Highlihgts of the results and outcomes of ITU-T Study Group 12 meeting in January 2017 : Performance, Quality of Service and Quality of Experience
Author : Kwame BAAH-ACHEAMFUOR, Study Group 12 Chairman
Real-time, non-Intrusive Evaluation of VoIPadil raja
This document describes research into developing non-intrusive models for estimating voice over IP (VoIP) call quality using genetic programming. The researchers created a VoIP simulation environment and used it to generate training data representing different network conditions. Genetic programming experiments were performed to evolve models relating network parameters like packet loss to MOS scores. The best models achieved good accuracy compared to intrusive quality estimation methods, using only 1-3 variables. The models allow real-time VoIP quality monitoring without a reference signal.
Dimitri Dey has over 6 years of experience testing Android and LTE technologies at Qualcomm. He has expertise in integration testing, performance testing, and debugging LTE applications and services. Currently, he is working on a project creating builds and testing automotive applications on the Mojave board for Qualcomm.
This document provides an overview of the ACM workshop on advanced video streaming techniques for peer-to-peer networks and social networking. It summarizes the topics of interest for the workshop, which included innovative P2P video streaming solutions, social media content distribution, and advanced video coding techniques for real-time applications. It also summarizes the 15 papers accepted to the workshop, which covered topics like multi-source video distribution, modeling end-to-end delay, and improvements to quality of experience for multiple description video transmission. An invited talk was also given by a representative from the BBC on audio/visual content delivery over P2P networks.
OPTIMIZING VOIP USING A CROSS LAYER CALL ADMISSION CONTROL SCHEMEIJCNCJournal
This document discusses optimizing VoIP quality over wireless networks using a cross-layer call admission control scheme. It proposes monitoring real-time control protocol reports and data rates at the MAC layer to determine when quality is degraded. When quality degrades due to issues like network congestion or variable transmission rates, the solution is to adapt the packet size or codec type. The proposed scheme is simulated using a wireless campus network model to improve performance.
An Evolutionary Approach to Speech Quality Estimationadil raja
This document presents research using genetic programming to develop non-intrusive models for estimating voice over IP (VoIP) quality. Researchers used a VoIP simulation environment to generate distorted speech files under different network conditions and trained genetic programs to map transport layer metrics like packet loss and delay to mean opinion scores. The best models achieved good accuracy compared to the intrusive PESQ standard with only 1-3 variables, making them suitable for real-time VoIP quality monitoring. Future work aims to include wideband codecs and develop a unified quality estimation model.
The document provides an overview of selected current activities within MPEG, including requirements and timelines. It discusses the Mobile Visual Search work item which aims to enable efficient transmission of local image features for mobile visual search applications. It also outlines the MPEG Media Transport work item which focuses on efficient delivery of media to enable content and network adaptive streaming. Additionally, it summarizes the Advanced IPTV Terminal work item and its goal of defining elementary services and protocols to enable interoperability.
This document discusses using Information and Communication Technologies (ICT) to help community radio stations in India. It outlines the author's past work developing systems like PhonePeti, an automated answering machine, and future planned work measuring rural internet connectivity and developing a content sharing tool. The author conducted a pilot study measuring internet performance at a community radio station which found DNS lookup latency around 110ms and download throughput varying significantly between 248-731kbps. Future work will include deploying measurement devices to further analyze rural internet access technologies and service providers.
PERFORMANCE EVALUATION OF H.265/MPEG-HEVC, VP9 AND H.264/MPEGAVC VIDEO CODINGijma
This study evaluates the performance of the three latest video codecs H.265/MPEG-HEVC, H.264/MPEGAVC
and VP9. The evaluation is based on both subjective and objective quality metrics. The assessment
metric Double Stimulus Impairment Scale (DSIS) is used to evaluate the subjective quality of the
compressed video sequences. The Peak Signal-to-Noise Ratio (PSNR) metricis used for the objective
evaluation. Moreover, this work studies the effect of frame rate and resolution on the encoders’
performance. The extensive number of experiments are conducted with similar encoding configurations for
the three studied encoders. The evaluation results show that H.265/MPEG-HEVC provides superior bitrate
saving capabilities compared to H.264 and VP9. However, VP9 shows lower encoding time than
H.265/MPEG-HEVC but higher encoding time compared to H.264.
Real-time, non-Intrusive Evaluation of VoIPadil raja
This document describes research into developing non-intrusive models for estimating voice over IP (VoIP) call quality using genetic programming. The researchers created a VoIP simulation environment and used it to generate training data representing different network conditions. Genetic programming experiments were performed to evolve models relating network parameters like packet loss to MOS scores. The best models achieved good accuracy compared to intrusive quality estimation methods, using only 1-3 variables. The models allow real-time VoIP quality monitoring without a reference signal.
Dimitri Dey has over 6 years of experience testing Android and LTE technologies at Qualcomm. He has expertise in integration testing, performance testing, and debugging LTE applications and services. Currently, he is working on a project creating builds and testing automotive applications on the Mojave board for Qualcomm.
This document provides an overview of the ACM workshop on advanced video streaming techniques for peer-to-peer networks and social networking. It summarizes the topics of interest for the workshop, which included innovative P2P video streaming solutions, social media content distribution, and advanced video coding techniques for real-time applications. It also summarizes the 15 papers accepted to the workshop, which covered topics like multi-source video distribution, modeling end-to-end delay, and improvements to quality of experience for multiple description video transmission. An invited talk was also given by a representative from the BBC on audio/visual content delivery over P2P networks.
OPTIMIZING VOIP USING A CROSS LAYER CALL ADMISSION CONTROL SCHEMEIJCNCJournal
This document discusses optimizing VoIP quality over wireless networks using a cross-layer call admission control scheme. It proposes monitoring real-time control protocol reports and data rates at the MAC layer to determine when quality is degraded. When quality degrades due to issues like network congestion or variable transmission rates, the solution is to adapt the packet size or codec type. The proposed scheme is simulated using a wireless campus network model to improve performance.
An Evolutionary Approach to Speech Quality Estimationadil raja
This document presents research using genetic programming to develop non-intrusive models for estimating voice over IP (VoIP) quality. Researchers used a VoIP simulation environment to generate distorted speech files under different network conditions and trained genetic programs to map transport layer metrics like packet loss and delay to mean opinion scores. The best models achieved good accuracy compared to the intrusive PESQ standard with only 1-3 variables, making them suitable for real-time VoIP quality monitoring. Future work aims to include wideband codecs and develop a unified quality estimation model.
The document provides an overview of selected current activities within MPEG, including requirements and timelines. It discusses the Mobile Visual Search work item which aims to enable efficient transmission of local image features for mobile visual search applications. It also outlines the MPEG Media Transport work item which focuses on efficient delivery of media to enable content and network adaptive streaming. Additionally, it summarizes the Advanced IPTV Terminal work item and its goal of defining elementary services and protocols to enable interoperability.
This document discusses using Information and Communication Technologies (ICT) to help community radio stations in India. It outlines the author's past work developing systems like PhonePeti, an automated answering machine, and future planned work measuring rural internet connectivity and developing a content sharing tool. The author conducted a pilot study measuring internet performance at a community radio station which found DNS lookup latency around 110ms and download throughput varying significantly between 248-731kbps. Future work will include deploying measurement devices to further analyze rural internet access technologies and service providers.
PERFORMANCE EVALUATION OF H.265/MPEG-HEVC, VP9 AND H.264/MPEGAVC VIDEO CODINGijma
This study evaluates the performance of the three latest video codecs H.265/MPEG-HEVC, H.264/MPEGAVC
and VP9. The evaluation is based on both subjective and objective quality metrics. The assessment
metric Double Stimulus Impairment Scale (DSIS) is used to evaluate the subjective quality of the
compressed video sequences. The Peak Signal-to-Noise Ratio (PSNR) metricis used for the objective
evaluation. Moreover, this work studies the effect of frame rate and resolution on the encoders’
performance. The extensive number of experiments are conducted with similar encoding configurations for
the three studied encoders. The evaluation results show that H.265/MPEG-HEVC provides superior bitrate
saving capabilities compared to H.264 and VP9. However, VP9 shows lower encoding time than
H.265/MPEG-HEVC but higher encoding time compared to H.264.
FREQUENCY AND TIME DOMAIN PACKET SCHEDULING BASED ON CHANNEL PREDICTION WITH ...ijwmn
1) The document discusses packet scheduling algorithms for LTE downlink systems that must operate under imperfect channel quality information (CQI) due to errors, delays, and other issues.
2) It proposes a new packet scheduling algorithm that uses a Kalman filter-based channel predictor in the frequency domain to estimate the true CQI from erroneous feedback, combined with a time domain grouping technique using proportional fair and modified largest weighted delay first algorithms.
3) Simulation results showed this approach achieves better performance than existing algorithms in terms of system throughput and packet loss ratio under imperfect CQI conditions.
Grant free IoT, Ericsson Research Presentationamin azari
The document outlines Amin Azari's research on grant-free radio access for IoT communications. It discusses several parts: (1) introduction and motivation, (2) coexistence analysis including interference and coverage modeling, (3) deployment and operation strategies with analytical modeling of reliability and cost, and (4) proposed protocol design and performance evaluation. The research aims to analyze the benefits of grant-free access for IoT, optimize network parameters, and design an effective protocol for asynchronous transmissions with random frequency offsets.
COMPARING VARIOUS CHANNEL ESTIMATION TECHNIQUES FOR OFDM SYSTEMS USING MATLABijwmn
This document compares the performance of five channel estimation techniques for OFDM systems over quasi-static channels using MATLAB simulations. The techniques evaluated are: decision directed (DD), linear interpolation, second-order interpolation, discrete Fourier transform (DFT) interpolation, and minimum mean square error (MMSE) interpolation. Simulation results show that DD channel estimation provides the lowest bit-error rate and mean square error, but has extra processing delay and is sensitive to channel variations. MMSE interpolation outperforms the other interpolation techniques.
