Showing papers on "Telephony published in 2021"

Evaluating Congestion Control for Interactive Real-time Media

Abstract: The Real-time Transport Protocol (RTP) is used to transmit media in telephony and video conferencing applications. This document describes the guidelines to evaluate new congestion control algorithms for interactive point-to-point real-time media.

Detection of Telephony Spam and Scams using Recurrent Neural Network (RNN) Algorithm

Abstract: Malicious phone calls, including spam and scams, caused millions of global financial losses every year and was a difficult problem over many years. This work introduces the solution based on machine learning for telecommunications without underlying the telephone network infrastructure. The major obstacle of this ten years old problem is building efficient functions without access to the telephony network infrastructures. The previous Spam Call data set is collected first. The dataset includes several labelbased features to predict malicious calling. We primarily focus on using Recurrent Neural Network (RNN) algorithm to detect the malicious calls. With the proposed features, we review different state of the art methods of machine-learning, and it is inferred that the most optimal approach can minimize malicious calls to 90% while keeping over 90% of the binary call accuracy. The outcomes also show that without significant overhead latency the models can be implemented effectively with the help of an evaluation analysis.

CallChain: Identity Authentication Based on Blockchain for Telephony Networks

Abstract: Telephony networks serve as a reliable communication channel, and phone numbers are usually used to authenticate users' identities in telephony networks. However, modern telephony networks does not provide end-to-end authentication for phone numbers, making it difficult for a receiver to confirm a caller's real identity. Moreover, due to the lack of effective authentication solutions for users, phone numbers are usually exploited to commit identity theft, credit card fraud, or telecom harassment by phone number spoofing a ttacks. I n t his paper, we propose a blockchain-based identity authentication system for telephony networks, CallChain. CallChain issues verifiable digital identities to users which bound a user to his/her phone number. Based on the digital identity, CallChain provides end-to-end identity authentication between the caller and receiver in the data channel without modifying the core technologies in telephony networks. In the proposed system, users first register decentralized identities (DIDs) and store them in the blockchain distributed ledger. Then Call Center issues verifiable PhoneNumber Credentials to users. And finally users conduct end-to-end identity authentication based on these PhoneNumber Credentials before the call is established. We implement a prototype as an App for Android-based phones based on an open source blockchain platform Hyperledger Indy. Moreover, we detect that CallChain only adds a worst-case 2.1 seconds call establishment overhead and can resist DDoS attacks effectively.

Rural Technology Adoption and Use Model in Rural Africa: A Predictive Approach to Telephony Acceptance

Abstract: Mobile telephony networks have seen a high rate of adoption worldwide in recent years. However, these networks do not exist everywhere, and even where they are, their adoption is lagging. Especially in uncovered rural areas, it is difficult to predict the technology's acceptance and adoption factors. This study deals with the usage gap of mobile telephone networks and attempts in a methodological approach based on structural equation modeling to prevent the telephone usage gap in rural Africa yet to be covered. To that purpose, the authors use a research model based on the technology acceptance model (TAM) and the unified theory of acceptance and use of technology (UTAUT). By combining these two models and incorporating the moderating effects of demographic variables such as age, gender, education, and experience of technology use, this paper has retained a model with the ability to determine how rural residents will accept and use future networks.

Implementasi Trunk Interkoneksi Multi Server menggunakan Singleboard Komputer

Abstract: Voice over Internet Protocol (also called VoIP, IP Telephony, Internet Telephony or Digital Phone)is a technology that enables remote voice conversation through internet media. Voice data is converted into digital code and streamed over a network that delivers data packets, rather than through a regular analog telephone circuit. In VoIP communication,users make telephone connection through a terminal in the form of a PC or a regularphone. By calling voip, manyadvantages that can be taken in terms of costare obviously cheaper than traditional phone rates, because IP networks are global. IP Phone can be added, moved and changed. This is because VoIP can be installed in any ethernet and IP address,unlike conventional phones that must have their own port in Sentral or PBX (Private branch exchange). In this study implementing connectedness between servers using single board computer installed elastix operating system that aims to implement prefix for between servers and use some audio codecs. The results of telephone research using prefix and without prefix as many as 6 clients or 3 pasng calls simultaneously the highest packet loss value in codec speex with prefix of 2.34%. The highest bandwidth value used is with pcmu prefix codec with an average of 82.3 Kbps and without prefix 79.3 Kbps. Keyword : Server, VoIP, IP Telphony, Internet telephony, Digital Phone, IP Address, PBX, Codec, Prefix.

