific exactitude of quality factors , we examine the proposed technique based on the realtime phone-call service through heterogeneous network systems : PSTN to SIP gateways . The experimental results confirm that realtime message report procedures and management schemes achieve a sufficient for reporting and managing of quality factors for the quality guaranteed realtime VoIP services . Many operators developed a variety of new services to attract users . This article will focus on the importance of the user interface design of mobile VoIP communication software , analyze and summarize the principles of UI design of mobile VoIP communication software through analysis of the existing UI design of mobile VoIP communication . Managing QoS of VoIP is a challenge especially in a heterogeneous network environment such as Internet where often exists limitations or bottlenecks for the bandwidth . As the Internet protocol is evolving to a new version ( v6 ) that offers a bigger address space at the cost of a bigger header size , there has been unawareness and worry how the increased header size affects real-time traffic , especially in bandwidth-limited wireless systems . PacketCable/sup TM/ defines a multimedia system architecture that overlays a high-speed cable modem access network , with the goal of enabling a wide variety of Internet protocol-based multimedia services , such as voice over IP , unified messaging , PBX extension , videoconferencing , and online gaming . Currently , the architecture provides a complete solution for delivering VoIP services . We present the core capabilities necessary to implement such services in a scalable fashion . Policy-Based Operations.- Policy-Based Management of Grids and Networks Through an Hierarchical Architecture.- Policy-Based Service Provisioning for Mobile Users.- Dynamic IP-Grouping Scheme with Reduced Wireless Signaling Cost in the Mobile Internet.- 3 G Wireless Networks Provisioning and Monitoring Via Policy Based Management.- Service Monitoring.- Combined Performance Analysis of Signal Level-Based Dynamic Channel Allocation and Adaptive Antennas.- Exploring Service Reliability for Content Distribution to Partial or Intermittent DVB-S Satellite Receivers.- Priority-Based Recovery Assurance for Double-Link Failure in Optical Mesh Networks with Insufficient Spare Resources.- Service Model and Its Application to Impact Analysis.- Intelligent Architectural Systems.- Active Networks and Computational Intelligence.- Distributed Artificial Intelligence for Network Management Systems - New Approaches.- Network Traffic Sensor for Multiprocessor Architectures : Design Improvement Proposals.- Software Modeling for Open Distributed Network Monitoring Systems.- Mobility and Wireless.- Analysis and Contrast Between STC and Spatial Diversity Techniques for OFDM WLAN with Channel Estimation.- Cumulative Caching for Reduced User-Perceived Latency for WWW Transfers on Networks with Satellite Links.- Mobility Agent Advertisement Mechanism for Supporting Mobile IP in Ad Hoc Networks.- Agent Selection Strategies in Wireless Networks with Multihomed Mobile IP.- Protocol Mechanisms.- An On-Demand QoS Routing Protocol for Mobile Ad-Hoc Networks.- Point-to-Group Blocking in 3-Stage Switching Networks with Multicast Traffic Streams.- Considerations on Inter-domain QoS and Traffic Engineering Issues Through a Utopian Approach.- Probabilistic Routing in Intermittently Connected Networks.- Performance Across Domains.- Communication Protocol for Interdomain Resource Reservation.- Performance Evaluation of Shortest Path Computation for IP and MPLS Multi-service Networks over Open Source Implementation.- Design and Evaluation of Redundant IPC Network Adequate for an Edge Router.- Leaky Bucket Based Buffer Management Scheme to Support Differentiated Service in Packet-Switched Networks.- An Improved Service Differentiation Scheme for VBR VoIP in Ad-Hoc Networks Connected to Wired Networks . The increasing number of users make VoIP also an interesting target for attackers . VoIP phishing and identity theft are gaining relevance . The lack of options for the reliable authentication between communication partners and the possibilities for anonymisation in IP-based telephony will allow attackers to steal confidential and personal information unperceived . SIP-based VoIP applications over Direct Mode of Operation ( DMO ) , which behaves in a way similar to Mobile Ad-hoc Network ( MANET ) systems , have three main performance categories related to the QoS. These categories are the SIP signaling , voice data transmission , and MANET routing . The SIP signaling controls the VoIP calls initiation , termination , and modifications . The major QoS parameters of VoIP that are managed by SIP signaling are the registration intervals , call setup time , and call termination time . Wi-Fi phones are becoming extremely popular for implementation