Prashant Desai has over 8 years of experience designing and developing software for telecom and consumer electronics products. Some of his areas of expertise include VoIP protocols, IPTV solutions, multimedia streaming, and network protocols. He has worked on projects involving peer-to-peer video telephony, media servers, set-top boxes, home gateways, and telecom equipment. Prashant is proficient in C/C++ and has experience with protocols such as SIP, RTP, HTTP, and network stacks.
Improved voice quality with the combination of transport layer & audio codec ...journalBEEI
Improving voice quality over wireless communication becomes a demanding feature for social media apps like facebook, whatsapp and other communication channels. Voice-over-internet protocol (VoIP) helps us to make quick telephone calls over the internet. It includes various mechanism which are signaling, controlling and transport layer. Over wireless links, packet loss and high transmission delay damage voice quality. Here VoIP quality will be measured by three main elements which are signaling protocol, audio codec and transport layer. To improve the overall voice quality, we need to combine these three elements properly to get the best score. Otherwise perceptual speech quality will not be the right tool to measure the voice quality. Here we will use Mean Opinion Score (MOS) for calculated jitter values and end to end delay. At the end, best combination of audio codec & signaling protocol produced the quality speech.
Multicasting Of Adaptively-Encoded MPEG4 Over Qos-Cognizant IP NetworksEditor IJMTER
we propose a novel architectural planning for multicasting of adaptively-encoded
layered MPEG4 over a QoS-aware IP network. We re-quire a QoS-aware IP network in this case to
(1) Support priority dropping of packets in time of congestion. (2) Provide congestion notification to
the multicast sender. For the first requirement, we use RED's extension for service differentiation. It
recognizes the priority of packets when they need to be dropped and drops lower priority packets
first. We couple RED with our proposal for the second requirement which is the adoption of
Backward Explicit Congestion Notification (BECN) for use with IP multicast. BECN will provide
early congestion notification at the IP layer level to the video sender. BECN detects upcoming
congestion based on size of the RED queue in the routers. The MPEG4 adaptive-encoder can change
the sending rate and also can divide the video packets into lower priority packets and high priority
packets. Based on BECN messages from the routers, a simple flow controller at the sender sets the
rate for the adaptive MPEG4 encoder and also sets the ratio between the high priority and low
priority packets within the video stream. We use a TES model for generating the MPEG4 traffic that
is based on real video traces. Simulation results show that combining priority dropping, MPEG4
adaptive encoding, and multicast BECN: (1) Improves bandwidth utilization (2) Reduces time to
react to congestion and hence improves the received video quality (3) Maintains graceful degradation
in quality with congestion and provides minimum quality even if congestion persists.
05 comparative study of voice print based acoustic features mfcc and lpccIJAEMSJORNAL
Voice is the best biometric feature for investigation and authentication. It has both biological and behavioural features. The acoustic features are related to the voice. The Speaker Recognition System is designed for the automatic authentication of speaker’s identity which is truly based on the human’s voice. Mel Frequency Cepstrum coefficient (MFCC) and Linear Prediction Cepstrum coefficient (LPCC) are taken in use for feature extraction from the provided voice sample. This paper provides a comparative study of MFCC and LPCC based on the accuracy of results and their working methodology. The results are better if MFCC is used for feature extraction.
FPGA Implementation of LDPC Encoder for Terrestrial TelevisionAI Publications
The increasing data rates in digital television networks increase the demands on data capacity of the current transmission channels. Through new standards, the capacity of existing channels is increased with new methods of error correction coding and modulation. In this work, Low Density Parity Check (LDPC) codes are implemented for their error correcting capability. LDPC is a linear error correcting code. These linear error correcting codes are used for transmitting a message over a noisy transmission channel. LDPC codes are finding increasing use in applications requiring reliable and highly efficient information transfer over noisy channels. These codes are capable of performing near to Shannon limit performance and have low decoding complexity. LDPC uses parity check matrix for its encoding and decoding purpose. The main advantage of the parity check matrix is that it helps in detecting and correcting errors which is a very important advantage against noisy channels. This work presents the design and implementation of a LDPC encoder for transmission of digital terrestrial television according to the Chinese DTMB standard. The system is written in Verilog and is implemented on FPGA. The whole work is then verified with the help of Matlab modelling.
V. Srinivas Rao is a software professional with over 6 years of experience developing protocols like LTE, TCP/IP, and BGP. He has worked on projects involving mobile applications, identity management services, and the simulation of LTE protocol stacks. Rao holds a B.Tech in Electronics and Communication Engineering and has expertise in C/C++, UNIX, Linux, and networking protocols. He is interested in telecom software development, embedded design, and verification.
Novel Approach for Evaluating Video Transmission using Combined Scalable Vide...IJECEIAES
One of the main problems in video transmission is the bandwidth fluctuation in wireless channel. Therefore, there is an urgent need to find an efficient bandwidth utilization and method. This research utilizes the Combined Scalable Video Coding (CSVC) which comes from Joint Scalable Video Model (JSVM). In the combined scalable video coding, we implement Coarse Grain Scalability (CGS) and Medium Grain Scalability (MGS). We propose a new scheme in which it can be implemented on Network Simulator II (NS-2) over wireless broadband network. The advantages of this new scheme over the other schemes are more realistic and based on open source program. The result shows that CSVC implementation on MGS mode outperforms CGS mode.
Flexible fec and fair selection of scalable units for high quality vod stream...IJMIT JOURNAL
Providing high quality video on demand (VoD) streaming service over wireless networks is very challenging due to the limited capacity and error-proneness of the wireless environment. We propose a flexible forward error correction (FEC) and a fair selection scheme of scalable units that utilize a layered
coding structure of H.264/SVC (scalable video coding). Three error-resilient techniques (e.g., unequal
error protection, FEC, and retransmission) are adapted to minimize the total distortion of VoD streaming
service. For flexible FEC, a rateless FEC code is adopted. The FEC code rates are based on the possible
number of retransmission, the condition of the wireless channel and the layered coding structure of H.264/SVC for each packet. A theoretical study is performed to show how to utilize the possible number of retransmission for an adaptive FEC code rate. With fair selection, regular and retransmission-requested
packets compete for resources without fixing the retry limit. Thus, excessive retransmission is prevented and
the proposed scheme effectively provides capacity-limited and delay-constrained VoD streaming services.
For this fair selection of scalable units, we formulate the problem using binary integer programming and
propose an effective low complexity selection algorithm based on a priority index. The proposed algorithm
prioritizes packets according to the priority index and the H.264/SVC structure. We show that the proposed
scheme can minimize the total video distortion compared to other heuristic procedures. Other effects of the
various factors are also considered for the performance of the new scheme.
FLEXIBLE FEC AND FAIR SELECTION OF SCALABLE UNITS FOR HIGH QUALITY VOD STREAM...IJMIT JOURNAL
Providing high quality video on demand (VoD) streaming service over wireless networks is very
challenging due to the limited capacity and error-proneness of the wireless environment. We propose a
flexible forward error correction (FEC) and a fair selection scheme of scalable units that utilize a layered
coding structure of H.264/SVC (scalable video coding). Three error-resilient techniques (e.g., unequal
error protection, FEC, and retransmission) are adapted to minimize the total distortion of VoD streaming
service. For flexible FEC, a rateless FEC code is adopted. The FEC code rates are based on the possible
number of retransmission, the condition of the wireless channel and the layered coding structure of
H.264/SVC for each packet. A theoretical study is performed to show how to utilize the possible number of
retransmission for an adaptive FEC code rate. With fair selection, regular and retransmission-requested
packets compete for resources without fixing the retry limit. Thus, excessive retransmission is prevented and
the proposed scheme effectively provides capacity-limited and delay-constrained VoD streaming services.
For this fair selection of scalable units, we formulate the problem using binary integer programming and
propose an effective low complexity selection algorithm based on a priority index. The proposed algorithm
prioritizes packets according to the priority index and the H.264/SVC structure. We show that the proposed
scheme can minimize the total video distortion compared to other heuristic procedures. Other effects of the
various factors are also considered for the performance of the new scheme
Robust audio watermarking based on transform domain and SVD with compressive ...TELKOMNIKA JOURNAL
The growth of the internet and digital data has resulted forgery, modification and sharing of digital data without property rights. Audio watermarking is one of a solution to protect the copyright of an audio from copyright infringement. This paper proposes an audio watermarking method which is robust against attacks and high capacity. First, a synchronization bit is added to the audio host. After the audio host is decomposed by Lifting Wavelet Transform (LWT), then choose a subband from the output of LWT to be transformed by discrete cosine transform (DCT). Next, the matrix of the signal from DCT is selected for the singular value decomposition (SVD) process, so that is obtained U, S and V matrix. S matrix is embedded with the watermark. Before the embedding process, the watermark image is compressed by Compressive Sampling. The results show that the proposed watermarking system is highly robust against a kind attack of LPF, resampling, and linear speed change which is proven by its BER is zero.
A Channel Allocation Algorithm for Cognitive Radio Users Based on Channel Sta...Alpen-Adria-Universität
Cognitive radio networks by utilizing the spectrum holes in licensed frequency bands are able to efficiently manage the radio spectrum. A significant improvement in spectrum use can be achieved by giving secondary users access to these spectrum holes. Predicting spectrum holes can save significant energy that is consumed to detect spectrum holes. This is because the secondary users can only select the channels that are predicted to be idle channels. However, collisions can occur either between a primary user and secondary users or among the secondary users themselves. This paper introduces a centralized channel allocation algorithm in a scenario with multiple secondary users to control both primary and secondary collisions. The proposed allocation algorithm, which uses a channel status predictor, provides a good performance with fairness among the secondary users while they have the minimal interference with the primary user. The simulation results show that the probability of a wrong prediction of an idle channel state in a multi-channel system is less than 0.9%. In addition, the channel state prediction saves the sensing energy up to 73%, and the utilization of the spectrum can be improved more than 77%.