VoIP Security: A Performance and Cost-benefit Analysis

Abstract: Voice over IP (VoIP) has become the standard technology for telephony and has replaced the old Public Switched Telephone Network (PSTN). This research focuses on the security aspect of VoIP systems. Unsecured VoIP systems are vulnerable to malicious attacks. However, the overhead of the security techniques hampers the performance of VoIP systems. This research analyzes how a VoIP system performs with different security techniques. The performance of the VoIP system is analyzed on different types of data networks such as IPv4, IPv6, and IPv4/IPv6 mixed networks, and in scenarios such as with and without network traffic. Additionally, the research includes a cost-benefit analysis of the security techniques, to determine their cost effectiveness. Based on the performance analysis and cost-benefit analysis, this research proposes three security techniques that can be applied to VoIP systems deployed on IPv4, IPv6, and IPv4/IPv6 networks.

Compromised Computers Meet Voice Assistants: Stealthily Exfiltrating Data as Voice over Telephony

Abstract: New security concerns arise due to the growing popularity of voice assistants (VA) in home and enterprise networks. We explore how malware infected computers can encode sensitive data into audio and leverage nearby VAs to exfiltrate it. Such low cost attacks can be launched remotely, at scale, and can bypass network defenses. By using Dual-Tone Multi-Frequency tones to encode data into audio that is played over ordinary computer speakers, modest amounts of data (e.g., a kilobyte) can be transmitted with a phone call lasting a few minutes. This can be done while making the audio nearly inaudible for most people. With the help of a prototype built by us, we experimentally assess the impact of several factors that impact data transfer rates and transmission accuracy achieved by such attacks. Our results show that voice assistants in the vicinity of computers can pose new threats to data stored on them.

Model for Predicting Customer Desertion of Telephony Service using Machine Learning

Abstract: In the present study, it is observed that many people are affected by the services provided by telephony, who leave the service for different reasons, for which the use of a model based on decision trees is proposed, which allows predicting potential dropouts from Customers of a telecommunications company for telephone service. To verify the results, several algorithms were used such as neural networks, support vector machine and decision trees, for the design of the predictive models the KNIME software was used, and the quality was evaluated as the percentage of correct answers in the predicted variable. The results of the model will allow acting proactively in the retention of clients and improves the services provided. A data set with 21 predictor variables that influence customer churn was used. A dependent variable (churn) was used, which is an identifier that determines if the customer left = 1, did not leave = 0 the company's service. The results with a test data set reach a precision of 91.7%, which indicates that decision trees turn out to be an attractive alternative to develop prediction models of customer attrition in this type of data, due to the simplicity of interpretation of the results.

Multicarrier Modulation and OFDM

Abstract: The division of a frequency band into dedicated channels by modulation, called frequency-division multiplexing (FDM), has a long history. Prominent examples are the analog telephone networks until the 1970s. They were replaced by digital telephony, operating with time-division multiplex (TDM), in which dedicated time slot are allocated to each application. Moreover, the analog and later the digital television broadcasting networks—terrestrial, satellite and cable based—operate with FDM. Most mobile and cellular networks today utilize combinations of FDM and TDM.

An Attack Vector Taxonomy for Mobile Telephony Security Vulnerabilities

Abstract: A simplified cybersecurity threat matrix may provide a unifying way to define the security risk posed by current and future generations of mobile telephony.

System and method for verifying telephone numbers

Abstract: Programmatically reversing numerical line identity presented at a communications services gateway into named IP Telephony users with “prior association”, delivers dynamic “reverse address resolution” switching connections from ground to cloud, permitting any conventional telephone to dial and connect to any associated IP Telephony endpoint in the world, without changes to the conventional telephone. Reversing line identity into associated named users bridges both the addressability and economic divide between mass conventional “paying” (mobile and fixed) and “free”. IP Telephony networks. A system for supporting communications between a user on an IP-addressed-communications-device and a telephony subscriber device, the telephony subscriber device having a corresponding telephone number, includes: one or more service nodes configured to: receive from the user the telephone number of the telephony subscriber device and create an association from the telephone number to the user, wherein the association allows the telephony subscriber device to connect to the user.

