Showing papers on "Telephony published in 2014"

5G: Personal mobile internet beyond what cellular did to telephony

Abstract: Cellular technology has dramatically changed our society and the way we communicate. First it impacted voice telephony, and then has been making inroads into data access, applications, and services. However, today potential capabilities of the Internet have not yet been fully exploited by cellular systems. With the advent of 5G we will have the opportunity to leapfrog beyond current Internet capabilities.

Video telephony for end-consumers: measurement study of Google+, iChat and Skype

Abstract: Video telephony requires high-bandwidth and low-delay voice and video transmissions between geographically distributed users. It is challenging to deliver high-quality video telephony to end-consumers through the best-effort Internet. In this paper, we present our measurement study on three popular video telephony systems on the Internet: Google+, iChat, and Skype. Through a series of carefully designed active and passive measurements, we uncover important information about their key design choices and performance, including application architecture, video generation and adaptation schemes, loss recovery strategies, end-to-end voice and video delays, resilience against random and bursty losses, etc. The obtained insights can be used to guide the design of applications that call for high-bandwidth and low-delay data transmissions under a wide range of "best-effort" network conditions.

"Can you SEE me now?" A Measurement Study of Mobile Video Calls

Abstract: Video telephony is increasingly being adopted by end consumers It is extremely challenging to deliver video calls over wireless networks In this paper, we conduct a measurement study on three popular mobile video call applications: Face- Time, Google Plus Hangout, and Skype, over both WiFi and Cellular links We study the following questions: 1) how they encode/decode video in realtime under tight resource constraints on mobile devices? 2) how they transmit video smoothly in the face of various wireless network impairments? 3) what is their delivered video conferencing quality under different mobile network conditions? 4) how different system architectures and design choices contribute to their delivered quality? Through detailed analysis of measurement results, we obtain valuable insights regarding the unique challenges, advantages and dis- advantages of existing design solutions, and possible directions to deliver high-quality video calls in wireless networks

Intelligent end user devices for clearinghouse services in an internet telephony system

Abstract: Clearinghouse services architectures that support the use of end user devices, such as personal computers, Internet Protocol (IP) phones, cable multimedia terminal adapters, and residential gateways, in an Internet telephony system. The innovative architectures include a proxy-based system model, a direct communication model, and a hybrid proxy/direct communication model. A user can operate an “intelligent” end user device. i.e., a device running a client program with knowledge of the architecture particulars, to access a clearinghouse service on an IP network. This enables the user to communicate a telephony call over the IP network and via the combination of a terminating gateway identified by the clearinghouse service and the Public Switched Telephone Network.

System and method for call and data matching in a contact center

Abstract: A system and method for processing interaction requests includes a computer system that receives over a data channel from an end user device accessible to the customer, a request for an interaction. The request includes data from an application running on the end user device. The data is stored by the computer system upon receipt. When a voice call from a telephony device accessible to the customer is received, a voice media session is established with the telephony device over a voice channel. The data is identified based on information associated with the voice call. A contact center resource is further identified for routing the voice call. The data is attached to a request to route the call to the identified contact center resource.

Population Mobility Dynamics Estimated from Mobile Telephony Data

Abstract: In the last decade, mobile phones and mobile devices using mobile cellular telecommunication network connections have become ubiquitous. In several developed countries, the penetration of such devices has surpassed 100 percent. They facilitate communication and access to large quantities of data without the requirement of a fixed location or connection. Assuming mobile phones usually are in close proximity with the user, their cellular activities and locations are indicative of the user's activities and movements. As such, those cellular devices may be considered as a large scale distributed human activity sensing platform. This paper uses mobile operator telephony data to visualize the regional flows of people across the Republic of Ireland. In addition, the use of modified Markov chains for the ranking of significant regions of interest to mobile subscribers is investigated. Methodology is then presented which demonstrates how the ranking of significant regions of interest may be used to estimat...