An Effective Approach for Colour Image Transmission using DWT Over OFDM for B...IJMTST Journal
Image transmission over the fading channels without degrading the perceptual quality is a challenging task while mitigating the power consumption in many fields such as broadband networks, mobile communications, Image sharing and video broadcasting. Also, it is not possible to resend the lost packets every time in many applications such as video broadcasting. Here, an effective approach for color image transmission has been proposed with power saving approach over OFDM system. Experimental results shows that the reception quality of received image is good enough with various peak signal to noise ratios also saved 60% of energy.
An Empirical Evaluation of VoIP Playout Buffer Dimensioning in Skype, Google ...Academia Sinica
VoIP playout buffer dimensioning has long been a challeng- ing optimization problem, as the buffer size must maintain a balance between conversational interactivity and speech quality. The conversational quality may be affected by a number of factors, some of which may change over time. Although a great deal of research effort has been expended in trying to solve the problem, how the research results are applied in practice is unclear.
In this paper, we investigate the playout buffer dimension- ing algorithms applied in three popular VoIP applications, namely, Skype, Google Talk, and MSN Messenger. We conduct experiments to assess how the applications adjust their playout buffer sizes. Using an objective QoE (Quality of Experience) metric, we show that Google Talk and MSN Messenger do not adjust their respective buffer sizes appropriately, while Skype does not adjust its buffer at all. In other words, they could provide better QoE to users by improving their buffer dimensioning algorithms. Moreover, none of the applications adapts its buffer size to the network loss rate, which should also be considered to ensure optimal QoE provisioning.
Review of video over IP testing tools including: video syntax analyzer, pixel based measurement indexes like PSNR and SSIM and the tools to measure them, IP based video quality testing.
QoS Constrained H.264/SVC video streaming over Multicast Ad Hoc NetworksIJERA Editor
Support for QoS enabled multimedia transmission over multicast ad hoc network is necessary these days.
Researchers have developed various encoding/decoding schemes which can efficiently deliver the multimedia
contents over wireless networks. In case of ad hoc networks, performance of routing protocol depends upon
different factors i.e. traffic type being used for wireless transmission, dynamic network behavior, bandwidth and
computational power of nodes etc. It is essential to investigate the performance of multicast routing protocol
using various data types because they may consume huge network resources thus results in degradation of
transmission quality. In case of multicast group communication, Audio/Video data stream can cause extra
overhead on network performance and it is quite difficult to maintain Quality of Services for such type of data.
H.264 offers a rich codec library for Scalable Video Coding, to transfer SVC video traffic efficiently over
wireless networks. In this paper, we will analyze the performance of MAODV and PUMA routing protocols
using H.264/SVC video streaming traffic under the various QoS constraints such as Throughput, PDR, Delay,
Routing Load and Jitter etc.
There is a massive growth in mobile video consumption which outpaces the capacity improvements in next generation mobile networks. Specifically, mobile network operators face the challenge of allocating the scarce wireless resources while maximizing the user quality of experience (QoE). The first part of this talk addresses the main challenges in uplink distribution of user-generated video content over fourth generation mobile networks. The second part explores the benefit of QoE-based traffic and resource management in the mobile network in the context of adaptive HTTP downlink video delivery.
The NGMN Alliance provides the following summary:
- The NGMN Alliance includes over 200 mobile network operators serving over 60% of global customers and works with vendors and other partners to advance 5G network technologies.
- The NGMN 5G work program focuses on areas like end-to-end architecture, security, spectrum, and trials and testing to help develop 5G standards and evaluate technologies.
- Recent deliverables include contributions to 3GPP on service-based architecture, recommendations to improve patent declarations, and a report on base station antenna standards. The work program is expanding its focus on security, architecture, and spectrum issues.
FREQUENCY AND TIME DOMAIN PACKET SCHEDULING BASED ON CHANNEL PREDICTION WITH ...ijwmn
1) The document discusses packet scheduling algorithms for LTE downlink systems that must operate under imperfect channel quality information (CQI) due to errors, delays, and other issues.
2) It proposes a new packet scheduling algorithm that uses a Kalman filter-based channel predictor in the frequency domain to estimate the true CQI from erroneous feedback, combined with a time domain grouping technique using proportional fair and modified largest weighted delay first algorithms.
3) Simulation results showed this approach achieves better performance than existing algorithms in terms of system throughput and packet loss ratio under imperfect CQI conditions.
Grant free IoT, Ericsson Research Presentationamin azari
The document outlines Amin Azari's research on grant-free radio access for IoT communications. It discusses several parts: (1) introduction and motivation, (2) coexistence analysis including interference and coverage modeling, (3) deployment and operation strategies with analytical modeling of reliability and cost, and (4) proposed protocol design and performance evaluation. The research aims to analyze the benefits of grant-free access for IoT, optimize network parameters, and design an effective protocol for asynchronous transmissions with random frequency offsets.
COMPARING VARIOUS CHANNEL ESTIMATION TECHNIQUES FOR OFDM SYSTEMS USING MATLABijwmn
This document compares the performance of five channel estimation techniques for OFDM systems over quasi-static channels using MATLAB simulations. The techniques evaluated are: decision directed (DD), linear interpolation, second-order interpolation, discrete Fourier transform (DFT) interpolation, and minimum mean square error (MMSE) interpolation. Simulation results show that DD channel estimation provides the lowest bit-error rate and mean square error, but has extra processing delay and is sensitive to channel variations. MMSE interpolation outperforms the other interpolation techniques.
Prashant Desai has over 8 years of experience designing and developing software for telecom and consumer electronics products. Some of his areas of expertise include VoIP protocols, IPTV solutions, multimedia streaming, and network protocols. He has worked on projects involving peer-to-peer video telephony, media servers, set-top boxes, home gateways, and telecom equipment. Prashant is proficient in C/C++ and has experience with protocols such as SIP, RTP, HTTP, and network stacks.
Improved voice quality with the combination of transport layer & audio codec ...journalBEEI
Improving voice quality over wireless communication becomes a demanding feature for social media apps like facebook, whatsapp and other communication channels. Voice-over-internet protocol (VoIP) helps us to make quick telephone calls over the internet. It includes various mechanism which are signaling, controlling and transport layer. Over wireless links, packet loss and high transmission delay damage voice quality. Here VoIP quality will be measured by three main elements which are signaling protocol, audio codec and transport layer. To improve the overall voice quality, we need to combine these three elements properly to get the best score. Otherwise perceptual speech quality will not be the right tool to measure the voice quality. Here we will use Mean Opinion Score (MOS) for calculated jitter values and end to end delay. At the end, best combination of audio codec & signaling protocol produced the quality speech.
Multicasting Of Adaptively-Encoded MPEG4 Over Qos-Cognizant IP NetworksEditor IJMTER
we propose a novel architectural planning for multicasting of adaptively-encoded
layered MPEG4 over a QoS-aware IP network. We re-quire a QoS-aware IP network in this case to
(1) Support priority dropping of packets in time of congestion. (2) Provide congestion notification to
the multicast sender. For the first requirement, we use RED's extension for service differentiation. It
recognizes the priority of packets when they need to be dropped and drops lower priority packets
first. We couple RED with our proposal for the second requirement which is the adoption of
Backward Explicit Congestion Notification (BECN) for use with IP multicast. BECN will provide
early congestion notification at the IP layer level to the video sender. BECN detects upcoming
congestion based on size of the RED queue in the routers. The MPEG4 adaptive-encoder can change
the sending rate and also can divide the video packets into lower priority packets and high priority
packets. Based on BECN messages from the routers, a simple flow controller at the sender sets the
rate for the adaptive MPEG4 encoder and also sets the ratio between the high priority and low
priority packets within the video stream. We use a TES model for generating the MPEG4 traffic that
is based on real video traces. Simulation results show that combining priority dropping, MPEG4
adaptive encoding, and multicast BECN: (1) Improves bandwidth utilization (2) Reduces time to
react to congestion and hence improves the received video quality (3) Maintains graceful degradation
in quality with congestion and provides minimum quality even if congestion persists.
05 comparative study of voice print based acoustic features mfcc and lpccIJAEMSJORNAL
Voice is the best biometric feature for investigation and authentication. It has both biological and behavioural features. The acoustic features are related to the voice. The Speaker Recognition System is designed for the automatic authentication of speaker’s identity which is truly based on the human’s voice. Mel Frequency Cepstrum coefficient (MFCC) and Linear Prediction Cepstrum coefficient (LPCC) are taken in use for feature extraction from the provided voice sample. This paper provides a comparative study of MFCC and LPCC based on the accuracy of results and their working methodology. The results are better if MFCC is used for feature extraction.
FPGA Implementation of LDPC Encoder for Terrestrial TelevisionAI Publications
The increasing data rates in digital television networks increase the demands on data capacity of the current transmission channels. Through new standards, the capacity of existing channels is increased with new methods of error correction coding and modulation. In this work, Low Density Parity Check (LDPC) codes are implemented for their error correcting capability. LDPC is a linear error correcting code. These linear error correcting codes are used for transmitting a message over a noisy transmission channel. LDPC codes are finding increasing use in applications requiring reliable and highly efficient information transfer over noisy channels. These codes are capable of performing near to Shannon limit performance and have low decoding complexity. LDPC uses parity check matrix for its encoding and decoding purpose. The main advantage of the parity check matrix is that it helps in detecting and correcting errors which is a very important advantage against noisy channels. This work presents the design and implementation of a LDPC encoder for transmission of digital terrestrial television according to the Chinese DTMB standard. The system is written in Verilog and is implemented on FPGA. The whole work is then verified with the help of Matlab modelling.