of VoIP ( or VoWLAN ) service . It is well known that voice service is quite critical to delay and delay jitter , therefore it appears important to evaluate VoWLAN service performance . However , voice is the most natural and direct way to express people 's sentiments in real-time . With the rapid development of smart phone voice dialogue applications ( e.g. , Siri and Sogou Voice Assistant ) , the large-scale networked voice data can help us better quantitatively understand the sentimental world we live in . In this paper , we study the problem of sentiment prediction from large-scale networked voice data . Compared to text or image data , voice is the most natural and direct way to express ones ' emotions in real-time . With the increasing adoption of smart phone voice dialogue applications ( e.g. , Siri and Sogou Voice Assistant ) , the large-scale networked voice data can help us better quantitatively understand the emotional world we live in . In this paper , we study the problem of inferring public emotions from large-scale networked voice data . This paper examines how pitch range declination and reset contribute to turn-taking organisation . This is part of a broader study of voice prosody , i.e. , how pitch , voice quality and temporal features combine for various prosodic functions , both linguistic and paralinguistic . The present study first investigates the effect of the speech unit position in a turn on its pitch range . As part of a broader study of voice prosody in speech communication , this paper looks at intonation in turn-taking . It examines the distribution of pitch patterns and communicative types in the interpausal units ( IPUs ) preceding pause or gap silences extracted from a corpus of spontaneous speech of Irish English . Based on this finding , a closed-form expression for the maximum number of voice calls is developed as a function of the TXOP value . Finally , we propose a simple yet accurate approximation for voice-capacity estimation and provide some insights gained from the approximation . Based on this finding a closed form expression for the maximum number of voice calls is developed as a function of TXOP value . Finally , we propose a simple yet accurate voice capacity approximation formula for voice capacity estimation in WLAN and provide some insights that an be gained from this formula . Basic purpose of ASCI services for railways is to provide the distribution of speech into a predefined geographical area , towards all or towards only a group of service subscribers located in that area . Therefore , Voice Broadcast Service ( VBS ) and Voice Group Call Service ( VGCS ) were defined for this purpose . The phonetogram represents an area limited by piano and forte contours of the sound pressure levels along the vocal range . Phonetograms are used in the diagnosis of voice status . Apart from reference phonetograms and the extraction of single parameters , phonetograms have not been evaluated quantitatively in medical practice . We demonstrate our proposed mechanisms by implementing two applications , Face RMS and voice recognition . The proposed framework gives a case study to illustrate software framework and design flow for emerging RMS-based and voice recognition applications on embedded multi-core systems equipped with Android systems . These findings support the assumption that at least some of these nasal-breathy sound changes involve a stage in which the two articulations co-occur . We claim that the production of breathy voice quality during nasals can arise through listener misperception or phonetic enhancement . These findings also contribute to the understanding of nasality as an abstract feature that involves multiple articulations . Spectrum Inversion , Time Segment Permutation and Reversed Time Segmentation are the analogue techniques implemented . Digital encryption of the voice signals is done by pseudorandom number masking . A Simplex communication system is designed and constructed . The ISDN offers various services through multipurpose digital subscriber lines . The switching program for the ISDN should be so designed as to realize the control of various voice and nonvoice services and provide for developing service enhancement . To attain the goal , the ISDN services are classified hierarchically and each category is allocated to appropriate network nodes for implementation . We propose to integrate a singing voice detector with a multimedia , multi-touch game where the user has to perform simple tasks at certain key points in the music . While the ultimate goal is to automatically create visual content in response to features extracted from the music , here we give special focus to the detection of voice segments in music songs . The solution presented extracts the Mel-Frequency Cepstral Coefficients of the sound and uses a Hidden Markov Model to infer if the sound has voice .