On premises gateways interconnecting VoIP systems the public switched telephone network and private branch exchanges and other telephony infrastructure

Abstract: On premises gateways located within organization sites interconnect VoIP systems, the public switched telephone network (PSTN), Private Branch Exchanges and other telephony infrastructure.

Factors Affecting the Accessibility of Voice Telephony for People with Hearing Loss: Audio Encoding, Network Impairments, Video and Environmental Noise

Abstract: This paper describes four studies with a total of 114 individuals with hearing loss and 12 hearing controls that investigate the impact of audio quality parameters on voice telecommunications. These studies were first informed by a survey of 439 individuals with hearing loss on their voice telecommunications experiences. While voice telephony was very important, with high usage of wireless mobile phones, respondents reported relatively low satisfaction with their hearing devices’ performance for telephone listening, noting that improved telephone audio quality was a significant need. The studies cover three categories of audio quality parameters: (1) narrowband (NB) versus wideband (WB) audio; (2) encoding audio at varying bit rates, from typical rates used in today's mobile networks to the highest quality supported by these audio codecs; and (3) absence of packet loss to worst-case packet loss in both mobile and VoIP networks. Additionally, NB versus WB audio was tested in auditory-only and audiovisual presentation modes and in quiet and noisy environments. With WB audio in a quiet environment, individuals with hearing loss exhibited better speech recognition, expended less perceived mental effort, and rated speech quality higher than with NB audio. WB audio provided a greater benefit when listening alone than when the visual channel also was available. The noisy environment significantly degraded performance for both presentation modes, but particularly for listening alone. Bit rate affected speech recognition for NB audio, and speech quality ratings for both NB and WB audio. Packet loss affected all of speech recognition, mental effort, and speech quality ratings. WB versus NB audio also affected hearing individuals, especially under packet loss. These results are discussed in terms of the practical steps they suggest for the implementation of telecommunications systems and related technical standards and policy considerations to improve the accessibility of voice telephony for people with hearing loss.

Unifying presence data and communication routing based thereon

Abstract: Maintaining accurate out-of-office notifications can be resource intensive and error-prone. This is compounded by duplicating efforts, such as between a telephony system and an email system. By automatically creating and presenting a personal interactive voice response (IVR) or dual-tone multi-frequency (DTMF) signaling in response to menu option, generated from an email system, a telephony system may automatically be updated to provide options for a caller to select delegates for a particular user. The delegates may further be identified by subject matter or other categorization. As a result, a user may maintain one system and, as a result, enable a second system to automatically present and process delegate selection without additional human input.

Deep Feature CycleGANs: Speaker Identity Preserving Non-parallel Microphone-Telephone Domain Adaptation for Speaker Verification

Abstract: With the increase in the availability of speech from varied domains, it is imperative to use such out-of-domain data to improve existing speech systems. Domain adaptation is a prominent pre-processing approach for this. We investigate it for adapt microphone speech to the telephone domain. Specifically, we explore CycleGAN-based unpaired translation of microphone data to improve the x-vector/speaker embedding network for Telephony Speaker Verification. We first demonstrate the efficacy of this on real challenging data and then, to improve further, we modify the CycleGAN formulation to make the adaptation task-specific. We modify CycleGAN's identity loss, cycle-consistency loss, and adversarial loss to operate in the deep feature space. Deep features of a signal are extracted from an auxiliary (speaker embedding) network and, hence, preserves speaker identity. Our 3D convolution-based Deep Feature Discriminators (DFD) show relative improvements of 5-10% in terms of equal error rate. To dive deeper, we study a challenging scenario of pooling (adapted) microphone and telephone data with data augmentations and telephone codecs. Finally, we highlight the sensitivity of CycleGAN hyper-parameters and introduce a parameter called probability of adaptation.