V. Srinivas Rao is a software professional with over 6 years of experience developing protocols like LTE, TCP/IP, and BGP. He has worked on projects involving mobile applications, identity management services, and the simulation of LTE protocol stacks. Rao holds a B.Tech in Electronics and Communication Engineering and has expertise in C/C++, UNIX, Linux, and networking protocols. He is interested in telecom software development, embedded design, and verification.
Novel Approach for Evaluating Video Transmission using Combined Scalable Vide...IJECEIAES
One of the main problems in video transmission is the bandwidth fluctuation in wireless channel. Therefore, there is an urgent need to find an efficient bandwidth utilization and method. This research utilizes the Combined Scalable Video Coding (CSVC) which comes from Joint Scalable Video Model (JSVM). In the combined scalable video coding, we implement Coarse Grain Scalability (CGS) and Medium Grain Scalability (MGS). We propose a new scheme in which it can be implemented on Network Simulator II (NS-2) over wireless broadband network. The advantages of this new scheme over the other schemes are more realistic and based on open source program. The result shows that CSVC implementation on MGS mode outperforms CGS mode.
Flexible fec and fair selection of scalable units for high quality vod stream...IJMIT JOURNAL
Providing high quality video on demand (VoD) streaming service over wireless networks is very challenging due to the limited capacity and error-proneness of the wireless environment. We propose a flexible forward error correction (FEC) and a fair selection scheme of scalable units that utilize a layered
coding structure of H.264/SVC (scalable video coding). Three error-resilient techniques (e.g., unequal
error protection, FEC, and retransmission) are adapted to minimize the total distortion of VoD streaming
service. For flexible FEC, a rateless FEC code is adopted. The FEC code rates are based on the possible
number of retransmission, the condition of the wireless channel and the layered coding structure of H.264/SVC for each packet. A theoretical study is performed to show how to utilize the possible number of retransmission for an adaptive FEC code rate. With fair selection, regular and retransmission-requested
packets compete for resources without fixing the retry limit. Thus, excessive retransmission is prevented and
the proposed scheme effectively provides capacity-limited and delay-constrained VoD streaming services.
For this fair selection of scalable units, we formulate the problem using binary integer programming and
propose an effective low complexity selection algorithm based on a priority index. The proposed algorithm
prioritizes packets according to the priority index and the H.264/SVC structure. We show that the proposed
scheme can minimize the total video distortion compared to other heuristic procedures. Other effects of the
various factors are also considered for the performance of the new scheme.
FLEXIBLE FEC AND FAIR SELECTION OF SCALABLE UNITS FOR HIGH QUALITY VOD STREAM...IJMIT JOURNAL
Providing high quality video on demand (VoD) streaming service over wireless networks is very
challenging due to the limited capacity and error-proneness of the wireless environment. We propose a
flexible forward error correction (FEC) and a fair selection scheme of scalable units that utilize a layered
coding structure of H.264/SVC (scalable video coding). Three error-resilient techniques (e.g., unequal
error protection, FEC, and retransmission) are adapted to minimize the total distortion of VoD streaming
service. For flexible FEC, a rateless FEC code is adopted. The FEC code rates are based on the possible
number of retransmission, the condition of the wireless channel and the layered coding structure of
H.264/SVC for each packet. A theoretical study is performed to show how to utilize the possible number of
retransmission for an adaptive FEC code rate. With fair selection, regular and retransmission-requested
packets compete for resources without fixing the retry limit. Thus, excessive retransmission is prevented and
the proposed scheme effectively provides capacity-limited and delay-constrained VoD streaming services.
For this fair selection of scalable units, we formulate the problem using binary integer programming and
propose an effective low complexity selection algorithm based on a priority index. The proposed algorithm
prioritizes packets according to the priority index and the H.264/SVC structure. We show that the proposed
scheme can minimize the total video distortion compared to other heuristic procedures. Other effects of the
various factors are also considered for the performance of the new scheme
Robust audio watermarking based on transform domain and SVD with compressive ...TELKOMNIKA JOURNAL
The growth of the internet and digital data has resulted forgery, modification and sharing of digital data without property rights. Audio watermarking is one of a solution to protect the copyright of an audio from copyright infringement. This paper proposes an audio watermarking method which is robust against attacks and high capacity. First, a synchronization bit is added to the audio host. After the audio host is decomposed by Lifting Wavelet Transform (LWT), then choose a subband from the output of LWT to be transformed by discrete cosine transform (DCT). Next, the matrix of the signal from DCT is selected for the singular value decomposition (SVD) process, so that is obtained U, S and V matrix. S matrix is embedded with the watermark. Before the embedding process, the watermark image is compressed by Compressive Sampling. The results show that the proposed watermarking system is highly robust against a kind attack of LPF, resampling, and linear speed change which is proven by its BER is zero.
A Channel Allocation Algorithm for Cognitive Radio Users Based on Channel Sta...Alpen-Adria-Universität
Cognitive radio networks by utilizing the spectrum holes in licensed frequency bands are able to efficiently manage the radio spectrum. A significant improvement in spectrum use can be achieved by giving secondary users access to these spectrum holes. Predicting spectrum holes can save significant energy that is consumed to detect spectrum holes. This is because the secondary users can only select the channels that are predicted to be idle channels. However, collisions can occur either between a primary user and secondary users or among the secondary users themselves. This paper introduces a centralized channel allocation algorithm in a scenario with multiple secondary users to control both primary and secondary collisions. The proposed allocation algorithm, which uses a channel status predictor, provides a good performance with fairness among the secondary users while they have the minimal interference with the primary user. The simulation results show that the probability of a wrong prediction of an idle channel state in a multi-channel system is less than 0.9%. In addition, the channel state prediction saves the sensing energy up to 73%, and the utilization of the spectrum can be improved more than 77%.
An Effective Approach for Colour Image Transmission using DWT Over OFDM for B...IJMTST Journal
Image transmission over the fading channels without degrading the perceptual quality is a challenging task while mitigating the power consumption in many fields such as broadband networks, mobile communications, Image sharing and video broadcasting. Also, it is not possible to resend the lost packets every time in many applications such as video broadcasting. Here, an effective approach for color image transmission has been proposed with power saving approach over OFDM system. Experimental results shows that the reception quality of received image is good enough with various peak signal to noise ratios also saved 60% of energy.
An Empirical Evaluation of VoIP Playout Buffer Dimensioning in Skype, Google ...Academia Sinica
VoIP playout buffer dimensioning has long been a challeng- ing optimization problem, as the buffer size must maintain a balance between conversational interactivity and speech quality. The conversational quality may be affected by a number of factors, some of which may change over time. Although a great deal of research effort has been expended in trying to solve the problem, how the research results are applied in practice is unclear.
In this paper, we investigate the playout buffer dimension- ing algorithms applied in three popular VoIP applications, namely, Skype, Google Talk, and MSN Messenger. We conduct experiments to assess how the applications adjust their playout buffer sizes. Using an objective QoE (Quality of Experience) metric, we show that Google Talk and MSN Messenger do not adjust their respective buffer sizes appropriately, while Skype does not adjust its buffer at all. In other words, they could provide better QoE to users by improving their buffer dimensioning algorithms. Moreover, none of the applications adapts its buffer size to the network loss rate, which should also be considered to ensure optimal QoE provisioning.
Review of video over IP testing tools including: video syntax analyzer, pixel based measurement indexes like PSNR and SSIM and the tools to measure them, IP based video quality testing.
QoS Constrained H.264/SVC video streaming over Multicast Ad Hoc NetworksIJERA Editor
Support for QoS enabled multimedia transmission over multicast ad hoc network is necessary these days.
Researchers have developed various encoding/decoding schemes which can efficiently deliver the multimedia
contents over wireless networks. In case of ad hoc networks, performance of routing protocol depends upon
different factors i.e. traffic type being used for wireless transmission, dynamic network behavior, bandwidth and
computational power of nodes etc. It is essential to investigate the performance of multicast routing protocol
using various data types because they may consume huge network resources thus results in degradation of
transmission quality. In case of multicast group communication, Audio/Video data stream can cause extra
overhead on network performance and it is quite difficult to maintain Quality of Services for such type of data.
H.264 offers a rich codec library for Scalable Video Coding, to transfer SVC video traffic efficiently over
wireless networks. In this paper, we will analyze the performance of MAODV and PUMA routing protocols
using H.264/SVC video streaming traffic under the various QoS constraints such as Throughput, PDR, Delay,
Routing Load and Jitter etc.
There is a massive growth in mobile video consumption which outpaces the capacity improvements in next generation mobile networks. Specifically, mobile network operators face the challenge of allocating the scarce wireless resources while maximizing the user quality of experience (QoE). The first part of this talk addresses the main challenges in uplink distribution of user-generated video content over fourth generation mobile networks. The second part explores the benefit of QoE-based traffic and resource management in the mobile network in the context of adaptive HTTP downlink video delivery.
The NGMN Alliance provides the following summary:
- The NGMN Alliance includes over 200 mobile network operators serving over 60% of global customers and works with vendors and other partners to advance 5G network technologies.
- The NGMN 5G work program focuses on areas like end-to-end architecture, security, spectrum, and trials and testing to help develop 5G standards and evaluate technologies.
- Recent deliverables include contributions to 3GPP on service-based architecture, recommendations to improve patent declarations, and a report on base station antenna standards. The work program is expanding its focus on security, architecture, and spectrum issues.
This document provides an overview of the activities of ITU-T Study Group 12 related to quality of service (QoS) and quality of experience (QoE). It discusses SG12's mandate to develop recommendations on performance, QoS, and QoE for networks and services. It outlines SG12's history and key events, lists its recommendations, and provides details on its working parties, groups, and work program which covers topics such as terminal performance, speech quality, multimedia quality assessment, and QoE for various network types and applications.