For elderly people who live alone , such a fall is a major risk . In this paper we present an approach for fall detection based on multiple cameras supported by a statistical behavior model . The model describes the spatio-temporal unexpectedness of objects in a scene and is used to verify a fall detected by a semantic driven fall detection .
The existing optical sensor-based fall monitoring systems have some disadvantages , such as limited monitoring range and inconvenience to carry for users . Furthermore , the fall detection system based only on an accelerometer often mistakenly determines some activities of daily living ( ADL ) as falls , leading to low accuracy in fall detection . We propose a human fall monitoring system consisting of a highly portable sensor unit including a triaxis accelerometer , a triaxis gyroscope , and a triaxis magnetometer , and a mobile phone .
This paper aims for avoidance of accidents and develop helmet detection system . We intend to use background subtraction and optical character recognition for fall detection and for helmet detection we use background subtraction and Hough transform descriptor .
One of these purposes is to help of the physically disabled people who use a cane as a mobility aid by detecting the fall . This paper surveys the most popular approaches that have been used in fall detection , the challenges related to developing fall detectors , the techniques that have been used with the Kinect in fall detection , best points of interest ( joints ) to be tracked and the well-known Kinect-Based Fall Datasets . Finally , recommendations and future works will be summarized .
Applications that perform continuous sensing on mobile phones have the potential to revolutionize everyday life . Examples range from medical and health monitoring applications , such as pedometers and fall detectors , to participatory sensing applications , such as noise pollution , traffic and seismic activity monitoring . Unfortunately , current mobile devices are a poor match for continuous sensing applications as they require the device to remain awake for extended periods of time , resulting in poor battery life .
This paper presents a system for fall detecting using off-the-shelf electronic devices to detect the fall . We use a smart phone with an embedded tri-axial accelerometer sensor .
The most important part of a fall detection algorithm is to distinguish fall from the other daily fall-like activities such as lie down and squat . This paper studied the algorithm of fall detection based on vision-approach , compared different approaches to detect falls and presents a new algorithm based on vision using multi-parameters to discriminate falls and other activities which turns out have a higher accuracy on fall detecting than some other vision-based fall down detection algorithms and the experimental results reveal this method made the fall detection more reliable .
Switching a communication path from one Access Point ( AP ) to another in inter-domain WLANs is a critical challenge for delay-sensitive applications such as Voice over IP ( VoIP ) because communication quality during handoff ( HO ) is more likely to be deteriorated . To maintain VoIP quality during HO , we need to solve many problems .
The existing channel allocation methods needs improvement in QoS parameters such as end-to-end delay , ripple factor and time factor in accessing the channel . To improve the QoS further , the admission control drop and block probability , efficient channel reservation approach is the better solution which leads to better QoS even for delay sensitive applications . A performance analysis is made among the existing methods for contention reduced channel allocation methods , energy conservation channel allocation methods and as last method Admission Control Drop_Block probability methods .
Handoff latency causes an interruption in data transfer to the MN . There are many ways of improving the handoff procedure so that delay sensitive applications do not suffer due to handoff latency . Some of the mechanisms are early transfer of context to the new Base Station ( BS ) , early channel scanning , efficient inter-BS communication , introducing extra buffers between BS and MN , using a proxy for seamless connectivity , etc .
Long-term evolution ( LTE ) narrowband Internet of Things ( NB-IoT ) is a Third Generation Partnership Project 3GPP defined cellular technology that is designed to enable connectivity to many low-cost and low-power/throughput IoT devices running delay-tolerant applications . NB-IoT can coexist within the LTE spectrum either in a standalone mode , in-band with LTE or in guard-band of LTE .
Based on the queueing analysis , the impacts of channel and system parameters on the radio link level performance can be determined and hence cross-layer design and engineering can be performed . Also , efficient admission control schemes can be designed for delay-constrained applications .
Analytical expressions of the outage probability and symbol error probability of the SU are obtained considering either perfect or statistical knowledge of the PU channel gains . Furthermore , we analyze the SU throughput for delay constrained applications with two hybrid automatic repeat request protocols , namely , repetition time diversity and incremental redundancy . Numerical results are provided to assess the effect of the number of SU-Txs and number of packet retransmissions on the secondary network performance .