VoIP BASED TELEPHONY METHOD AND SYSTEM

Abstract: A virtual phone application providing telephone service, for example a virtual phone using a Voice over Internet Protocol (VoIP) network. An internet enabled device ( 10 ), configured to implement the virtual phone application, wherein the virtual phone application is assigned with a telephone number, such as a E.164 number, with which the virtual phone application can initiate or receive a telephone call and SMS via VOIP. The Virtual phone of internet enabled device ( 10 ) is able to provide a communication system for establishing regular telephony services using a VoIP network to any phone line such as telephone call and short message service (SMS). Further, the virtual phone application is able to provide a ring back tone to the caller, making an emergency number dialling via VOIP, doing an identification of user location (GPS).

Closed communication system

Abstract: A system is described herein that includes mobile communication devices and a cellular base station to communicate with a telecommunication network. The mobile communication devices access telephony service through the cellular base station when inside a geofence. The mobile communication devices determine whether they are inside the geofence based on distances from proximity beacons. When outside of the geofence, the mobile communication devices are prevented from accessing telephony service through the cellular base station. When inside the geofence, the mobile communication devices may access telephony service through the cellular base station. The mobile communication devices may only make and receive calls from mobile communication devices of a specific other system. When a call is received at the system, the mobile communication devices of the system are rung. To enable this, a serving call session control function of the telecommunication network forks a call directed to the system.

Voice over Internet Protocol

Abstract: Voice over IP or telephony via the Internet, characterizes a set of protocols/technologies [H.323, Session Initiation Protocol (SIP)], which provide vocal conversation in real time with relatively good quality at little or no cost, thanks to worldwide broadband connections. Traditionally, such conversations took place exclusively through a PC which was connected to the Internet and with the aid of a microphone, headphones and the appropriate software (soft phones). The call ended up in another similarly equipped PC at no extra charge other than the one needed to access the Internet, since this specific type of communication does not require a provider of standard land-line services, but only the Internet. In addition to this, there are autonomous telephone devices (VoIP SIP Phones) and analog telephone adaptors (ATA) on the market which directly connect to an IP network, like the Internet. With the right adjustments and assembly and without the need of a PC, use of this service is facilitated making VoIP even more accessible to its users. The object of this paper is the study, realization and operation of an initially autonomous system of telephony via the Internet protocol (VoIP) and subsequently, its connection to other telephony systems (either via the Internet or traditional telephony), for their intercommunication, using the advantages of VoIP. First, the existing phone network [Public Switched Telephone Network (PSTN)] as well as its development is retraced. Then, VoIP technology is analytically presented along with its functions, protocols used, as well as its advantages and disadvantages. The quality of the service (QoS) offered by VoIP and the factors that affect its quality are also studied. As we continue, the importance of IP Private Branch eXchange (PBX) and the additional services they provide will be presented. In conclusion, an application of VoIP for military use will be executed so as to enable the military to reap the benefits it offers.

Personalized User Session Information for Call Handoff

Abstract: Techniques for hosting a communication session between a first communication device and a second communication device in an IP network are described, A contact server establishes separate communication links between itself and the communication devices wherein the communication links may traverse one or more telephony networks. The contact server recei ves personalized user session data generated by a user of a communication device. The personalized user session data may be indicative of a subjective evaluation of the quality of the communication link or a preference for a particular telephony network. The contact server may then cause the communication device to establish a new communication link with the contact server and handoff from its current communication link to the new communication link when the subjective evaluation of the quality of the communication link crosses a threshold level or a condition for a preference for a particular telephony network is detected.

Telephone caller authentication

Abstract: Techniques for authenticating the identity of a caller in a telephone call are disclosed. The device generating the call implements encrypting caller identification information and initiating the telephone call, said initiating comprising providing a voice carrier signal including the caller identification information; and simultaneously providing a data carrier signal including the encrypted caller identification information. The receiving device implements receiving the telephone call; decrypting the encrypted caller identification information at the recipient device; and verifying the identity of at least an unknown caller at the recipient device by at least one of verifying the authenticity of a certificate used to encrypt the caller identification information; and comparing the decrypted caller identification information with the caller identification information from the voice carrier signal. A telephony device implementing such an authentication technique is also disclosed.

Systems and methods for managing an event scheduling request in a telephony system

Abstract: Systems and methods are disclosed for managing an event scheduling request in a telephony system. According to certain embodiments, a call between a first communication device and a second communication device is established. Upon authenticating a user identity associated with the second communication device, an event scheduling request is determined based on one or more attributes provided to the first communication device. A notification is provided to the second communication device based on the event scheduling request.