Ingrid Moerman, Stefan Bouckaert: IP CREW - Cognitive Radio Experimentation ...FIA2010
The document summarizes the IP CREW Cognitive Radio Experimentation World project. The project aims to establish an open federated test platform to facilitate experimentally-driven research on advanced spectrum sensing, cognitive radio, cognitive networking, and spectrum sharing in licensed and unlicensed bands. The platform will federate existing heterogeneous wireless testbeds and augment them with state-of-the-art cognitive sensing platforms. This will allow experimental evaluation and comparison of cognitive radio and cognitive networking solutions in a controlled and repeatable environment.
1) The document discusses a research project called Q-SAC that aimed to enable service providers to ensure video quality through remote configuration of appropriate network policies.
2) Key findings from the project were contributed to standardization bodies like CableLabs and the UPnP Forum to develop specifications for quality of service (QoS).
3) The research organization helped contribute to market adoption by developing reference implementations, testing suites, and showcasing technologies at events with industrial partners.
QoE- and Energy-aware Content Consumption for HTTP Adaptive StreamingDanieleLorenzi6
Video streaming services account for the majority of today’s traffic on the Internet, and according to recent studies, this share is expected to continue growing. This implies that many people around the globe utilize video streaming services on a daily basis to fruit video content. Given this broad utilization, research in video streaming is recently moving towards energy-aware approaches, which aim at the minimization of the energy consumption of the devices involved. On the other side, the perception of quality delivered to the user plays an important role, and the advent of HTTP Adaptive Streaming (HAS) changed the way quality is perceived. The focus moved from the Quality of Service (QoS) towards the Quality of Experience (QoE) of the user taking part in the streaming session. Therefore video streaming services need to develop Adaptive BitRate (ABR) techniques to deal with different network environments on the client side or appropriate end-to-end strategies to provide high QoE to the users. The scope of this doctoral study is within the end-to-end environment with a focus on the end-users domain, referred to as the player environment, including video content consumption and interactivity. This thesis aims to investigate and develop different techniques to increase the delivered QoE to the users and reduce the energy consumption of the end devices in HAS context. We present four main research questions to target the related challenges in the domain of content consumption for HAS systems.
This document discusses two real-time streaming protocols - Real Time Streaming Protocol (RTSP) and Web Real-Time Communication (WebRTC). It presents a comparative study of these two protocols by developing new mobile applications that implement each protocol on Android. The applications are analyzed based on connection establishment time and stream reception time, which influence quality of service. The applications are also compared to popular video streaming apps to evaluate stream packet delay, with results showing the WebRTC implementation improves upon other apps.
This document summarizes a study on the usability and safety of searching for radio stations using in-vehicle Digital Audio Broadcasting (DAB) devices. The study tested different strategies for searching stations based on station name, genre, and sub-genre using a driving simulator. Results showed searching by sub-genre took less time and eye fixation than other methods. User questionnaires also indicated DAB radio was perceived as slightly more satisfying to use than traditional FM radio. The study concluded that categorizing stations by genre and sub-genre can help users search more easily and safely while driving, and recommended further testing with more participants.
Evaluation of Multiplexing and Buffer Policies Influence on VoIP Conversation...Jose Saldana
Jose Saldana, Jenifer Murillo, Julian Fernandez-Navajas, Jose Ruiz-Mas, Eduardo Viruete, Jose I. Aznar. "Evaluation of Multiplexing and Buffer Policies Influence on VoIP Conversation Quality" . In Proc. CCNC 2011- 3rd IEEE International Workshop on Digital Entertainment, Networked Virtual Environments, and Creative Technology, pp 1147-1151, Las Vegas. Jan. 2011. ISBN 9781424487882.
1) The document discusses a proposed Channel Bonding Dynamic Bandwidth Allocation (CB-DBA) scheme for next generation passive optical networks.
2) The CB-DBA scheme is compared to existing DBA schemes and is shown to improve performance in terms of upstream delay and better bandwidth utilization at higher traffic loads.
3) Simulation results indicate that the CB-DBA scheme achieves bandwidth improvement, lower delay, and higher data rates compared to existing solutions.
Internet2: VoIP Phone Codec Testing White PaperJoshua Reola
This document summarizes a study that tested the effects of packet loss on voice quality for various Voice over IP (VoIP) codecs. Key findings include:
- The G.711 codec maintained acceptable audio quality (MOS >3.0) up to 5% packet loss, more than typically claimed by vendors.
- The G.729 codec degraded more quickly above 5% packet loss.
- The GIPS codec significantly outperformed standard codecs, maintaining quality up to 15% packet loss.
- There were no significant differences between vendors in codec performance. The study recommends the G.711 codec for networks with ample bandwidth and considering the GIPS codec. Further research on additional codecs and factors like packet jitter was advised
FSO: Efficient Connectivity Solution for Campus Area NetworkLaraib Khan
This project proposes using a free space optics (FSO) system as an efficient connectivity solution for the campus area network of the University of Sindh. The document compares FSO to fiber optic networks and finds that FSO has lower installation costs and time while providing high bandwidth connectivity. Simulations show that FSO links can achieve data rates similar to fiber optics over distances up to 2km before signal degradation becomes an issue. Cost estimates indicate that an FSO system would cost approximately 70% less than a comparable fiber optic network for the University of Sindh campus. The project aims to increase awareness of FSO as a feasible alternative to fiber optics for shorter-distance, high-speed data communication in Pakistan.
This document provides an application progress report for research on evaluating the performance of LTE-Advanced (LTE-A) networks for quality of service (QoS) in internet protocol television (IPTV) from June 2019 to December 2019. The researcher introduces LTE and QoS, describes the LTE architecture, and states the problem of ensuring best video quality for users. Observations from simulations on cell data and extracted data are summarized. Next steps include writing a research paper.
This document summarizes the outcomes of the ITU-T Study Group 16 meeting held from 16-27 January 2017 in Geneva. It discusses the number of contributions reviewed and recommendations approved. It provides details on the collaborative work done with other groups on topics like video coding, IPTV, accessibility, e-health, and more. Major accomplishments included completing new recommendations on scalable vector graphics, speech translation services, and requirements for areas like vehicle gateways and content delivery networks. Work also progressed on developing standards for immersive media, visual surveillance, and blockchain technologies.
Achievements and future works of ITU-T Study Group 11 on Signalling requirements, protocols and test specifications
Presented at WTSA-16 by Mr Kaoru Kenyoshi, Vice-Chairman, on behalf of Mr Wei Feng, Chairman of of ITU-T Study Group 11
Analysis of Impact of Channel Error Rate on Average PSNR in Multimedia TrafficIOSR Journals
Abstract : The performance of the multimedia traffic in Ad-Hoc networks is highly impacted with the Signal to Noise Ratio. The Average PSNR (Peak Signal to Noise Ratio) is an important parameter for the evaluation of multimedia traffic in Ad-Hoc Networks. With the increase of bandwidth of the channels, it becomes necessary to take care of other network parameters like PSNR and ASNR( Average Signal to Noise Ratio) .Enhanced bandwidth with higher channel error rates demand a careful analysis of signal to noise ratio for optimum performance. In this paper, we have evaluated the effect of channel error rate on Average PSNR for the MPEG-4 traffic in Ad-hoc Networks. Keywords: MANETs, Evalvid, MPEG-4, Fragmentation, PSNR
This document reports on user trials conducted for the LinkedTV project scenarios. University of St Gallen conducted interviews with 11 industry experts to evaluate the LinkedTV concept and scenarios. User trials were also held at Sound and Vision and CWI to gather feedback from viewers. The trials provided insight into expectations for enriched television experiences and identified challenges around additional information, annotations, personalization, and privacy. Based on trial results, the Interactive News, Hyperlinked Documentary, and Media Arts scenarios were revised to address issues such as simplifying the user interface and making additional content more compact.
The document proposes an adaptive high-quality video service for network-based multi-party collaboration using the Access Grid platform. It describes designing an Access Grid Media Architecture to enable digital video and high-definition video support with application-level quality of service controls. A prototype was implemented and tested on a test bed to demonstrate improved user quality of experience through a one-to-many network adaptation scheme that conceals quality variations from network problems. Evaluation results showed the adaptive video service was able to support high-quality video streams while adapting to changing network conditions.