A method of identifying and classifying the behavior modes of a plurality of data relative to a telephony infrastructure for network function virtualization

Abstract: A method of identifying and classifying the behavior modes of a plurality of data relative to a telephony infrastructure for Network Function Virtualization comprising the steps of: providing (1) a first database containing historical telephony infrastructure monitoring data concerning the level of use of the telephony infrastructure resources; providing (2) a second database containing historical telephony infrastructure application monitoring data concerning the level of use of the telephony infrastructure application; providing (3) a control unit in signal communication with the first and second databases; providing (4) a neural network of Self-Organizing Map type; extrapolating (5), by the control unit, at least one first subset of historical data from the historical telephony infrastructure monitoring data, said first subset of data being relative to a predetermined time window; extrapolating (6), by the control unit, at least one second subset of historical data from the historical application monitoring data, said second subset of historical data being relative to the predetermined time window; defining (7), by the control unit, at least one historical input vector comprising the historical data of the first and second subsets of historical data; training (8), by the control unit, the neural network to define a weight vector comprising a plurality of weight coefficients for each neuron of the neural network such that by providing the at least one historical input vector as an input to the neural network, the behavior mode of the historical data of the at least one historical input vector is obtained as an output, each weight vector representing a respective behavior mode; acquiring (9), by the control unit, a plurality of current telephony infrastructure monitoring data and current telephony infrastructure application monitoring data, such current data being acquired in real time and being relative to a current window; defining (10), by the control unit, at least one current input vector comprising the current acquired data; providing (11), by the control unit, the at least one current input vector at the input of the trained neural network to obtain, at the output, the neuron of neural network that is most stimulated by the current input vector; analyzing (12) the weight vector associated with the neuron obtained in the previous step to define the behavior mode of the current data of the current input vector.

Systems and methods for utilizing http for telephony trunking between a provider and a consumer

Abstract: Systems and methods are described herein for providing a Voice over Internet Protocol (VoIP) call. In an embodiment, a load balancing processor receives a re-initiated HTTP request from a client processor upon detection that an initial call server is no longer active, and sends the re-initiated HTTP request to a second call server. The second server generates updated call resource information that identifies the second server as the new server resource for the call, and sends the updated call resource information over the IP network to the client processor. Subsequent HTTP requests from the client processor for sending and receiving signaling and media data for the call are received at the second server using the updated call resource information.

Detection of Calls From Smart Speaker Devices

Abstract: The ubiquity of smart speakers is increasing, with a growing number of households utilising these devices to make calls over the telephony network. As the technology is typically configured to retain the cellular phone number of the user, it presents challenges in applications where knowledge of the true call origin is required. There are a wide range of makes and models for these devices, as is the case with cell phones, and it is challenging to detect the general category as a smart speaker or cell, independent of the designated phone number. In this paper, we present an approach to differentiate between calls originating from smart speakers and ones from cellular devices using only the audio. We present a feature set that characterises the relevant acoustic information, such as the degree of reverberation and noise, to distinguish between these categories. When evaluated on a dataset spanning multiple models for each device category, as well as different modes-of-usage and microphone-speaker distances, the method yields an Equal Error Rate (EER) of 12.6%.

Adaptive Data Hiding Based Telephony Speech Enhancement

Abstract: Public telephone systems transmit speech across a limited frequency range, about 300–3400 Hz, called narrowband (NB) which results in a significant reduction of quality and intelligibility of speech. This paper proposes a fully backward compatible novel method for bandwidth extension of NB speech. The method uses adaptive data hiding technique to provide a perceptually better wideband speech signal. The spectral envelope parameters are extracted from the high frequency components of speech signal existing above NB, which are then spread by using spreading sequences, and are embedded in the NB speech signal using adaptive data hiding technique. The embedded information is extracted at the receiving end to reconstruct the wideband speech signal. Theoretical and simulation analyses show that the proposed method is robust to quantization and channel noises. The log spectral distortion test clearly show that the reconstructed wideband signal gives a much better performance in terms of speech quality when compared to the conventional speech bandwidth extension methods employing data hiding.

Blockchain network communication management

Abstract: A device that is capable of sending/receiving telephony-based messages may communicate with a blockchain network without having a direct connection (e.g., internet connection) to the blockchain network. A message may be communicated via a telephony network to a telephony carrier system. The telephony carrier system may translate the telephony-based message into a blockchain-based communication and provide the blockchain-based communication to the blockchain network. In addition, an entity on the blockchain network may communicate with a device that does not have an internet connection to the blockchain network. The entity may initiate a blockchain-based communication that is received by an on-chain interface of a telephony network carrier. In response, the telephony network carrier may generate a telephony-based message and communicate the telephony-based message to the user device.