DTMF technology applied to the identification and control of a small mobile robot

Abstract: An element to emphasize in student learning is the use of a motivational strategy that it is able to generate the intention to learn. The incorporation of active and collaborative methodologies in engineering classes such as: Problem and Project based Learning (PBL) and Challenge Problem (CP) have allowed that the student should reach a much more significant learning, at the same time that it assumes the responsibility of his own learning. This fact allows to the students to acquire knowledge and to put in practical numerous theoretical bases of an entertaining and fun form. The use of these educational strategies helps in turn to the evaluation of skills, abilities, knowledge and competences acquired during the learning process, being confirmed easily of experimental form. To control electronic devices of a local or remote way always has supposed an example of additional motivation on the students. In this document there appears the use of this type of devices as educational resource. The tone dialing system has been and is widely used by the digital telephony as method of communication between terminals and computers. Thus there have been developed different applications based on a system of dialing DTMF (Dual-Tone Multi-Frequency) tones used in the standard telephone system. So there have been implemented several circuits constituted by the MT8870 as a tones decoder module and \iC PIC16F876 as manager of the control and signal processing. A mobile telephone recycling NOKIA 3310 has been used to transmit the different frequencies to the control circuit and a small autonomous robot is controlled by the DTMF tones generated by the system via a GSM network.

Method and apparatus for use in supplying power over a telephone line

Abstract: A system comprising customer premises equipment including a reverse power supply unit and an access node to which the customer premises equipment is connected for permitting telephony devices connected to the customer premises equipment to make and receive telephone calls via the customer premises equipment and the access node. The reverse power supply unit is suitable for supplying electrical power from the customer's premises to an access node via a twisted copper pair, the reverse power supply unit including a power consumption signature detector operable to detect if the power drawn from the reverse power supply unit corresponds to a signature power consumption pattern indicative of a telephony device being in or transitioning into an off-hook state when connected directly to the reverse power supply unit and not being connected via a current limiting interface.

System and method of operating a secondary mobile device

Abstract: Using various embodiments, methods and systems for operating a secondary cellular telephone are described. The secondary cellular telephone, which is associated with subscriber ID account information of a primary cellular telephone, can be used to receive incoming calls or make outgoing calls. The secondary cellular telephone, in one embodiment, has its own telephone number, which is different from that of the primary cellular telephone. Incoming calls made to the primary cellular telephone number, by association with the subscriber ID account information of a primary cellular telephone, can be forwarded to the secondary cellular telephone. Similarly, with the association of the subscriber ID account information of the primary cellular telephone, outgoing calls made by the secondary cellular telephone can be spoofed with the primary cellular telephone number such that the receiver of the call would assume the call was made via the primary cellular telephone. Various other novel features, such as, finding the primary cellular telephone and remotely erasing data on the primary cellular telephone, by using the secondary cellular telephone are described herein.

Live person detection in an automated calling system

Abstract: A method of operating a telephony system for determining whether a telephone call made to a call destination terminal is a live call which is answered by a live person and not by an answering machine, the system performing steps comprising detecting that the telephone call has been answered and that the call is in progress transmitting one or more voice messages in a transmitted signal to the call destination terminal analyzing a received signal from the call destination terminal to detect voice signals, and determining whether or not the telephone call is a live call by applying test criteria to relative timing of events in the received signal and transmitted signal, the test criteria being based on an expected pattern of speech and silence representative of normal conversation.

Video pause indication in video telephony

Abstract: The disclosure relates to video telephony and, more particularly, to techniques for detecting a video pause in a video telephony application. In one example of the disclosure, a method for video telephony comprises detecting, at a receiving device, that video data packets associated with a video telephony call have stopped arriving from a sending device, and determining that the sending device has paused a video portion of the video telephony call based on information contained in video control packets.

Systems and methods for outgoing call redirection

Abstract: Systems and methods for redirecting calls are provided herein. In some embodiments, a method for redirecting calls may include receiving a first call request from a first user device via a first telephony service provider network to establish an off-net call with a second user device registered on a second telephony service provider network, determining that the first user device is associated with an account of the second telephony service provider network based on a communication identifier of the first user device included in the first call request, and sending a notification message to the first user device indicating that the off-net call may be established as an on-net call over the second telephony service provider network.