Similar to Highlights of ITU-T Study Group 12 meeting in Jan 2017 : Performance, Quality of Service and Quality of Experience (20)
Do we need a wakeup call to keep driver-less cars protected? ITU
Do we need a wakeup call to keep driver-less cars protected? This presentation was given at a Symposium on the Future Networked Car 2018 (FNC-2018) in Geneva, Switzerland on 8 March 2018. Find more information on this symposium here: https://www.itu.int/en/fnc/2018/Pages/programme.aspx
Global Virtual Mobile Network for Car manufacturersITU
This presentation discussed Global Virtual Mobile Network for Car manufacturers. The presentation was given at was given at a Symposium on the Future Networked Car 2018 (FNC-2018) in Geneva, Switzerland on 8 March 2018. Find more information on this symposium here: https://www.itu.int/en/fnc/2018/Pages/programme.aspx
Coordination of Threat Analysis in ICT EcosystemsITU
This presentation discussed Coordination of Threat Analysis in ICT Ecosystems. The presentation was given at ITU Workshop on 5G Security in Geneva, Switzerland, on 19 March 2018. Find more information about this workshop here: https://www.itu.int/en/ITU-T/Workshops-and-Seminars/20180319/Pages/programme.aspx
Learning from the past: Systematization for Attacks and Countermeasures on Mo...ITU
This presentation discussed Learning from the past: Systematization for Attacks and Countermeasures on Mobile Networks. The presentation was given at ITU Workshop on 5G Security in Geneva, Switzerland, on 19 March 2018. Find more information about this workshop here: https://www.itu.int/en/ITU-T/Workshops-and-Seminars/20180319/Pages/programme.aspx
Trustworthy networking and technical considerations for 5GITU
This presentation discussed Trustworthy networking and technical considerations for 5G. The presentation was given at ITU Workshop on 5G Security in Geneva, Switzerland, on 19 March 2018. Find more information about this workshop here: https://www.itu.int/en/ITU-T/Workshops-and-Seminars/20180319/Pages/programme.aspx
The role of Bicycles and E-Bikes in the future development of Intelligent Tra...ITU
This presentation discussed the role of Bicycles and E-Bikes in the future development of Intelligent Transport Systems. It was given at was given at a Symposium on the Future Networked Car 2018 (FNC-2018) in Geneva, Switzerland on 8 March 2018. Find more information on this symposium here: https://www.itu.int/en/fnc/2018/Pages/programme.aspx
This presentation discusses connected cars & 5G and was given at a Symposium on the Future Networked Car 2018 (FNC-2018) in Geneva, Switzerland on 8 March 2018. Find more information on this symposium here: https://www.itu.int/en/fnc/2018/Pages/programme.aspx
This presentation discusses 5G for Connected and Automated Driving and was given at a Symposium on the Future Networked Car 2018 (FNC-2018) in Geneva, Switzerland on 8 March 2018. Find more information on this symposium here: https://www.itu.int/en/fnc/2018/Pages/programme.aspx
The document discusses the need for increased security in automotive Ethernet architectures as connectivity and the use of Ethernet expands in vehicles. It summarizes Marvell's secure automotive Ethernet switch system-on-chip, which provides trusted boot, update, and runtime access concepts to securely configure, update, and access the device. It also details intrusion prevention and detection solutions to monitor data flows, detect anomalies, and execute countermeasures.
The Connected Vehicle - Challenges and Opportunities. ITU
This presentation discusses challenges and opportunities of the connected vehicle. The presentation was given at a Symposium on the Future Networked Car 2018 (FNC-2018)
held in Geneva, Switzerland on 8 March 2018. More information on the symposium can be found here: https://www.itu.int/en/fnc/2018/Pages/default.aspx
Machine learning for decentralized and flying radio devicesITU
This presentation discusses matters of machine learning for decentralized and flying radio devices. This presentation was given during the ITU-T workshop on Machine Learning for 5G and beyond, held at ITU HQ in Geneva, Switzerland on 29 Jan 18. More information on the workshop can be found here: https://www.itu.int/en/ITU-T/Workshops-and-Seminars/20180129/Pages/default.aspx
Join our upcoming forums and workshops here: https://www.itu.int/en/ITU-T/Workshops-and-Seminars/Pages/default.aspx
https://www.slideshare.net/ITU/ai-and-machine-learning
AWS offers a suite of AI and machine learning services including:
- Rekognition for image and video analysis including object detection, facial recognition and analysis, and image moderation.
- Polly for text-to-speech conversion with many voices and languages.
- Lex for building conversational bots using voice and text across channels like Alexa, Slack, and Facebook Messenger.
- Comprehend for natural language processing including keyword extraction, sentiment analysis, and topic modeling from text.
- SageMaker as a fully managed platform for building, training, and deploying machine learning models at scale.
This document discusses machine learning techniques for reconstructing radio maps in wireless networks. It addresses challenges like high mobility, noisy channels, and stringent 5G requirements. It proposes using adaptive learning to reconstruct pathloss, traffic, and load maps online from user measurements. Key ingredients discussed are sparse multi-kernel approaches for pathloss, Gaussian processes for traffic, and hybrid-driven methods for load estimation. The techniques can provide probabilistic bounds and optimize network configuration for energy efficiency.
This presentation consist of models and explanations of deep learning, artificial intelligence and today's systems and communications. This was presented at the ITU-T Workshop on Machine Learning for 5G held at the ITU HQ in Geneva, Switzerland on 29 January 2018. More information on this workshop can be found here: https://www.itu.int/en/ITU-T/Workshops-and-Seminars/20180129/Pages/default.aspx
Driven by the rapid progress in Artificial Intelligence (AI) research, intelligent machines are gaining the ability to learn, improve and make calculated decisions in ways that will enable them to perform tasks previously thought to rely solely on human experience, creativity, and ingenuity. As a result, we will in the near future see large parts of our lives influenced by AI.
AI innovation will also be central to the achievement of the United Nations' Sustainable Development Goals (SDGs) and will help solving humanity's grand challenges by capitalizing on the unprecedented quantities of data now being generated on sentiment behavior, human health, commerce, communications, migration and more.
With large parts of our lives being influenced by AI, it is critical that government, industry, academia and civil society work together to evaluate the opportunities presented by AI, ensuring that AI benefits all of humanity. Responding to this critical issue, ITU and the XPRIZE Foundation organized AI for Good Global Summit in Geneva, 7-9 June, 2017 in partnership with a number of UN sister agencies. The Summit aimed to accelerate and advance the development and democratization of AI solutions that can address specific global challenges related to poverty, hunger, health, education, the environment, and others.
The Summit provided a neutral platform for government officials, UN agencies, NGO's, industry leaders, and AI experts to discuss the ethical, technical, societal and policy issues related to AI, offer reccommendations and guidance, and promote international dialogue and cooperation in support of AI innovation.
Please visit the AI for Good Global Summit page for more resources: https://www.itu.int/en/ITU-T/AI/Pages/201706-default.aspx
If you would like to speak, partner or sponsor the 2018 edition of the summit, please contact: ai@itu.int
Standardization of XDSL and MGfast in ITU-T SG15ITU
The document summarizes work being done in ITU-T SG15 Q4 on standardizing xDSL and MGfast access technologies. It provides an overview of recent and ongoing enhancements to VDSL2 and G.fast specifications, as well as emerging work on the new MGfast technology. It discusses topics like spectral compatibility of VDSL2 and G.fast, operation over coax, dynamic time assignment, software downloads, and future work areas like impulse noise monitoring and coordinated dynamic time assignment.
Join ITU today and apply for an International Mobile Subscriber Identity (IMSI) ranges signified by the shared Mobile Country Code ‘901’, which has no ties to any single country. ‘Global SIMs’ are important for enabling cross-border global M2M & IoT connectivity, helping manufacturers to build once and sell anywhere.
For more information contact: membership@itu.int
Report on the progress made by least developed countries towards universal + affordable Internet with recommendations to achieve Sustainable Development Goal 9C https://www.itu.int/en/ITU-D/LDCs/Pages/ICTs-for-SDGs-in-LDCs-Report.aspx
Collection Methodology for Key Performance Indicators for Smart Sustainable C...ITU
These indicators have been developed to provide cities with a consistent and standardised method to collect
data and measure performance and progress to:
achieving the Sustainable Development Goals (SDGs)
becoming a smarter city
becoming a more sustainable city
The indicators will enable cities to measure their progress over time, compare their performance to other
cities and through analysis and sharing allow for the dissemination of best practices and set standards for
progress in meeting the Sustainable Development Goals (SDGs) at the city level.
For more information visit: https://www.itu.int/en/ITU-T/ssc/united/Pages/default.aspx
Enhancing innovation and participation in smart sustainable citiesITU
The Dubai Government launched an electronic shared services initiative to increase efficiency and collaboration across government entities. Over 60 shared services have been implemented, including enterprise resource planning, e-services, and infrastructure services. Implementation involved defining requirements, design, rollout, and ongoing management. Usage has grown significantly, with the human resources and financial management services now used by over 90% and 95% of Dubai government employees and budgets, respectively. The shared services have achieved over $1.2 billion in cost savings since 2003 and received high user satisfaction ratings.
Communications Mining Series - Zero to Hero - Session 1DianaGray10
This session provides introduction to UiPath Communication Mining, importance and platform overview. You will acquire a good understand of the phases in Communication Mining as we go over the platform with you. Topics covered:
• Communication Mining Overview
• Why is it important?
• How can it help today’s business and the benefits
• Phases in Communication Mining
• Demo on Platform overview
• Q/A
Dr. Sean Tan, Head of Data Science, Changi Airport Group
Discover how Changi Airport Group (CAG) leverages graph technologies and generative AI to revolutionize their search capabilities. This session delves into the unique search needs of CAG’s diverse passengers and customers, showcasing how graph data structures enhance the accuracy and relevance of AI-generated search results, mitigating the risk of “hallucinations” and improving the overall customer journey.
UiPath Test Automation using UiPath Test Suite series, part 6DianaGray10
Welcome to UiPath Test Automation using UiPath Test Suite series part 6. In this session, we will cover Test Automation with generative AI and Open AI.
UiPath Test Automation with generative AI and Open AI webinar offers an in-depth exploration of leveraging cutting-edge technologies for test automation within the UiPath platform. Attendees will delve into the integration of generative AI, a test automation solution, with Open AI advanced natural language processing capabilities.
Throughout the session, participants will discover how this synergy empowers testers to automate repetitive tasks, enhance testing accuracy, and expedite the software testing life cycle. Topics covered include the seamless integration process, practical use cases, and the benefits of harnessing AI-driven automation for UiPath testing initiatives. By attending this webinar, testers, and automation professionals can gain valuable insights into harnessing the power of AI to optimize their test automation workflows within the UiPath ecosystem, ultimately driving efficiency and quality in software development processes.
What will you get from this session?
1. Insights into integrating generative AI.
2. Understanding how this integration enhances test automation within the UiPath platform
3. Practical demonstrations
4. Exploration of real-world use cases illustrating the benefits of AI-driven test automation for UiPath
Topics covered:
What is generative AI
Test Automation with generative AI and Open AI.