The proposed transmission scheme is numerically evaluated and shown to perform well , particularly in the low-SNR regime . Furthermore , it requires no encoding or decoding delay , making it suitable for delay-critical applications in wireless sensor networks .
Real-time video transmission over the Internet is becoming increasingly desirable for videoconferencing and other interactive applications . The reliable transport protocols used by the Internet were designed mainly for non-real time data , and can not be used for delay critical applications , which must therefore be able to cope with packet loss . Motion compensated video coding is especially sensitive to loss because of temporal error propagation .
In this article we describe how we integrated the WavPack audio coding scheme into our NMP system for interactive distributed network centric music performances . Our evaluation shows that WavPack performs very well in such types of delay-critical applications . In NMP it provides high compression ratios and good audio quality without additional buffering delays .
The quality of service ( QoS ) limitation of Internet is major challenge for Voice over IP ( VoIP ) . Delay jitter and packet loss are two essential problems which impair the communication quality .
It is essential to establish a relationship between QoE and QoS reacting to the network performance . The voice service delivered by the LTE system is a kind of Voice over IP ( VoIP ) service with no QoS-guaranteed mechanism . LTE Standard specifies the QoS , but does n't define the QoS/QoE relationship .
These two different technolgies will co-exist while complementing each other in some regions , hence , a mobile node ( MN ) with dual interfaces traverses and executes many handovers ( HOs ) between 802.11 g and 802.16e . Meanwhile , there is a huge demand for Voice over IP ( VoIP ) service over wireless networks . However , VoIP is a delay and loss sensitive application , hence , VoIP session is more likely to be deteriorated during HO between 802.11 g and 802.16e .
In this research , we propose an architectural solution to implement the voice over IP ( VoIP ) service in campus environment network . Voice over IP ( VoIP ) technology has become a discussion issue for this time being .
Thus , a single communication may experience several handovers between WLANs divided into different IP subnets by Internet Service Providers ( ISPs ) , during which the communication quality may be degraded due to packet loss . In this paper , we propose an effective handover management scheme for Voice over IP ( VoIP ) communication in which the quality is sensitive to packet loss . For this purpose , our handover management scheme employs a transport protocol supporting multiple connections for VoIP communication and makes handover decisions based on a new criterion : the number of retransmissions experienced by a data frame on layer 2 .
Accordingly , the cooperation among geographically close fog nodes and the cloud center is desired in fog computing with the ever-increasing computational demands from IoT applications . This paper investigates a workload allocation scheme in an IoT – fog – cloud cooperation system for reducing task service delay , aiming at satisfying as many as possible delay-sensitive IoT applications ’ quality of service ( QoS ) requirements . To this end , we first formulate the workload allocation problem in an IoT-edge-cloud cooperation system , which suggests optimal workload allocation among local fog node , neighboring fog node , and the cloud center to minimize task service delay .
Voice over IP applications in wireless networks have gained increasing popularity in recent years . As a delay-sensitive real-time application , a VoIP flow is usually given higher priority in accessing the shared wireless channel than delay-insensitive non-real-time flows . In contention-based wireless networks two widely used prioritizing MAC mechanisms are class-dependent arbitration interframe space and class-dependent contention window .
In order to improve estimation performance for wireless communications , statistical models of noise-suppressed speech are also incorporated . Next , a hybrid signal-and-link-parametric quality estimation paradigm is proposed for emerging wireless-VoIP communications . The algorithm uses VoIP connection parameters to estimate a base quality representative of the packet switching network .