Network telephony processing method for smart mobile terminal

Abstract: The invention relates to the field of mobile communication and mobile internet, and provides a network telephony processing method integrated with a voice call based on a public mobile communication network and a network call based on the mobile internet. The network telephony processing method is applied to a smart mobile terminal with a voice communication function. When a user conducts telephone dial-up, the smart mobile terminal can detect the network condition of the smart mobile terminal, a common call mode or a network call mode can be automatically chosen according to the detecting result to conduct dialing and communicating by telephone, and network access modes are automatically chosen. According to the network telephony processing method for the smart mobile terminal, the advantages of the network call are fully performed to lower the communication cost, the complexity of network users, network coverage, network quality and network charges is also completely considered at the present stage, application schemes which are simpler and more convenient to use are provided for the user, and conditions are created for establishing further converged communication.

Evaluation of wavelet measures on automatic detection of emotion in noisy and telephony speech signals

Abstract: Detection of emotion in humans from speech signals is a recent research field. One of the scenarios where this field has been applied is in situations where the human integrity and security are at risk. In this paper we are propossing a set of features based on the Teager energy operator, and several entropy measures obtained from the decomposition signals from discrete wavelet transform to characterize different types of negative emotions such as anger, anxiety, disgust, and desperation. The features are measured in three different conditions: (1) the original speech signals, (2) the signals that are contaminated with noise, or are affected by the presence of a phone channel, and (3) the signals that are obtained after processing using an algorithm for Speech Enhancement based on Karhunen-Love Transform. According to the results, when the speech enhancement is applied, the detection of emotion in speech is increased in up to 22% compared to results obtained when the speech signal is highly contaminated with noise.

Method and apparatus for intra-network roaming for IP telephony network

Abstract: A method of operating an internet-based telephony system comprises (1) determining that a telephony device associated with a customer of the internet-based telephony system has access to service of the internet-based telephony system; (2) sending a first location update signal to a public land mobile network associated with the customer; (3) receiving an indication that the public land mobile network associated with the customer no longer considers the internet-based telephony system as being the network visited by the telephony device associated with the customer; (4) determining the telephony device associated with the customer and the internet-based telephony system are in data communication, and (5) sending a second location update signal to the public land mobile network associated with the customer to register again the internet-based telephony system as the network visited by the telephony device associated with the customer.

A social interaction system based on cloud computing for mobile video telephony

Abstract: As the use of smartphones become popular, people heavily depend on smartphone applications to deal with their social activities. For this reason, traditional message texting between mobile applications does not fulfill the versatile requirements of social networking. Many mobile applications use multimodality to deliver multimedia messages including sticker, voice and photo message, video call, and snap movie to enhance the communicative capability. However, without face-to-face interaction, people may fail to detect the other side’s non-verbal social behavior such as fine-grain facial expressions, body movements, or hand gesture. During social interaction, non-verbal behavior conveys information about the involved individuals and help the speakers express their social emotion in an implicit way. It is so important for real-world face-to-face interaction but is often blocked on the mobile telephony. To cope with this problem, we propose an affective computing model to assist the representation of social emotion and then help the progress of social interaction on the mobile telephony. In this model, for the purpose of real-time affective analysis, we delegate the computing loading to the cloud side service and enhance the system’s scalability and availability. The result of this experiment approves the feasibility of our system design for the applications of social intelligent. Also, the system provides a research framework of the social intelligent system on the mobile telephony.

Conversational quality and visual interaction of video-telephony under synchronous and asynchronous transmission delay

Abstract: In this study, the effect of pure synchronous and asynchronous transmission delay was examined in an interactive videotelephony setting. It was found that the conversational quality rated by participants of the study was clearly dependant on the tested conditions. This outcome can to some extend be ascribed to the used task which facilitated the usage of the visual channel for social interaction. Besides the conversational quality, a measure to quantify visual interaction was investigated for the first time in delayed mediated communication. The overall motion extracted from the recorded videos increased if synchronous transmission delays were inserted in the video-telephony connection.