UiPath integration with generative AI
Speaker:
Deepak Rai, Automation Practice Lead, Boundaryless Group and UiPath MVP
Pushing the limits of ePRTC: 100ns holdover for 100 daysAdtran
At WSTS 2024, Alon Stern explored the topic of parametric holdover and explained how recent research findings can be implemented in real-world PNT networks to achieve 100 nanoseconds of accuracy for up to 100 days.
Sudheer Mechineni, Head of Application Frameworks, Standard Chartered Bank
Discover how Standard Chartered Bank harnessed the power of Neo4j to transform complex data access challenges into a dynamic, scalable graph database solution. This keynote will cover their journey from initial adoption to deploying a fully automated, enterprise-grade causal cluster, highlighting key strategies for modelling organisational changes and ensuring robust disaster recovery. Learn how these innovations have not only enhanced Standard Chartered Bank’s data infrastructure but also positioned them as pioneers in the banking sector’s adoption of graph technology.
A tale of scale & speed: How the US Navy is enabling software delivery from l...sonjaschweigert1
Rapid and secure feature delivery is a goal across every application team and every branch of the DoD. The Navy’s DevSecOps platform, Party Barge, has achieved:
- Reduction in onboarding time from 5 weeks to 1 day
- Improved developer experience and productivity through actionable findings and reduction of false positives
- Maintenance of superior security standards and inherent policy enforcement with Authorization to Operate (ATO)
Development teams can ship efficiently and ensure applications are cyber ready for Navy Authorizing Officials (AOs). In this webinar, Sigma Defense and Anchore will give attendees a look behind the scenes and demo secure pipeline automation and security artifacts that speed up application ATO and time to production.
We will cover:
- How to remove silos in DevSecOps
- How to build efficient development pipeline roles and component templates
- How to deliver security artifacts that matter for ATO’s (SBOMs, vulnerability reports, and policy evidence)
- How to streamline operations with automated policy checks on container images
Full-RAG: A modern architecture for hyper-personalizationZilliz
Mike Del Balso, CEO & Co-Founder at Tecton, presents "Full RAG," a novel approach to AI recommendation systems, aiming to push beyond the limitations of traditional models through a deep integration of contextual insights and real-time data, leveraging the Retrieval-Augmented Generation architecture. This talk will outline Full RAG's potential to significantly enhance personalization, address engineering challenges such as data management and model training, and introduce data enrichment with reranking as a key solution. Attendees will gain crucial insights into the importance of hyperpersonalization in AI, the capabilities of Full RAG for advanced personalization, and strategies for managing complex data integrations for deploying cutting-edge AI solutions.
GraphSummit Singapore | The Future of Agility: Supercharging Digital Transfor...Neo4j
Leonard Jayamohan, Partner & Generative AI Lead, Deloitte
This keynote will reveal how Deloitte leverages Neo4j’s graph power for groundbreaking digital twin solutions, achieving a staggering 100x performance boost. Discover the essential role knowledge graphs play in successful generative AI implementations. Plus, get an exclusive look at an innovative Neo4j + Generative AI solution Deloitte is developing in-house.
Why You Should Replace Windows 11 with Nitrux Linux 3.5.0 for enhanced perfor...SOFTTECHHUB
The choice of an operating system plays a pivotal role in shaping our computing experience. For decades, Microsoft's Windows has dominated the market, offering a familiar and widely adopted platform for personal and professional use. However, as technological advancements continue to push the boundaries of innovation, alternative operating systems have emerged, challenging the status quo and offering users a fresh perspective on computing.
One such alternative that has garnered significant attention and acclaim is Nitrux Linux 3.5.0, a sleek, powerful, and user-friendly Linux distribution that promises to redefine the way we interact with our devices. With its focus on performance, security, and customization, Nitrux Linux presents a compelling case for those seeking to break free from the constraints of proprietary software and embrace the freedom and flexibility of open-source computing.
Securing your Kubernetes cluster_ a step-by-step guide to success !KatiaHIMEUR1
Today, after several years of existence, an extremely active community and an ultra-dynamic ecosystem, Kubernetes has established itself as the de facto standard in container orchestration. Thanks to a wide range of managed services, it has never been so easy to set up a ready-to-use Kubernetes cluster.
However, this ease of use means that the subject of security in Kubernetes is often left for later, or even neglected. This exposes companies to significant risks.
In this talk, I'll show you step-by-step how to secure your Kubernetes cluster for greater peace of mind and reliability.
Building RAG with self-deployed Milvus vector database and Snowpark Container...Zilliz
This talk will give hands-on advice on building RAG applications with an open-source Milvus database deployed as a docker container. We will also introduce the integration of Milvus with Snowpark Container Services.
Let's Integrate MuleSoft RPA, COMPOSER, APM with AWS IDP along with Slackshyamraj55
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Presented by Vladimir Iglovikov:
- https://www.linkedin.com/in/iglovikov/
- https://x.com/viglovikov
- https://www.instagram.com/ternaus/
This presentation delves into the journey of Albumentations.ai, a highly successful open-source library for data augmentation.
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People: The contributors and community that have supported Albumentations.
Metrics: The success indicators such as downloads, daily active users, GitHub stars, and financial contributions.
Challenges: The hurdles in monetizing open-source projects and measuring user engagement.
Development Practices: Best practices for creating, maintaining, and scaling open-source libraries, including code hygiene, CI/CD, and fast iteration.
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Marketing: Both online and offline marketing tactics, focusing on real, impactful interactions and collaborations.
Mental Health: Maintaining balance and not feeling pressured by user demands.
Key insights include the importance of automation, making the adoption process seamless, and leveraging offline interactions for marketing. The presentation also emphasizes the need for continuous small improvements and building a friendly, inclusive community that contributes to the project's growth.
Vladimir Iglovikov brings his extensive experience as a Kaggle Grandmaster, ex-Staff ML Engineer at Lyft, sharing valuable lessons and practical advice for anyone looking to enhance the adoption of their open-source projects.
Explore more about Albumentations and join the community at:
GitHub: https://github.com/albumentations-team/albumentations
Website: https://albumentations.ai/
LinkedIn: https://www.linkedin.com/company/100504475
Twitter: https://x.com/albumentations
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#MongoDB #VectorSearch #AI #SemanticSearch #TechInnovation #DataScience #LLM #MachineLearning #SearchTechnology
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Highlights of ITU-T Study Group 12 meeting in Jan 2017 : Performance, Quality of Service and Quality of Experience
1. ITU-T Study Group 12:
Performance, Quality of Service and
Quality of Experience
Highlights of January 2017 meeting
Kwame BAAH-ACHEAMFUOR
Study Group 12 Chairman
Geneva, 10-19 January 2017
3. WP1/12: Terminals and multimedia
subjective assessment
Highlights of January 2017 meeting
Lars Birger Nielsen
WP1/12 Chairman
Gunilla Berndtsson
WP1/12 Vice-Chairman
4. P.381 “Technical requirements and test methods
for the universal wired headset or headphone
interface of digital mobile terminals”
Produced by Q.3 (Speech transmission characteristics
of communication terminals for fixed circuit-switched,
mobile and packet-switched (IP) networks)
Requirements for electroacoustic test and
corresponding test methods – incl. EN 50332-1 and 2
Specification of pin assignments of a socket connector
with three/four contact points
For characterisation of headset/headphones as well
as the interface of digital wireless terminals
P.381: one microphone
Revision for support of super-wideband and fullband
4
Revised Recommendations Consented
5. P.11x0 “In vehicle Recommendations”
Produced by Q.4 (Hands-free communication in
vehicles)
Revised Recommendation P.1100 (NB), P.1110
(WB) and P.1140 (eCall)
Measurement of Delay was aligned with 3GPP specifications
Adaptation of a new background noise equalization method
Alignment of descriptions and editorial updates
New Recommendation P.1120 (SWB + FB)
5
Recommendations Consented
6. P.501 “Test signals for use in telephonometry”
Produced by Q.6 (Analysis methods using complex
measurement signals including their application for
speech enhancement techniques and hands-free
telephony)
Texts of amendments have been moved into the
P.501 document
All speech samples and other samples will become
available for download once the approval process of
P.501 has been concluded.
6
Revised Recommendations Consented
7. New Recommendations Consented
P.1310 “(Spatial audio meetings quality
evaluation”
Produced by Q.10 (Conferencing and
telemeeting assessment)
Provide guidance on appropriate subjective
testing methods for evaluating of spatial audio
telemeeting systems – w/wo video
Methods for spatial audio on how to
Collect subjective quality ratings
Assess accuracy of the spatial alignment of
audiovisual systems
Obtain task performance indicators for the
intelligibility of concurrent speakers
Measure communication effectiveness using task
performance
Obtain task performance indicators for the cognitive
load experienced
7
Method Annex A Annex B Annex C Annex D Annex E
Rendering
Scenario
Audio-only
or audio component of
an audiovisual system
Audiovisual
Test Focus Overall System
Individual Components
Target
Variables
Overall Quality (3
(3
Media
Quality
Non-spatial
quality
Spatial
quality
Communication quality (2
(2
(2
Measurement
Sensitivity
Sufficient (test added
value of spatial audio)
High (differentiate
variants of spatial
audio)
1)
Test
Paradigm
Listening-Only (Non-
interactive) Test
Conversation Test
Measurement
Paradigm
Subjective Ratings
Performance Measures
Notes:
1)
Depending on the actual implementation of the method details, e.g. training phase, it is possible to
achieve high measurement sensitivity.
2)
These methods (indirectly) measure individual aspects of communication quality.
3)
These methods are not measuring overall quality, but they may be combined with subjective ratings
of overall quality.