This quick reference summarizes all commands , keywords , command arguments , and associated prompts relevant for Exam 640 - 461 , and offers tips and examples for applying these commands in live Cisco Unified Communications environments . Topics covered include Fundamentals : Unified Communications components , functions , call signaling , media flows , and VoIP quality Preparation : Configure Cisco switches , routers , and other devices to support Unified Voice Provisioning : Create/modify users , accounts , endpoints , directory numbers , user and mobility features , and calling privileges Messaging and Presence : Configure voice messaging , Cisco Unity Connection , and Cisco Unified Presence Management/Monitoring : Generate reports , monitor voicemail usage , perform backups , and more Support : Verify PSTN connectivity and define fault domains ; troubleshoot endpoints , call quality , and voicemail Access all CCNA Voice commandsuse as a quick , offline resource for research and solutions Logical how-to topic groupings provide one-stop research Great for review before CCNA Voice 640 - 461 certification exams Includes configuration examples for Cisco Unified Communications Manager , Cisco Unified Communications Manager Express , Cisco Unity Connection , and Cisco Unified Presence Compact size makes it easy to carry with you , wherever you go Create Your Own Journal section with blank , lined pages allows you to personalize the book for your needs What Do You Want to Do ? chart inside cover helps you quickly reference specific tasks This book is part of the Cisco Press Certification Self-Study Product Family , which offers readers a self-paced study routine for Cisco certification exams . Titles in the Cisco Press Certification Self-Study Product Family are part of a recommended learning program from Cisco that includes simulation and hands-on training from authorized Cisco Learning Partners and self-study products from Cisco Press .
Overall , for the evaluation with scientific exactitude of quality factors , we examine the proposed technique based on the realtime phone-call service through heterogeneous network systems : PSTN to SIP gateways . The experimental results confirm that realtime message report procedures and management schemes achieve a sufficient for reporting and managing of quality factors for the quality guaranteed realtime VoIP services .
Many operators developed a variety of new services to attract users . This article will focus on the importance of the user interface design of mobile VoIP communication software , analyze and summarize the principles of UI design of mobile VoIP communication software through analysis of the existing UI design of mobile VoIP communication .
Managing QoS of VoIP is a challenge especially in a heterogeneous network environment such as Internet where often exists limitations or bottlenecks for the bandwidth . As the Internet protocol is evolving to a new version ( v6 ) that offers a bigger address space at the cost of a bigger header size , there has been unawareness and worry how the increased header size affects real-time traffic , especially in bandwidth-limited wireless systems .
PacketCable/sup TM/ defines a multimedia system architecture that overlays a high-speed cable modem access network , with the goal of enabling a wide variety of Internet protocol-based multimedia services , such as voice over IP , unified messaging , PBX extension , videoconferencing , and online gaming . Currently , the architecture provides a complete solution for delivering VoIP services . We present the core capabilities necessary to implement such services in a scalable fashion .
Policy-Based Operations.- Policy-Based Management of Grids and Networks Through an Hierarchical Architecture.- Policy-Based Service Provisioning for Mobile Users.- Dynamic IP-Grouping Scheme with Reduced Wireless Signaling Cost in the Mobile Internet.- 3 G Wireless Networks Provisioning and Monitoring Via Policy Based Management.- Service Monitoring.- Combined Performance Analysis of Signal Level-Based Dynamic Channel Allocation and Adaptive Antennas.- Exploring Service Reliability for Content Distribution to Partial or Intermittent DVB-S Satellite Receivers.- Priority-Based Recovery Assurance for Double-Link Failure in Optical Mesh Networks with Insufficient Spare Resources.- Service Model and Its Application to Impact Analysis.- Intelligent Architectural Systems.- Active Networks and Computational Intelligence.- Distributed Artificial Intelligence for Network Management Systems - New Approaches.- Network Traffic Sensor for Multiprocessor Architectures : Design Improvement Proposals.- Software Modeling for Open Distributed Network Monitoring Systems.- Mobility and Wireless.- Analysis and Contrast Between STC and Spatial Diversity Techniques for OFDM WLAN with Channel Estimation.- Cumulative Caching for Reduced User-Perceived Latency for WWW Transfers on Networks with Satellite Links.- Mobility Agent Advertisement Mechanism for Supporting Mobile IP in Ad Hoc Networks.- Agent Selection Strategies in Wireless Networks with Multihomed Mobile IP.- Protocol Mechanisms.- An On-Demand QoS Routing Protocol for Mobile Ad-Hoc Networks.- Point-to-Group Blocking in 3-Stage Switching Networks with Multicast Traffic Streams.- Considerations on Inter-domain QoS and Traffic Engineering Issues Through a Utopian Approach.- Probabilistic Routing in Intermittently Connected Networks.- Performance Across Domains.- Communication Protocol for Interdomain Resource Reservation.- Performance Evaluation of Shortest Path Computation for IP and MPLS Multi-service Networks over Open Source Implementation.- Design and Evaluation of Redundant IPC Network Adequate for an Edge Router.- Leaky Bucket Based Buffer Management Scheme to Support Differentiated Service in Packet-Switched Networks.- An Improved Service Differentiation Scheme for VBR VoIP in Ad-Hoc Networks Connected to Wired Networks .