A comprehensive SPIT detection and prevention framework based on reputation model on call communication patterns

Abstract: VolP is a promising technology for voice transmission on IP-based networks and it has many advantages over PSTN. One of the most important threats in these networks is unsolicited bulk calls, known as SPIT (Spam over Internet Telephony). The aim in this paper is to present this topic, a survey on detection techniques and anti-SPIT mechanisms. We propose an Anti-SPIT mechanism which is based on calculation of reputation using analysis of behaviors compared between normal users and SPITTERs for allow or block calls. Both the Service Provider and the Callee have role in the detection process. Also call lists will be updated in time.

A Scalable Management Method for Asterisk-based Internet Telephony System

Abstract: Internet telephony is an Internet service which supports voice telephone using VoIP technology on the IP-based Internet. It has some advantages in that voice telephone services can be accompanied with multimedia services such as video communication and messaging services. In this paper we suggested an Asterisk-based Internet telephony system which can be easily scalable. Most current systems use text files to manage their configuration: SIP users, dialplans, IVR service and etc. But we designed the management system which introduces database tables for efficiency and scalability. It also supports web-based functions developed by using Asterisk, Apache, MySQL, jQuery, PHP and open source softwares. Key Words : Asterisk, Internet Telephony, IVR, Open Source Software, SIP, VoIP * 본 논문은 안식년 기간 중 연구되었음Received 27 May 2014, Revised 7 July 2014Accepted 20 August 2014Corresponding Author: Eun-Yong Ha(Anyang University) Email: eyha@anyang.ac.krⒸ The Society of Digital Policy & Management. All rights reserved. This is an open-access article distributed under the terms of the Creative Commons Attribution Non-Commercial License (http://creativecommons.org/licenses/by-nc/3.0), which permits unrestricted non-commercial use, distribution, and reproduction in any medium, provided the original work is ISSN: 1738-1916 properly cited.

Methods and apparatus for conducting internet protocol telephony communications

Abstract: IP telephony communications are conducted by sending both data produced by a CODEC that represents received spoken audio input, and a textual representation of the spoken audio input. A receiving device utilizes the textual representation of the spoken audio input to help recreate the spoken audio input when a portion of the CODEC data is missing. The textual representation can be generated by a speech-to-text function. Alternatively, the textual representation can be a notation of extracted phonemes.

G.fast Technology and the FTTdp Network

Abstract: This white paper gives an overview of use cases and applications of G.fast transmission technology that is currently under development. The goal of G.fast is to deliver data at fiber speed to the customers using telephony copper wires. This solution shall provide high speed services at reasonable deployment cost for many subscribers. Low power consumption and ease of use for the customers are the main goals of the technology. Hardware-based simulations shall demonstrate the capabilities of G.fast for different applications. The channel models used to build different network topologies are based on a joint work of Deutsche Telekom and Lantiq. Lantiq has provided solutions for specific core features of G.fast which are presented in this paper.

Method and system for determining telephony system response

Abstract: An approach for simulating a telephony system to enable the evaluation of rules employed by the system for routing inbound calls to representatives is described. A simulation manager receives call production data, representative production data, or a combination thereof as generated by a telephony system during a production run for directing one or more inbound calls to one or more representatives. The simulation manager also generates an expected call handling response of the telephony system during a subsequent production run due to (a) one or more subsequent inbound calls, (b) a change associated with the one or more representatives, (c) a change associated with the call handling rules, or (d) a combination thereof based on execution of a simulation of the execution of the telephony system.

Method and apparatus for voip communication completion to a mobile device

Abstract: Systems and methods for establishing an IP telephone call to a called party's telephony device include attempting to send a call setup request directly to the called party's telephony device, and requesting that a push notification be sent to the called party's telephony device. If the direct request does not reach the called party's telephony device, the push notification will result in the called party's telephony device contacting the IP telephony system to setup the requested call. In some embodiments, the request for a push notification is not sent until the direct request is determined to have failed.

Apparatus and methods for modem assisted video telephony

Abstract: Aspects of the present disclosure provide an apparatus and method for modem-assisted video telephony. As one example, a user equipment (UE) utilizes a modem to perform video telephony (VT) communication with a remote device through a wireless network. The UE determines a network metric indicative of congestion in the wireless network based on modem information obtained from the modem, independent of feedback information received from the remote device. The UE dynamically adjusts a data rate of the VT communication by an amount proportional to a quantity based on the network metric.