8. New work items
Two new work items has been started
New work item on “Transmission characteristics for In
Car Communication” or P.ICC
New work item on “Subjective methodology for
Speech and Audio multi-channel Test” or P.SAT
8
9. Rapporteurs & SG12 Meetings
9
Q.4/12 “Objective methods for speech and audio
evaluation in vehicles”
Date: August 2nd
Place: Geneva, Switzerland
Topic: To progress on P.ICC
Q.5/12 “Telephonometric methodologies for handset
and headset terminals”
Date: May 29th and 30th
Place: Bern, Switzerland
Topic: To progress on P.TBN and P.Loudness
SG12 meeting
Date: September 19-28 (2017)
Place: Geneva, Switzerland
10. WP1/12: Terminals and multimedia
subjective assessment
Highlights of January 2017 meeting
Lars Birger Nielsen
WP1/12 Chairman
Gunilla Berndtsson
WP1/12 Vice-Chairman
END of WP1 PRESENTATION
11. WP2/12: Objective models and tools
for multimedia quality
Highlights of January 2017 meeting
Vincent Barriac
WP2/12 Chairman
Ludovic Malfait
WP2/12 Vice-Chairman
12. List of Questions under study
Q9/12
Perceptual-based objective methods for voice, audio and
visual quality measurements in telecommunication services
Q14/12
Development of parametric models and tools for multimedia
quality assessment of packet-based video services
Q15/12
Parametric and E-model-based planning, prediction and
monitoring of conversational speech quality
Q16/12 Framework for diagnostic functions
Q19/12 Objective and subjective methods for evaluating perceptual
audiovisual quality in multimedia services
12
13. Q.9: summary of discussion (1/2)
13
P.863
A modified version of P.863 that minimizes the under-
prediction in case of extended micropauses was introduced.
This version is available for testing to all interested parties in
ITU-T and commercial POLQA customers.
The results of a P.800 ACR subjective test focusing on the
evaluation of EVS and Opus coding was introduced and
discussed. Contributions providing additional experimental
results would be greatly appreciated to verify the performance
of P.863 and other objective measures on Opus and other
proprietary codecs.
14. Q.9: summary of discussion (2/2)
14
P.AMD and P.SAMD
Next SG12 meeting is targeted as the submission date for
candidate models for both sets A (5 dimensions) and B (7
dimensions).
For the new work item P.SAMD the existing Requirement
Specification is extended to cover both P.AMD and P.SAMD in
a single document.
P.SPELQ
The latest results are looking promising for narrowband
databases. Up to now, there are no published training results
for super-wideband databases.
A stable version of the P.SPELQ model is expected to be
submitted to TSB before the next SG12 meeting so the
evaluation can start
15. Q.9: outputs
15
No liaison statement
One document agreed:
Supplement for P series addressing P.863 and P.863.1:
“Application of P.863 and P.863.1 for speech processed by
blind bandwidth extension approaches”, available as TD163
rev 1.
One interim meeting is requested
Proposed time frame is June / Beginning of July
Main topics for discussion: P.AMD/P.SAMD and probably
P.SPELQ.
Exact date and location will be discussed and agreed via the
Q9/12 reflector. Discussion topics will be updated according
to the submitted contributions.
16. Q.14: summary of discussion (1/2)
16
P.1203: Parametric bitstream-based quality
assessment of progressive download and adaptive
audiovisual streaming services over reliable transport
amendment with the final performance figures for all four
modes, concluding the P.NATS phase 1 work
New work item for a new recommendation being the
extension of the P.1201 Recommendations
This new work item focuses on UDP-based video for new
codecs and higher video resolutions.
17. Q.14: summary of discussion (2/2)
17
New study item AVHD-AS/P.NATS phase 2 (joint
project with VQEG)
extension of the scope of P.1203 with new video codecs and
resolutions (including UHD), inclusion of a larger variety of
model types (new: pixel-based)
11 proponents have responded positively to the provisional
call for participation.
44 subjective test databases to be created
device types and viewing distance discussed and agreed
distribution of short vs. long sequence databases set to 70/30, training
vs validation databases to 50/50 for short sequence databases and
25/75 for long sequence databases.
18. Q.14: outputs
18
One document agreed
Amendment to P.1203 (available as TD139)
No liaison statements
Two Rapporteur Meetings are requested
Q.13, 14 and 17/12 (“Q.44”) plus VQEG AVHD (to discuss
AVHD / P.NATS Phase 2): End of March with the most likely
dates 22-24 March 2017, hosted by Deutsche Telekom / T-
Labs
Q.14/12 / AVHD-AS meeting on AVHD/P.NATS Phase 2 at
VQEG meeting to be held 8-12 May 2017 in Los Gatos,
California, USA, hosted by Netflix
19. Q.15: summary of discussion
19
Wideband extension to G.107.1
The decision has been taken to work on a basic SWB
version of the E-model. Will be re-used by Q.13 for
G.1070
P.CQO
The data received on the validation of the underlying
subjective procedure (P.CQS) was considered very useful,
but should be augmented by results from other test labs,
involving different test set-ups, participant groups, and
potentially languages to substantiate the subjective
method.
20. Q.16: summary of discussion
20
P.TCA: Technical Cause Analysis
It was proposed to draft a Recommendation that
provides P.TCA degradation classes, their description
and exemplary audio samples. This Recommendation
would provide the industry with a reference for common
terminology.
E.FINAD: Framework for Intelligent Network
Analytics and Diagnostics
A new work item is proposed to recommend a
framework for identifying anomalies and diagnosing root
cause from data collected from production networks
such as logs, statistics, alerts, etc…
21. Q.19: summary of discussion
21
J op-tr (Methods for Optimizing Bitrates and
Transmission Resolution by Considering Display
Characteristics and Available Bandwidth)
New progress based on experimental results and statistical
analyses provided in C 67
showed that equivalent perceptual video quality could be provided with
reduced bandwidth by considering the display characteristics (size and
resolutions).
J.noref
Potential interaction with Q.14 (P.NATS phase 2) in the near
future
test data of P.NATS-Phase 2 could be re-used for the validation of J.noref
possibility to any model originating from Q.19 to be submitted as
candidate models to P.NATS phase 2 competition and work item
distinction between scopes: P.NATS phase 2 work item is focused on TCP
based video, while J.noref work item originates from broadcast type of
video and includes UDP based video and packet loss.
22. IRG AVQA
22
Decision to continue
Co-chairman on behalf of SG12 unchanged: Jens
Berger (Q.9 rapporteur)
One liaison statement sent to SG9 (available as TD
160)
23. WP2/12: Objective models and tools
for multimedia quality
Highlights of January 2017 meeting
Vincent Barriac
WP2/12 Chairman
Ludovic Malfait
WP2/12 Vice-Chairman
END OF WP2 PRESENTATION
24. WP3/12: Multimedia QoS and QoE
Highlights of January 2017 meeting
Paul Coverdale
WP3/12 Chairman
Al Morton
Tiago Sousa Prado
WP3/12 Vice-Chairmen
25. New Recommendations Consented
E.847 (G.PoiCong) “QoS norms for
TDM Interconnection between
Telecom Networks”
This Recommendation analyses and
identifies QoS parameters for TDM
interconnection between telecom
networks, which are needed to facilitate
effective interconnections with
reasonable traffic handling capacities.
25
26. New Recommendations Consented
Y.1545.1 (Y.FMIPQoS) “Framework
for Monitoring the QoS of IP network
services”
This Recommendation is a diagnostic
reference for IP network quality of
service monitoring, and primarily as a
guide to assist regulators monitor the
quality of service of internet that is
provided by service providers (although
subscribers and network service
providers may also derive benefit). 26
27. New Recommendations Consented
E.811 (E.QMME) “Quality
Measurement in Major Events”
This Recommendation addresses the
quality assessment of mobile broadband
and voice services provided during
major events, such as FIFA World Cups
and Olympic and Paralympic Games, by
creating a useful international reference
to be considered by operators and
regulators when preparing to host such
events. 27
28. Amended Recommendations
Consented
E.802 “New Annex A on guidelines
on selection of representative
samples”
provides guidance on the selection of
representative samples in the
measurement of QoS parameters
28
29. Liaisons
To SG11 and TSAG regarding the
work item Y.FMIPQoS (now
Y.1545.1)
We have consented Y.1545.1
The WTSA-16 guidance indicates that
SG 12 has clear responsibility for
various aspects of QoS, including
measurement methodologies
29
30. Liaisons
To 3GPP on KPIs for mobile services
requests information and appropriate
references related with Voice and Data
QoS KPI thresholds on mobile networks.
30
31. Liaisons
To SG16, VQEG, MPEG, 3GPP SA4
Informs about a new study item on
quality of experience (QoE) for virtual
reality (VR) services (G.QoE-VR)
31
32. Liaisons
To SG2 on study of website traffic
characterization
Informs that we have removed G.MFWT
from our work items
32
33. Liaisons
To SG5 on proposed collaboration on
accuracy and availability
requirements regarding soft error
requirements
Recommends to recognize the evolution
to VNF-based telecom architectures in
K.soft_des
33
34. New work items
G.ViLTE
End-to-end QoS for Video Telephony
over 4G mobile networks
E.RQUAL
Strategies to Establish Quality
Measurement Frameworks
E.NetPerfRank
Statistical Framework for QoE Centric
Benchmarking Scoring and Ranking
34
35. New work items
E.RQST
Voice and Data QoS KPI thresholds on
mobile networks
G.QoE-VR
QoE for Virtual Reality
35
36. Interim meetings
To be co-located with Q14/12
Question 13/12
One meeting requested
Tentatively March 2017 in Berlin
Question 17/12
One meeting requested
Tentatively March 2017 in Berlin
36
37. WP3/12: Multimedia QoS and QoE
Highlights of January 2017 meeting
Paul Coverdale
WP3/12 Chairman
Al Morton
Tiago Sousa Prado
WP3/12 Vice-Chairmen
END OF WP3 PRESENTATION
38. Next Study Group 12 meeting
19-28 September 2017, Geneva,
Switzerland
SG12 Website: www.itu.int/go/tsg12
38