Apparatus and method for providing video telephony service for vehicle

Abstract: A video telephony service provision apparatus for a vehicle is provided. The apparatus includes a display unit having a plurality of window displays that output an image of an occupant of a first other vehicle and a speaker unit having a plurality of speakers that output a voice of the occupant of the first other vehicle. A manipulation unit receives user input requesting a change of output of at least one of the image or the voice of the occupant of the first other vehicle and a controller operates the display unit and the speaker unit.

Performance Evaluation of DWT based Multiple Input Multiple Output OFDM for Internet Telephony Applications

Abstract: Digital communication plays a vital role in transmitting and receiving the information With the advent of advancements in telecommunications, requirement of high data rate is in evident in Digital communication for the transmission of the signal The information which has to be transmitted over a long distance needed to be error free while receiving in order to obtain the quality of the information Emphasizing on the high data rate and also error free requirements, the role of Multiple Input Multiple Output and Wavelet as a combination makes progress towards high extent in achieving successful transmission The proposed work emphasizes on the application of Discrete wavelet transform for filtering the Quadrature Amplitude modulated wave for the internet telephonic communication where the digital communication plays a major role This proposed work purely concentrates on the digital communication part of the internet telephony communications Stein’s Unbiased Risk Estimate is used to obtain the information samples with soft thresholding The Bit to Error rate performance is measured for various modulation order and the simulated results are shown and also the values of the Bit to Error rate are tabulated

End-to-End Delay Model of the Main Types Satellites for VoIP-telephony over Satellite Links

Abstract: Over the past few decades, the Internet has become the world's information superhighway, leading to the development of new applications and services. Among them, Voice over IP (VoIP) is one of the most promising industries. Voice over Internet Protocol (VoIP or IP) telephony is a new way of communication. This technology allows users to make calls over a network using Internet protocols. This paper will describe VoIP-telephony, to demonstrate the ability of people to communicate through the Internet, which works via satellite links. The research in this paper demonstrates the basic formula of the latency model when using any type of satellite network such as LEO, GEO, and MEO systems for IP-telephony. In addition, this work provides methods for reducing the latency.

A method of predicting the time course of a plurality of data relative to a telephony infrastructure for network function virtualization

Abstract: A method of predicting the time course of a plurality of data relative to a telephony infrastructure for Network Function Virtualization comprising the following steps: providing (1) a first database containing historical telephony infrastructure monitoring data concerning the level of use of the telephony infrastructure resources and/or any error conditions or anomalies occurred therein; providing (2) a second database containing historical telephony infrastructure application monitoring data concerning the level of use of the telephony infrastructure application, its performance or reliability and/or any error conditions or anomalies occurred therein; providing (3) a control unit in signal communication with the first and second databases; providing (4) an automatic learning mechanism; extrapolating (5), by the control unit, at least one first subset of historical data from the historical telephony infrastructure monitoring data and/or from the historical telephony infrastructure application monitoring data, said first subset of historical data being relative to a first predetermined time window; extrapolating (6), by the control unit, at least one second subset of historical data from the historical telephony infrastructure monitoring data and/or from the historical telephony infrastructure application monitoring data, said second subset of historical data being relative to a second predetermined time window; defining (7), by the control unit, a historical input vector comprising the historical data of the first subset of historical data; defining (8), by the control unit, a historical output vector comprising the historical data of the second subset of historical data; training (9), by the control unit, the automatic learning mechanism to obtain, at its output, the historical output vector by providing the historical input vector at the input of the automatic learning mechanism; acquiring (10), by the control unit, at least one subset of current telephony infrastructure monitoring data and/or telephony infrastructure application monitoring data, such current data being relative to a current time window; defining (11) a current input vector comprising such subset of current data; providing (12) the current input vector at the input of the trained automatic learning mechanism to obtain a prediction output vector.

Method and system for low density hosted telephony regulatory compliance

Abstract: A method for facilitating regulatory compliance with respect to telephony communications is provided. The method includes: assigning, to a mobile telephone, an access number; routing a communication that relates to the assigned access number to a gateway; processing the routed communication to ensure compliance with all jurisdictional regulations, including real-time call recording and short message service (SMS) capturing for compliance archival and search, applying data loss prevention (DLP) rules, and any other suitable processing; and forwarding the processed communication to the mobile telephone. A subscriber identification module (SIM) that is associated with the access number is physically installed at the gateway. The communication may be a voice communication and/or an SMS text communication.