The increasing number of users make VoIP also an interesting target for attackers . VoIP phishing and identity theft are gaining relevance . The lack of options for the reliable authentication between communication partners and the possibilities for anonymisation in IP-based telephony will allow attackers to steal confidential and personal information unperceived .
SIP-based VoIP applications over Direct Mode of Operation ( DMO ) , which behaves in a way similar to Mobile Ad-hoc Network ( MANET ) systems , have three main performance categories related to the QoS. These categories are the SIP signaling , voice data transmission , and MANET routing . The SIP signaling controls the VoIP calls initiation , termination , and modifications . The major QoS parameters of VoIP that are managed by SIP signaling are the registration intervals , call setup time , and call termination time .
Wi-Fi phones are becoming extremely popular for implementation of VoIP ( or VoWLAN ) service . It is well known that voice service is quite critical to delay and delay jitter , therefore it appears important to evaluate VoWLAN service performance .
However , voice is the most natural and direct way to express people 's sentiments in real-time . With the rapid development of smart phone voice dialogue applications ( e.g. , Siri and Sogou Voice Assistant ) , the large-scale networked voice data can help us better quantitatively understand the sentimental world we live in . In this paper , we study the problem of sentiment prediction from large-scale networked voice data .
Compared to text or image data , voice is the most natural and direct way to express ones ' emotions in real-time . With the increasing adoption of smart phone voice dialogue applications ( e.g. , Siri and Sogou Voice Assistant ) , the large-scale networked voice data can help us better quantitatively understand the emotional world we live in . In this paper , we study the problem of inferring public emotions from large-scale networked voice data .
This paper examines how pitch range declination and reset contribute to turn-taking organisation . This is part of a broader study of voice prosody , i.e. , how pitch , voice quality and temporal features combine for various prosodic functions , both linguistic and paralinguistic . The present study first investigates the effect of the speech unit position in a turn on its pitch range .
As part of a broader study of voice prosody in speech communication , this paper looks at intonation in turn-taking . It examines the distribution of pitch patterns and communicative types in the interpausal units ( IPUs ) preceding pause or gap silences extracted from a corpus of spontaneous speech of Irish English .
Based on this finding , a closed-form expression for the maximum number of voice calls is developed as a function of the TXOP value . Finally , we propose a simple yet accurate approximation for voice-capacity estimation and provide some insights gained from the approximation .
Based on this finding a closed form expression for the maximum number of voice calls is developed as a function of TXOP value . Finally , we propose a simple yet accurate voice capacity approximation formula for voice capacity estimation in WLAN and provide some insights that an be gained from this formula .
Basic purpose of ASCI services for railways is to provide the distribution of speech into a predefined geographical area , towards all or towards only a group of service subscribers located in that area . Therefore , Voice Broadcast Service ( VBS ) and Voice Group Call Service ( VGCS ) were defined for this purpose .
The phonetogram represents an area limited by piano and forte contours of the sound pressure levels along the vocal range . Phonetograms are used in the diagnosis of voice status . Apart from reference phonetograms and the extraction of single parameters , phonetograms have not been evaluated quantitatively in medical practice .
We demonstrate our proposed mechanisms by implementing two applications , Face RMS and voice recognition . The proposed framework gives a case study to illustrate software framework and design flow for emerging RMS-based and voice recognition applications on embedded multi-core systems equipped with Android systems .
These findings support the assumption that at least some of these nasal-breathy sound changes involve a stage in which the two articulations co-occur . We claim that the production of breathy voice quality during nasals can arise through listener misperception or phonetic enhancement . These findings also contribute to the understanding of nasality as an abstract feature that involves multiple articulations .
Spectrum Inversion , Time Segment Permutation and Reversed Time Segmentation are the analogue techniques implemented . Digital encryption of the voice signals is done by pseudorandom number masking . A Simplex communication system is designed and constructed .
The ISDN offers various services through multipurpose digital subscriber lines . The switching program for the ISDN should be so designed as to realize the control of various voice and nonvoice services and provide for developing service enhancement . To attain the goal , the ISDN services are classified hierarchically and each category is allocated to appropriate network nodes for implementation .
We propose to integrate a singing voice detector with a multimedia , multi-touch game where the user has to perform simple tasks at certain key points in the music . While the ultimate goal is to automatically create visual content in response to features extracted from the music , here we give special focus to the detection of voice segments in music songs . The solution presented extracts the Mel-Frequency Cepstral Coefficients of the sound and uses a Hidden Markov Model to infer if the sound has voice .
Introducing a Statistical Behavior Model into Camera-Based Fall Detection Highly Portable, Sensor-Based System for Human Fall Monitoring Automatic Helmet Detection on Public Roads Computer Vision-based Fall Detection Methods Using The Kinect Camera: A Survey SIDEWINDER: Efficient and Easy-to-Use Continuous Sensing Fall Detecting and Alarming Based on Mobile Phone A Novel Video-Surveillance-Based Algorithm of Fall Detection Seamless VoWLAN handoff management based on estimation of AP queue length and frame retries Survey on Channel Allocation Techniques for Wireless Mesh Network to Reduce Contention with Energy Requirement A Review of Handoff Optimization Techniques in Data Networks NB-IoT Uplink Receiver Design and Performance Study Queueing analysis and admission control for multi-rate wireless networks with opportunistic scheduling and ARQ-based error control Performance analysis of opportunistic scheduling with HARQ for cognitive radio networks Analog distributed source-channel coding using sinusoids Packet loss resilient videoconferencing using H.263 Using wavpack for real-time audio coding in interactive applications A New Quality-Driven QoS Control Method for AMR-WB in VoIP The Regressive QoE Model for VoLTE Vertical Handover Management for VoIP Session over Broadband Wireless Networks Analyzing of MOS and Codec Selection for Voice over IP Technology Handover management based on the number of data frame retransmissions for VoWLAN Online Workload Allocation via Fog-Fog-Cloud Cooperation to Reduce IoT Task Service Delay Voice capacity analysis of WLANS with channel access prioritizing mechanisms Blind estimation of perceptual quality for modern speech communications CCNA Voice Portable Command Guide A Study of the Realtime Message Report Procedures and Management Schemes for the Quality Guaranteed VoIP Services User interface design research of mobile voip communication software Comparative analysis of VoIPv4 and VoIPv6 in a bandwidth-limited wireless LAN testbed The PacketCable architecture First International Workshop, SAPIR 2004, Fortaleza, Brazil, August 1-6, 2004, Proceedings Approach to Identity Card-Based Voice-over-IP Authentication QoS and Performance Evaluation for SIP-Based VoIP Over DMO On VoWLAN service performance in indoor environment Acoustics, content and geo-information based sentiment prediction from large-scale networked voice data Learning to Infer Public Emotions from Large-Scale Networked Voice Data Pitch range declination and reset in turn-taking organisation The distribution of pitch patterns and communicative types in speech-chunks preceding pauses and gaps VoIP Capacity—Analysis, Improvements, and Limits in IEEE 802.11 Wireless LAN VoIP capacity analysis in IEEE 802.11 WLAN Advanced Speech Call Items for GSM-Railway Computer aided evaluation of phonetograms. Support of software framework for embedded multi-core systems with Android environments Breathy voice during nasality: A cross-linguistic study Microprocessor based voice scrambling techniques Switching System Software Design for ISDN Detecting key features in popular music: case study ? singing voice detection
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